Q. Can measurement mics be used for recording music?

By Hugh Robjohns

Earthworks' omnidirectional QTC1 mic might look like one of the company's measurement mics, but it produces excellent results in the studio.

Can you tell me if it's common practice to use measurement microphones for recording? I have access to a good selection of measurement mics, and on paper they all have much flatter frequency responses than conventional 'recording' mics. So would they be a better choice when trying to record something without colouring the sound?

SOS Forum Post

Technical Editor Hugh Robjohns replies: Measurement mics are not often used in music recording for a number of reasons. When selecting the right microphone for the job, there are many more factors to consider than frequency response alone. In this case I'd suggest that polar pattern and self-noise are likely to be critical areas.

Measurement mics are almost always omnidirectional, which limits their practical miking applications somewhat. Spaced omni mics can sound great in a simple stereo recording, but close miking with omnidirectional mics never really works.

For a variety of technical reasons, measurement mics also tend to have very small-diameter capsules (typically 12mm or less) which means they are inherently noisier than the small (around 16mm) and large (around 25mm) capsules more commonly used in music recording mics. In general terms, the electronics of measurement mics tend not to be particularly quiet, either, because low noise isn't usually a requirement in measurement applications — it is far more common to want to know how loud something is than how quiet!

Having said all that, despite being designed with a different purpose in mind, some measurement mics can actually muster a performance which is equal to or even exceeds some high-end studio mics — but this is rare.

Published December 2004

KORG BEATLAB mini - RHYTHM TRAINER

Q. What can I use to trigger drums?

By David Greeves

 
Roland's HPD1 'Handsonic' is a great hands-on percussion controller.

I love banging away on my thighs and knees, and I feel that I make some of my most interesting drum patterns in this way. However, I have never felt I could get the same level from using MIDI drum pads, or bashing away at my MIDI keyboard. How could I build a controller with a pair of panels that are velocity-sensitive to sit atop my thighs? Preferably these would be as thin as possible to maintain the feeling of my playing style. Any ideas?

Ascher Nathan

News Editor David Greeves replies: First of all, you're not alone! Lots of people find bashing out a rhythm on the edge of the desk or on their knees much more intuitive than using a MIDI keyboard or the tiny finger pads on many drum machines. There are plenty of alternatives out there, though which is the right one for you will depend on your personal thigh-slapping style.

Some favour dedicated drum pad controllers which offer larger pads than most all-in-one drum machines. Akai's MPD16 provides 16 MPC-sized pads, while Roland's SPD6 offers six large pads. Both will sit happily on your lap. Roland's HPD15 Handsonic percussion controller, which we reviewed in SOS October 2000 could be an even better option. It has a very large circular pad, divided into various different zones. If you just want something to bash away at, you could ignore these (along with the HPD15's other bells and whistles, like its ribbon controllers and D Beam), hit the pad wherever you like and then change the MIDI notes in your sequencer once they have been recorded. Their timing and velocity will be preserved, and that's the important thing.

If you can't find a commercially manufactured device that meets your needs, you may have to take matters into your own hands. You could have a go at building your own controller using piezo transducers. Paul White's excellent how-to article on the subject is available on our web site — surf to www.soundonsound.com/sos/1995_articles/aug95/diydrumpads.html — and is recommended reading. The suggested method is to attach a piezo pickup to the underside of an old coaster or table mat, the top of which is then covered with rubber to provide a comfortable playing surface. A second coaster or mat is attached to the underside, seperated by spacers and the piezo pickup connected to a standard two-wire jack socket. However, you'll need a MIDI drum machine which accepts a drum trigger pad input in order to turn the signal from the piezo pads into MIDI. Not many machines have trigger pad inputs these days — the Alesis DM5 and DM Pro modules are a notable exception, and are listed in their current catalogue. You could also hunt down a second-hand Alesis D4 or Akai ME35T.

You will need to consider the issue of velocity sensitivity too. The pads which Paul White explains how to make are really intended to be played with sticks. You'll need to experiment with different materials to produce a pad sensitive enough to be played with your hands, whilst isolating the piezo tranducer sufficiently so it isn't triggered accidentally. In any case, home-made pads will be less velocity-sensitive than manufactured units, and even they can struggle to translate the subtleties of acoustic percussion (which includes knees!) into MIDI data. This is something you'll have to live with.

