Keyboard gets a first look at Garritan Instant Orchestra at NAMM 2013

Q. Why aren’t my vocals audible in the car?

I’ve used a ‘stereo-izer’/widener on a track, along with two more separately taken tracks panned far left and right, respectively. There are three tracks in total for the chorus of the track.

When the vocals play on the chorus, it’s barely audible, but only when I play the track in the car. There may be a difference on my iPod or my monitors, or my cinema system. But the difference in the car is very dramatic.

I can only guess that the separate audio tracks are cancelling each other out. (I’ve read something about phasing, I think). But why is the effect so pronounced in the car?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

Without knowing exactly what the source tracks are and what you’ve done, it’s hard to give a precise answer, but it does sound as though there is some serious cancellation going on between channels, and that also implies that the two channels are being summed to mono at some point in the car.

Firstly, that stereo-widening effect you mentioned will inherently reduce the mono compatibility and may well lead to significant sound-quality changes between listening in mono and listening in stereo. Secondly, if you have two near-identical signals, and one has an inverted polarity with respect to the other, adding them together will result in them trying to cancel each other out, leaving nothing behind at all. But for that to happen you (a) need to have opposite polarity signals, and (b) need to add them together (either electrically or acoustically).

From your descriptions, you can hear a hint of the effect on your other systems, but it is very pronounced in the car. What that suggests to me is that there is some out-of-phase aspect to the signals — probably caused by the stereo widener effect — but that your other systems aren’t summing the left and right channels together, so the full cancellation isn’t happening. However, in your car it would appear that the channels are being summed together somehow, and hence near total cancellation. Exactly how that is happening, I can’t say. It could be that your speakers are wired in a peculiar way, or that there is some odd signal processing (surround effects?) happening in your car hi-fi.

If you have two almost-identical signals, and one is polarity inverted in respect to the other, adding them together will result in phase cancellation. This could be causing our reader’s vocal track audibility problems.

The only way to get to the bottom of this is to analyse the source material carefully for phase anomalies, and use some known reference material to evaluate exactly what is going on in your car. 

Published in SOS August 2012

Schubert: Symphony No. 8 "Great" / Rattle · Berliner Philharmoniker

Q. How can I record my band with limited space and equipment?

I’m in a band and we’re hoping to record ourselves live in our tiny rehearsal space using the limited equipment we have. My plan so far is to have drum overheads, snare, and kick mics going into channels one to four of my Alesis Multimix 8. Everything else (vocals, guitars and bass) will be miked into the PA system, then I’ll go out of that into channels five and six on my Alesis and record a stereo mix into Reaper (on my laptop) via USB from there. I figure that if I get the EQ and panning right first, that should give a reasonable representation of how we sound, but am I missing something? I’m worried that the PA speakers will feed too much into the drum mics; for space reasons, the PA speakers are just behind and to the sides of the drums. 

But if I unplug the PA speakers, we won’t all hear the vocal. If you have any words of wisdom then I’d be most grateful. We have two regular dynamic mics, two small-diaphragm condensers, and five dynamic drum mics.

With affordable stand-alone multitrack recorders, such as the Zoom R16, offering eight-channel simultaneous recording, musicians on a budget have the tools to create surprisingly respectable small-scale live-band recordings.

Via SOS web site

SOS contributor Mike Senior replies: 

Given the cramped conditions, I’d recommend turning off the PA speakers first of all, and living with the compromise that the other players won’t hear the vocals. In practice it shouldn’t really matter as long as everyone can see each other well, and everyone’s fairly clear on the structure of the song. Beyond that, though, your general miking/routing plan seems feasible, and here are a few tips you might find handy.

Firstly, I’d try to catch as full a drum sound as possible through the overheads. That usually means not sticking them right above the cymbals; either side of the drummer’s head is often a better starting point balance-wise. Given the inevitable resonance-mode problems in small rehearsal rooms, you’ll almost certainly want to roll out quite a bit of low end using the Alesis Lo-EQ controls on the overhead channels, but otherwise try to get the best sound you can from the drums by repositioning the mics. Remember that cardioid mics tend to give their brightest sound for whatever they’re pointing most directly at.

As far as the snare is concerned, try not to get the mic so close to the drum that all you get is ‘donk’. Whatever you end up with, though, hopefully your overheads should supply enough snare sound that you don’t have to use the close mic much. Try to baffle the kick mic in some way (or put it inside the kick drum) so that you don’t get masses of bass spill and low guitar woolliness on it, especially since you’ve got no facility to gate it. The one thing you’re really missing on the Alesis mixer is any phase/polarity control, so if you have any phase-inversion XLR leads (leads that swap the hot and cold XLR pins), then have those handy in case combining the snare or kick with the overheads sucks the heart out of your drum sound.