One final suggestion is that you get hold of a USB-capable dance mat (of the sort which accompanies arcade-style dancing games) and assign its sensors to trigger MIDI notes. This is easily done in Ableton Live and you could happily drape the matt across your knees. The sensors on this kind of device aren't velocity sensitive but timing could at least be recorded. You could even go one better, cut up the mat and incorporate the sensors in the world's first pair of MIDI lederhosen! Where you put the USB port is up to you...

KORG monologue | Next-generation monophonic analog synthesizer

Q. How do I compress a stereo source?

By Hugh Robjohns

When the TLA 5052 dual valve recording channel is in stereo mode, one set of knobs controls both channels.

I am recording using a spaced-pair miking setup and I want to apply light compression to the signal. Where in the signal chain should I insert the compressor (pre- or post-fade), and will I have to buy a dedicated stereo compressor to do the job properly?

SOS Forum Post

Technical Editor Hugh Robjohns replies: You will have to use a stereo compressor, or two mono compressors that can be linked together for stereo operation. It is also wise to ensure that all the control parameters on both channels are set the same — that means attack and release time, threshold, ratio and make-up gain settings. In many cases, when a stereo or dual-channel compressor is operating in stereo mode, or when two compressors and stereo-linked, one set of controls becomes redundant while the other controls both channels. However, this is far from standard practice, and any differences between the settings of each channel can produce some very odd and undesirable effects.

When applying compression to a stereo source, it's very important that both channels experience the same amount of gain reduction regardless of which channel signal exceeds the threshold — hence the need for both identical control settings and stereo linking between channels. If you use separate, unlinked compressors for the two channels, then if one compressor reacts to a peak that the other doesn't see, the stereo image will pull towards the uncompressed side, and your listeners will start to feel very sea-sick!
Uniform compression on both channels is particularly important if you're recording using a coincident mic setup, as stereo imaging from a coincident pair is determined by tiny level differences between the two channels. In the case of a spaced-pair miking arrangement — the kind of setup you'll be using — the stereo image is dependent on phase differences as well as level differences, but uneven compression will disrupt the stereo image just the same.

It doesn't matter whether you choose to insert the two compressor channels (one handling each mic of your stereo pair) as a pre-fade or post-fade insert in terms of the stereo linking requirements. However, if you insert it post-fader, then the channel faders effectively become threshold controls and will affect the onset and amount of compression.


Published October 2004

Human League keyboard player, Ian Burden talks about the synths Part 3

Q. Can I get a more accurate bass sound using a subwoofer?

By Martin Walker

Subwoofer placement is critical when using a 2.1 monitoring system, like Blue Sky's Media Desk.

What are your opinions on having a subwoofer in the studio? I have recently demoed the Blue Sky Media Desk 2.1 system and I was very impressed by its quality and its price, but I have a few concerns. I'm worried that having a sub in my bedroom studio may cause more problems than it solves. The room is on the second floor of a semi-detached house, has thin walls and a lot of other annoyances (fitted cupboards and so on) though I will be treating the room as best I can.

There are calibration tests that you can do for the 2.1 system but I cannot afford to spend another £200 or more on a spectrum analyser kit to carry them out. How easy is it to do these kind of tests by ear, and, in any case, can the response of this sort of system be made accurate in any kind of room, no matter how unsuitable? My music tends to be quite bass-heavy (drum & bass and hip-hop), so I want as accurate a bass response as possible. I also demoed the Genelec 8030As as a stereo pair, which sound very nice, but I have read that they need their matching sub to reach their full potential (the satellites go down to 58Hz).
So, what would give me a more accurate bass response under my conditions: the 2.1 system set up as well as possible by ear, or a standard stereo setup like the 8030As?

Simon Epstein

PC music specialist Martin Walker replies: Thankfully, small Genelec monitors like the 8030As don't hype up the upper bass area just above the point where their response suddenly disappears altogether like many ported (bass reflex) speakers do, which is what helps to make them sound so natural. However, their response does fall off pretty dramatically below 58Hz, so for drum & bass or hip-hop you'll still need some low-end reinforcement from a subwoofer like Genelec's 7050A model to judge the low end of your mixes — assuming, that is, that your thin walls still make this acceptable to those on the other side! So, I suspect that whichever of these two systems you choose, you'll end up with two smaller satellites and a subwoofer.

Personally I'm always a little wary of trying to achieve really low bass in a small room, whether using a large pair of speakers or small satellites and a sub. It's certainly possible, but without some acoustic treatment most rooms already have serious problems below 200Hz that will only be aggravated by trying to generate sizeable amounts of additional low end below about 50Hz or 60Hz.

There's some good advice to be found in Mallory Nicholls' article on choosing and installing a subwoofer from SOS July 2002. One useful technique for finding the optimum position for the subwoofer, as described in SOS April 2003's Studio SOS, is to start by putting it in the centre of your listening position and then crawl around close to the wall behind the satellites until you find the spot where the bass sounds most even and balanced, then put the subwoofer there.

Adjusting its level is also crucial. Remember that if you dial in unnatural, window-rattling bass your friends may be impressed, but your mixes will sound thin on most other systems. In the absence of any test equipment other than your ears, slowly increase the relative level of the subwoofer until it just starts to draw attention to itself as a separate source, and then back it down a little — the perfect setting should seamlessly add bass to the satellites without being obvious. By listening to some familiar CDs on your setup, you can judge its settings, gain a reference point for your own mixes and gauge how well they will translate to other systems.

Mind you, if you want an accurate bass response you'll have to install at least some acoustic treatment. In most rooms, you'll encounter room modes (the resonant frequencies determined by the room's dimensions) that give an extremely lumpy bass end with several huge peaks and troughs, which would still make judging mixes of drum & bass and hip-hop extremely difficult.

You can hear these yourself by playing a bass guitar over its bottom octave (the low 'E' on a four-string is 41Hz, and the low 'B' on a five-string bass 31Hz) or a clean bass sound on a keyboard. Most of the notes should have similar volumes, but a few will be significantly louder (peaks) and a couple will probably be a lot quieter (troughs). If you don't have any other test equipment, you can just move the subwoofer a few inches at a time and try again for a smoother response until you find the best position in your room.

However, in an untreated room this response will also vary hugely depending on your listening position, with huge bass levels in the corners (where the modes pile up). In the case of a small room like yours where your listening position will probably be about halfway between the front and back of the room, with no acoustic treatment you may even find an almost complete null at one high-bass frequency at the listening position. The easiest way to hear these positional variations is to play sustained bass tones at the peak or trough frequencies you've already determined and walk around the room — I guarantee you'll be shocked at how their levels will vary.

So if you want an accurate bass end, installing some bass traps in the corners, like the Real Traps Minitraps I reviewed in SOS September 2004 will provide a good start in flattening out the response of your room, and only then will your mixes sound neutral at the listening position, but also remain reasonably consistent when you move around.


Published November 2004

Human League keyboard player, Ian Burden talks about the synths Part 2

Q. What mics should I use on a snare drum?

By Hugh Robjohns

I am looking at buying a matched pair of SE Electronics SE1 mics for drum miking. I am prepared to pay more for the right mics, but would the SE1s be suitable for 'over and under' miking of the snare? If not, could you offer any alternatives for this kind of configuration?

SOS Forum Post
 
If you're going to 'over-and-under' mike a snare, remember to switch one of the mics out of phase. 

Technical Editor Hugh Robjohns replies: The SE1s are great as overheads, but I'd be wary of using them for close-miking a snare. In the case of jazz drumming you might get away with it, but for a heavy-handed rock drummer with a loud snare, you stand a very good chance of overloading the mic's internal head amp.

Looking at the published specs, the SE1 is rated with a max sound pressure level (SPL) of 130dB for 0.5% total harmonic distortion (THD). If you really want to use small-diaphragm condensers, I'd suggest something like the Rode NT5, which is rated at 143dB SPL (albeit at 1% distortion), and so should be able to cope with a close snare a lot better.