Given that spill is going to be a fact of life here, I’d be tempted to grasp the bull by the horns and make the best of the situation in that respect. In other words, I’d actually not try to separate the guitars and bass from the drums especially, but rather put them as close as they’d be on stage so that you get more of the benefit of a live-style performance situation (albeit without vocals). If you mic the guitars close, spill from the drums should still be fairly low in level if the instruments are well-balanced in the room, and it may actually improve the overall drum sound. If not, then try moving/rotating the whole ‘guitar plus close mic’ setup a little to get a better result, or try another polarity-flip XLR cable. Again, you’ll probably want to roll quite a bit of low end out of the guitar close mics, given the spill situation and the likely strength of the proximity-effect bass boost.

With the bass, I have to say that, again because of the inevitable room-resonance problems, I’d record his DI rather than his amp if at all possible (through something like a Bass Pod if an amped sound is really important), even if he still has the amp live in the room for performance purposes. I’d have the vocalist out the front of the drums facing the drummer, and then put duvets or something behind him/her to soak up some of the spill. If you have something like an SE Electronics Reflexion filter you can put up around the mic, then that’d help the spill issue too, but be careful not to interfere with sight lines between the players. Once more, low cut on the vocals will probably help stop the overall mix sounding muddy.

Setting all this up without the luxury of a separate monitoring room will be a challenge, but the best way (if a little time-consuming) is to do quick test recordings as you go, so you can judge the sounds without the spill from the room putting you off. You can make life easier for yourself in this respect if you do your best to get the sound in the room as close to the sound you’re after on record as you can. In practice, I’d expect it to take two or three hours of experimentation to get a reasonable sound going in this way, not including the time taken to set up the instruments and plug up and test the mic lines, so my final advice would be just to allow yourselves enough time, and warn the other band members that they might need a bit of patience!

If you were able to lay hands on an eight-channel interface of some kind (or even a small eight-track multitrack recorder: the Zoom R16 is ridiculously affordable, for instance) then that would afford you a lot of scope for improvements in a separate mixing stage. It’d also take some of the pressure off you in terms of judging the best phase/polarity relationships between the different mics right there on the session, so I’d seriously consider making that investment.

Published in SOS December 2012

Korg MS-20 Mini: The Legendary All-Analog MS-20, Reborn!

Q. How do I get the best from guitar-amp simulator software?

Due to space and other limitations, I am not able to mic up my amp, so I’m going to have to rely on plug-ins for my guitar recordings, but I’m worried about how close I can really get to the sound of my amped-up guitar using this method. What do you think is the best amp-sim plug-in for guitar? And what can I do to make the sound better or more realistic? Is this a fairly common practice with DAWs, or should I be finding a way to mic up a real amplifier?

Via SOS web site

SOS Reviews Editor Matt Houghton replies: 

The first thing to say is that yes, use of software amp-modelling in home and professional DAWs is commonplace, and the technology has come a long way in recent years: the sounds you can achieve with one of many commercially available software amp and speaker simulators are really pretty good now. What tops your personal list will be pretty much down to taste. For example, IK Multimedia’s Amplitube, Native Instruments’ Guitar Rig, Line 6’s Pod Farm and Peavey’s Revalver III are all capable of great results, but Softube Vintage Amp Room is my favourite software for re-amping work. 

That’s partially due to the amps that are modelled, partly due to the quality of those models, and partly due to the simplicity of the interface, which isn’t cluttered with a gazillion effects and preset menus, and offers only three amp models. This means that it’s easy to get to know it inside out and back to front, just like you would a real amp.

There are some excellent amp sims on the market, such as the Softube Vintage Amp Room, which benefits from the simplicity of its interface.

Whatever your personal preference in terms of sound, though, the key with any of this software is to keep latency as low as possible while playing, and to play it pretty loud over your speakers if you can, just so that you can get some interaction between the speakers and your pickups, as you would with a real amp. It’s the ‘playability’ side of things that concerns me most with much current software, and hardware modellers too: while the sound itself can be great on playback, and is usually perfectly good for re-amping purposes, I find the performance can suffer when you’re not playing and monitoring through your amp. To get around this, in part, you can use a basic modelled speaker emulation while tracking — rather than an impulse response, so that the convolution process isn’t adding unnecessarily to the latency — and then maybe experiment with speaker impulses later on, to produce a more realistic recorded sound. On the question of latency and playability, remember that you’re often standing away from a loud guitar amp, and if you’re playing closer to your monitors, that should compensate a little for the latency, simply because the sound reaches your ears that bit sooner than usual. When it comes to impulse responses, when they’re played back through a suitable convolution engine (there’s one bundled with most DAWs in the guise of a reverb), they can sound very convincing, but do bear in mind that they capture a static response and, therefore, don’t offer you control over mic selection and position, and don’t respond dynamically to variations in level as a real speaker would.