It all depends on the kind of snare drum sound you are after, but condenser mics on snares can sound rather lightweight and thin. They deliver the transient 'thwack' of the hit very well, but often lack body. On the other hand, a good dynamic mic, like the venerable Shure SM57, inherently 'soft-limits' the transients and gives a much thicker, more full-bodied snare sound.

The over-and-under technique can be useful, as long as you remember to phase-reverse one of the mics. This is because when the stick hits the batter head, the head moves away from the top mic, producing lower air pressure (an initial rarefaction), while the snare head moves towards the bottom mic, producing a rise in air pressure (an initial compression). If you mix these two together without inverting the polarity of one of them, the two opposite pressure waves will tend to cancel each other out, and result in a very thin sound. Flip the phase of one mic (usually the bottom one, but it depends on the phasing of any other mics around the kit), and you should get a really big-sounding snare.

It's a good trick to experiment with, but not essential, and in general if you position the right mic in the right place above a good, well-tuned snare drum with a decent batter head and a competent drummer, you should still get excellent results.

Published October 2004

Human League keyboard player, Ian Burden talks about the synths Part 1

Q. Can I use acoustic screens to prevent drum spill?

By Hugh Robjohns

Using mic pickup patterns to reduce spill.

I'm wanting to record a band all at the same time and in the same room, and my main problem is that the drummer is much too loud. I'll be recording them in a rehearsal room and I'm worried about the other performers' mics picking up excessive spill from the kit. Would it be possible or advisable to try and isolate the kit using some screens? I've seen live performances on TV where the drummer is boxed off with clear perspex screens. Would there be a suitable DIY alternative?

SOS Forum Post

Technical Editor Hugh Robjohns replies: It's virtually impossible to use free-standing screens to generate useful or effective sound isolation, but they can be helpful in reducing spill. They are most effective at high-mid and high frequencies — the laws of physics, the wavelengths of the sounds involved and the physical size of the individual screens determine which frequencies a screen will stop (or, at least, reduce) and which it won't.

Don't expect screens to be able to help much in controlling low-frequency spill — the long wavelengths of low-frequency sounds mean that the spill will get everywhere no matter what! But you can screen off some of the mid- and high-frequency sounds, such as cymbal splashes and snare noise.

In my experience, rather than trying to box in the drum kit in an effort to stop its sound from escaping, screens work much more effectively if you place them around the microphones that are picking up the most spill, to try to keep the unwanted sound from reaching them. In terms of reducing the amount of drum spill recorded, stopping it leaving and preventing it from arriving come to pretty much the same thing, but the latter is far easier and more effective!

You can also reduce the amount of spill by paying careful attention to where you position the musicians relative to each other in the studio, the polar response patterns of their microphones, and where you point those mics. Cardioid and hypercardioid pickup patterns (see diagram above), which reject sound coming from the rear and, to a lesser extent, from the sides, are the order of the day. Don't forget that, in addition to the direct sound from the kit, the other mics may also be picking up reflected sound from the surrounding walls. In such cases, an additional screen behind or to the sides of the performer and microphone can help to cut out those unwanted reflections.

Boxing in the drum kit with hard reflective surfaces (like perspex panels) often causes more problems than it solves. It is a necessary evil on sound stages sometimes, but as I say, I find you can get better results if you concentrate on trying to stop unwanted sound getting into specific mics from specific directions, rather than trying to stop the unwanted sound from leaving the source. That's more of a job for King Canute!

At the end of the day, you will probably have to live with a degree of spill, but this is not the end of the world, especially if you're aiming for a 'live' sound. In fact, in most cases a modest amount of spill only becomes a major problem if you want to overdub something like a completely different guitar solo, when you can still hear the original in the spill on other tracks! Barring any problems with phase cancellation (where spill from a more distant mic interferes with the close-miked signal — switching in phase reverse on the mixer channels for the distant mics should help solve the problem), spill can even help to knit the mix together in a subtle way.

As you are recording in a rehearsal room, in conditions which, I assume, will not be ideal, you will have to compromise — if you can get a dry enough drum sound, and can isolate each instrument enough to have control over the balance of the mix, consider your job done.

Published November 2004

Herbie Hancock and his Korg Kronos

Q. What is the whining noise coming from my soundcard?

By Martin Walker

I noticed a quiet high-pitched whine coming from my Tannoy Reveal Active monitors, which are plugged into my Spirit Folio F14 mixer. Eventually I found that the whine stops when I press the button on my Edirol DA2496 soundcard to select an external word clock.

I get the whine even if the Edirol is not plugged into the mixer, but not if the monitors are not plugged into the mixer. I currently have unbalanced cables from the mixer to my monitors.
I have to use the internal word clock. When any music is playing, it easily drowns out the whine, but I'd rather get to the source of the problem and sort it out.

SOS Forum Post

PC music specialist Martin Walker replies: If you're getting interference even when your soundcard isn't plugged into the mixer, but it stops when you disconnect the mixer from your speakers, it doesn't sound like a problem with the soundcard itself, although you may well be hearing sounds that originally come from the soundcard, probably from its word clock.

As you can hear the interference even when the connection between the soundcard and mixer is broken, I suspect that you're suffering from a ground loop problem which is causing background digital noises to be heard — contrary to popular opinion, ground loops don't only cause background hum. Once there are multiple earth paths, noises from mouse movements, hard drive accesses, and graphic redraws can become audible in the background. Breaking the loop normally results in these all disappearing.

If you have the option of fitting balanced cables anywhere in the chain (mixer to monitors, and soundcard to mixer) this should cure the problem, so try this first.

It could also help if you plug everything into one mains socket via a distribution board, since this will generally make any existing ground loop smaller. It's also worth reseating your soundcard in its socket and tightening down its backplate screw. This ensures a good earth connection from soundcard to the PC chassis.
Active monitors can be a source of problems because they are earthed via their mains lead and via the audio input lead. This is why using balanced connections helps.

However, the only sure way to completely cure ground loop problems is to temporarily unplug all your audio cables and start from scratch. First listen to your powered monitors to make sure they don't exhibit any background noises with nothing connected to their audio inputs (sometimes US models with transformers designed for 60Hz will buzz when running on 50Hz). Then connect the stereo outputs from your mixer to the monitors and check again — if you have the option to use balanced I/O, do so.

Finally, connect the inputs of your mixer one by one to your various synths and soundcard devices with them powered up, temporarily turning up the mixer output volume fairly high after each one is plugged in to see when any hums or buzzes appear. As soon as you hear any noises you've found the offending connection, and can either try making up special pseudo-balanced cables if your mixer has balanced inputs and the source is unbalanced, or try inserting a line-level DI (Direct Injection) box between the device and the mixer input.

Published September 2004

The members of S U R V I V E check out the Monologue

Q. What do the different 'colours' of noise do?

I'm familiar with white noise, but I've also seen mention of pink, red and even blue noise in your pages. What are these other kinds of noise, how are they produced and what are they used for?

Derek Amesbury

 
Technical Editor Hugh Robjohns replies: There are several 'colours' of noise, the most commonly encountered in audio circles being white and pink.

White noise is a signal with the property of having constant energy per Hz bandwidth (an amplitude-frequency distribution of 1) and so has a flat frequency response. It is mainly used for testing audio equipment.

Pink noise contains equal energy per octave (or per 1/3 octave). Of all the 'coloured' test tones, this one sounds most like naturally occurring noise (a waterfall, for instance) and the amplitude follows the function 1/f, which corresponds to the level falling by 3dB per octave, or 10dB per decade. Pink noise is mainly used for acoustic measurements.

Brown noise, whose name is actually derived from Brownian motion, is similar to pink noise except that the frequency function is 1/(f squared). This produces a 6dB-per-octave attenuation (20dB per decade).
Blue noise is essentially the inverse of pink noise, with its amplitude increasing by 3dB per octave (the amplitude is proportional to f); and violet noise is the inverse of brown noise, with a rising response of 6dB per octave (amplitude is proportional to f squared).

There are several ways of creating broadband noise. The simplest is to use a special noisy diode or transistor and amplify the noise it produces naturally. However, a more modern practice is to use a microprocessor to generate random numbers which are then converted with a D-A coverter to produce the noise signal. The various colours of noise are produced either by simple analogue filtering, or by clever programming of the random number generator used to create the noise in the first place.

volca kick & FM with Luke Edwards and Steve Levine

Q. What kind of stands are best for mounting monitors?

By Hugh Robjohns

Is it best to use table-top stands, floor stands or wall brackets to mount my speakers? They're Mackie HR624s, which aren't ported, so does it matter if they're positioned close to the wall?

SOS Forum Post

Technical Editor Hugh Robjohns replies: The HR624 is not ported in the conventional sense, but it does employ a 'passive radiator' which is mounted on the rear panel behind the amplifier chassis. Essentially, this is a port with a diaphragm stretched across it. Consequently, it is not a good idea to place the speaker hard against a rear wall, although, as the passive radiator is tucked in behind the amplifier chassis, it is impossible to place it hard against a wall anyway.

Personally, I'd recommend wall-mounting your monitors using good sturdy hardware that holds the speaker about 4 to 6 inches away from the wall. That way, the speaker vibrations are completely isolated from everything else and the speaker is held firmly in position. Adjust the wall brackets or the speaker mounting so that the tweeters are aimed at the listening position.
 
Mo-Pads by Auralex Acoustics.

Many wall brackets are designed to be fixed to the speaker with bolts through its back panel. Some speakers intended for wall mounting in this way come with bolt holes already in place (the PMC DB1 and Genelec 1029, for example). However, if you plan to do the mounting yourself, take great care — the crossover of passive speakers is often mounted internally on the back panel behind the terminal plate, so it might be safer to bolt onto the base of the speaker instead. You can easily check by removing the connector panel and having a peek inside the back of the speaker.

If wall-mounting is impractical, then floor stands are better than table stands, again because they're better at controlling vibrations. If you have to go down the table-mounting route, some form of high-mass damping is usually a good idea. Try either placing the monitors on high-density foam isolators, such as the Auralex Mo-Pads, or putting some Spectra Dynamics Deflex anti-vibration sheeting on the table. Place a small concrete paving stone or heavy quarry tile on top of the foam, and Blu-tak the speaker on top of that. The extra mass of the slab will help control and damp vibration, and the foam will help prevent low-frequency vibrations from passing into the table structure.

Published October 2004

Korg PA4X Sound Tutorial Part 3: Creating Layers and Splits

Q. How do I transfer SysEx files to my Korg Wavestation synth?

By Martin Walker

 
Korg's landmark Wavestation synth, as seen in the original review from SOS August 1990.

I bought some CDs of new sound banks for my Korg Wavestation off Ebay recently, only to find that most of them are in SysEx format rather than MIDI files. I've loaded sounds into my synths from MIDI files before using Cubase, but I don't know how to do it with a SysEx file. Can you help?

SOS Forum Post

PC music specialist Martin Walker replies: All you need is a small utility to download these SysEx files into your Wavestation. I use MIDI-OX (www.midiox.com) on my PC, which is a multi-purpose MIDI utility and SysEx librarian. Similar shareware or freeware utilities are available for the Mac, including Snoize's SysEx Librarian for Mac OS X (www.snoize.com/sysexlibrarian) and SysEx for OS 9 (http://members.cox.net/sgrace9/sysex/index.html). A quick Google search will turn up several alternatives for either platform.

I'd also suggest that you read my Korg Wavestation Tips article from SOS June 2002 for more details, because the Wavestation can be a little tricky to download SysEx to, as the files are so large in comparison with most other synths of the period.



Published September 2004

Q. Can dust affect microphone performance?

By Hugh Robjohns

 
Protecting condenser mics like Neumann's TLM103 from dust will prolong their lifespan. 

This is probably a silly question, but I was just wondering: if I left my Neumann TLM103 mic out on its stand when not in use for long periods of time, would its performance be affected, by dust in particular? It does have a lovely wooden case to store it in when not in use, but I would prefer to leave it on the shockmount if possible.

SOS Forum Post

Technical Editor Hugh Robjohns replies: If anything, taking the mic on and off the stand and shockmount is likely to cause more damage in the long term, and certainly increases the risk of dropping it! I would suggest leaving the mic rigged on the stand, and placing a clean plastic bag (a large freezer bag, for example) over the mic when not in use to keep the dust off. Don't seal the bag at the bottom (to prevent condensation), but make sure the bottom of the bag hangs down well over the mic to minimise dust entering the bag.

Many professional studios do exactly this, rather than putting the mics away in boxes or cupboards. Some even leave the mics connected to phantom power sources to ensure a constant internal temperature, but that might be going a bit far for a home studio.

Microphones are delicate instruments capable of detecting minute changes in air pressure, so keep them away from any drafts and never slam shut the lid of a box with a mic inside. Ribbon mics are particularly prone to damage in this way. Condenser mics are also prone to rapid changes in humidity, so avoid moving the mic from warm to cool places, and if you have to, allow the mic a chance to equalise in temperature before you power it up and use it. They also don't like airborne contaminents like cigarette smoke which will degrade performance quite significantly over time. Capacitor capsules can be cleaned and restored, but it is a very specialised job and only cost-effective for the higher-end mics. Dynamic mics, on the other hand, are pretty robust in most conditions — you have to try fairly hard to destroy a good dynamic!


Published October 2004

Korg PA4X Sound Tutorial Part 2: DNC – Defined Nuance Control, Controllers

Q. How do I compress a stereo source?

By Hugh Robjohns

When the TLA 5052 dual valve recording channel is in stereo mode, one set of knobs controls both channels.

I am recording using a spaced-pair miking setup and I want to apply light compression to the signal. Where in the signal chain should I insert the compressor (pre- or post-fade), and will I have to buy a dedicated stereo compressor to do the job properly?

SOS Forum Post

Technical Editor Hugh Robjohns replies: You will have to use a stereo compressor, or two mono compressors that can be linked together for stereo operation. It is also wise to ensure that all the control parameters on both channels are set the same — that means attack and release time, threshold, ratio and make-up gain settings. In many cases, when a stereo or dual-channel compressor is operating in stereo mode, or when two compressors and stereo-linked, one set of controls becomes redundant while the other controls both channels. However, this is far from standard practice, and any differences between the settings of each channel can produce some very odd and undesirable effects.

When applying compression to a stereo source, it's very important that both channels experience the same amount of gain reduction regardless of which channel signal exceeds the threshold — hence the need for both identical control settings and stereo linking between channels. If you use separate, unlinked compressors for the two channels, then if one compressor reacts to a peak that the other doesn't see, the stereo image will pull towards the uncompressed side, and your listeners will start to feel very sea-sick!

Uniform compression on both channels is particularly important if you're recording using a coincident mic setup, as stereo imaging from a coincident pair is determined by tiny level differences between the two channels. In the case of a spaced-pair miking arrangement — the kind of setup you'll be using — the stereo image is dependent on phase differences as well as level differences, but uneven compression will disrupt the stereo image just the same.

It doesn't matter whether you choose to insert the two compressor channels (one handling each mic of your stereo pair) as a pre-fade or post-fade insert in terms of the stereo linking requirements. However, if you insert it post-fader, then the channel faders effectively become threshold controls and will affect the onset and amount of compression.

Published October 2004

Korg PA4X Sound Tutorial Part 1: Sound Selection and Organization

Q. Is USB too slow for MIDI interfacing?

By Martin Walker

My question is about USB MIDI interfaces, which seem to be the only kind of MIDI interfaces people make now. I've just bought Tascam Gigastudio, and in the manual it says 'Note: Nemesys recommends ISA or PCI-based MIDI interfaces, as they are faster than USB or Parallel Port interfaces'. As well as being too slow, I've also read that USB is unsuitable for MIDI because USB MIDI has timing jitter that could smear the timing of dense MIDI passages. If software manufacturers think USB is so unsuitable that they discourage people from using USB MIDI interfaces, then why do hardware manufacturers make them, and to the exclusion of PCI MIDI interfaces ? Are there any multiple I/O PCI interfaces around any more?

SOS Forum Post

PC music specialist Martin Walker replies: I'm not surprised that you're confused, since there's lots of conflicting information around, and much of it is out of date. Although you've presumably just bought Gigastudio 2.5, that quote is actually from an FAQ document dated October 2001, which also says that Gigastudio is compatible with Creative Labs' SB Live soundcard using Direct Sound drivers (which is no longer true, since Gigastudio no longer supports Direct Sound under Windows 2000 and XP), and that laptops are not recommended due to the lack of GSIF-compatible PCMCIA soundcards (no longer true either, thanks to Echo's excellent Indigo range).

However, the most obvious giveaway is the mention of ISA-based MIDI interfaces, since I don't know of any modern PC motherboard that still has any of the now extremely elderly ISA expansion slots that you'd need to plug one in — the last time I bought one was back in 1998!

You could complain that manufacturers' support documents should be updated more often, but there is nevertheless still a grain of truth in the recommendation of PCI over USB, as the results of my two-part investigation into 'The Truth About Latency' in SOS September and October 2002 proved. My PCI-based MIDI interface gave me around 3.6ms input latency when capturing a keyboard performance, with a latency jitter of just 0.2ms; a serial-port interface gave around 4.2ms with jitter of 1.2ms; and a USB interface gave about 4.8ms with jitter of up to 1.9ms, all running under Windows XP.

This should hopefully prove to your satisfaction that USB isn't too slow, since an increase of just 1.2ms over PCI is simply not discernible while playing — a MIDI Note On message will itself last nearly 1ms, and a six-note chord could therefore emerge spread over 6ms.

I personally doubt that an increased jitter of up to 2ms would be noticeable in most situations either — many musicians can apparently detect timing jitter when it exceeds about 5ms, but below this it's likely to go unnoticed. It is possible that during dense MIDI passages the situation could get worse, but I don't think you should worry too much. Moreover, when playing software synths in 'real time' via MIDI, their timing jitter can be two to three times that of the MIDI interface.

Sadly, nowadays, it's extremely rare to find a PCI-based multi-channel MIDI interface — nearly all are USB devices. But there are various things you can do to minimise timing jitter problems with a USB MIDI interface. It's important to plug it into a dedicated USB port rather than a USB hub (powered or otherwise) so that the interface isn't fighting for its share of the USB bandwidth with other devices. Always use the latest interface drivers, and try not to use too many USB devices simultaneously, even if they are plugged into separate ports. Also, I personally still think it's tempting providence to try to run separate USB audio and USB MIDI interfaces simultaneously, since their drivers may end up fighting for supremacy.

The bottom line is that loads of musicians (including me) are now running multi-channel USB MIDI interfaces with no obvious timing problems, while some of the problems that others run into aren't necessarily due to the interface, but to other issues with their computers. When Nemesys wrote that FAQ, USB MIDI was still in its early stages, and things have improved since then.

It's also important to remember that while the capturing of a MIDI performance may be subject to a couple of milliseconds of timing uncertainty, many of us are relying more and more on software synths, whose playback timing is always accurate.

Published October 2004

Korg monologue Video Overview Part 4: Sequencer, Microtunings, and User Scales [with CC]

Q. Is it OK to paint acoustic foam?

By Hugh Robjohns

Can you tell me how I should go about painting my acoustic foam panels? They are currently white and rather ugly! Is there a particular type of paint — spray paint perhaps — or any other way to colour them that won't damage them or affect their acoustic properties?



SOS Forum Post

Technical Editor Hugh Robjohns replies: Acoustic foams work because of their open cellular structure. If you paint over the surface of the foam you are very likely to clog up the pores and the effectiveness of the foam in absorbing high- and mid-frequencies will be greatly diminished. I do know people who have used thin emulsion paint with a degree of success, but I really wouldn't recommend it.

A much better option would be to cover the foam panels with some cloth — anything thin and acoustically transparent would be suitable. Build a simple four-sided wooden frame to surround the foam panels, with a hardboard back onto which the foam can be glued (perforated board can be used to some advantage). The depth of the frame should be slightly greater than that of the foam, and you can then stretch a coloured fabric of your choice over the frame and staple or pin it tight at the rear.

This construction will have negligible effect on the acoustic properties of the foam, will allow any colour scheme you like, and will look more professional to boot!



Published November 2004

Korg monologue Video Overview Part 3: Modulation, LFO, Envelope [with CC]