Notwithstanding this advice, though, I typically only use the software modellers if I have to, as I know I can get better sounds from a real amp. To keep amp recordings quiet in a compact space, there are a few options, which basically revolve around sticking the speaker inside a box to keep the sound down, if that’s one of your concerns, which you can do with isolation cabs such as the Hermit Cab. Alternatively, you could put a power soak in between the amp and speaker, so that you can attenuate the signal after it has passed through the amp but before it hits the speaker. If space is at a premium, this might not be the best approach, and while power soaks all allow you to use your amp, most impart plenty of their own coloration to the signal, which may not be to your taste. I much prefer to play through a nice tube-amp head into a power soak (such as the THD Hot Plate or the Sequis Motherload), and while I usually just use this to attenuate the output (so that I can drive the amp harder at sensible recording levels), you can use a power soak to attenuate the signal completely, and run a line output into your DAW. You’ll still need something to model the speaker, of course, and some power soaks include a reasonable speaker emulation, or you could, once again, look to impulses such as those from Redwirez. A slightly less complex variation on this theme is to use a high-quality guitar preamp (or take a preamp output feed from your amp if you have one) and run the resulting signal through a dedicated power amp and speaker emulator such as the Two Notes Torpedo VB101 plug-in. Again, it’s difficult to say how good you’ll find the results from these different approaches. 

My sense is that if you’re comfortable playing through software, and happy with the sound, it’s a problem that doesn’t need solving. But if you really do want to go in search for the ultimate compact and quiet recording solution for guitar, then I hope I’ve given some useful pointers!  

Published in SOS January 2013

WNAMM13: 'King Korg Video

Q. When was the click invented?

I’ve been wondering: when was recording to a click first used? And when did it start to become widely used? Did they used to record to metronomes before the advent of MIDI? I’ve tried searching the Internet for answers to these questions but haven’t found any answers.

Via SOS web site

We owe a lot to Disney’s production of Fantasia, which, among other things, introduced the method of using a click track.

SOS Technical Editor Hugh Robjohns replies: I’m sure people must have recorded using metronomes in the early days but, as far as I’m aware, the first documented use of a ‘click track’ in the modern understanding of the term was by Walt Disney’s team for the Fantasia film soundtrack back in 1940. The requirement was to be able to pan different orchestral sections around the auditorium via six speaker arrays: three across the front and three across the back. To do that, they needed to record the sections to separate tracks (remember this was in the days of mono optical film audio machines locked together with chains and sprockets!) and so they used a click track to keep the sections in time on each take.

The Fantasia project introduced a lot of things we take for granted today such as click tracks, pan-pots, VCA level automation, multitrack recording, overdubbing, surround sound and more besides!

However, the Second World War took the focus away from sophisticated surround-sound cinema productions, and the click-track idea didn’t really surface again until MIDI sequencing and quantising became commonplace in the 1980s. 

Published in SOS February 2013

[NAMM] Korg MS-20 Mini

Q. Why shouldn’t I use mastering limiting during mixing?

I often read recommendations to mix with compressors/limiters in the main bus so we can adjust to the effects of mastering during our mixing, and then to bypass those dynamics plug-ins when we export for mastering. Why not put in the full mastering chain and mix and master your track in one pass? I mixed my latest track with the following chain in the master bus: Cubase’s full-band compressor with a 1.2:1 ratio for 4-5dB of reduction; Powercore EQsat plug-in with a broad four-octave dip of 1dB at 850Hz; Powercore Master 3X multi-band compressor plug-in operating at a 3.2:1 ratio with 2dB of gain reduction; and ToneBoosters’ Barricade limiter set to a -1dBFS ceiling and showing 3-4dB gain reduction.

Damien McEwan, via email

Although full-band compression and EQ inserted on the master channel can often be helpful during mixing, multi-band dynamics and mastering-style limiting are more likely to hinder your progress.

SOS contributor Mike Senior replies: 

Using a compressor on the main mix bus during mixdown is indeed very common (although by no means universal) in order to ‘glue’ the mix together or create extra excitement via gain-pumping effects. Given that this bus-processing can impact quite heavily on the way you balance the track, it makes sense to have it working while you mix, particularly so that you can judge your effects levels and fader automation sensibly within context.

However, limiting the main output bus during mixdown is a whole different kettle of fish, because the main purpose of full-mix limiting is simply to boost the subjective loudness within the digital headroom. As such it’s usually much faster-acting, and the goal is usually to make as little difference to the mix balance as possible. Furthermore, setting up a limiter for the best results is usually a delicate process, where small shifts of the input level and plug-in controls can make big differences to the sound. So on the basis that mastering limiting shouldn’t normally affect mix balance, and that it adds to the already considerable complication of creating a decent mix, I usually recommend that this process be left until after mixdown.

Clearly there are some chart-oriented producers for whom the loudness of the master is an important primary concern. In that context having a preview of what the side-effects of heavy-handed loudness processing (including limiting) will do to the mix tone and balance can allow some pre-emptive compensatory steps to be taken by the mix engineer. However, even in that case, I’d favour bouncing your mix out to a separate project to experiment with this processing, even if that means that you then have to hop between the mix and a pseudo-mastering project. One reason I prefer working this way is that it puts fewer limitations on the mastering-style plug-ins I can use within my PC’s available CPU resources, and usually makes it a lot simpler to switch between my own pseudo-mastered mix and a selection of commercial reference tracks — an essential process when judging the results of your own mastering. Also, from a psychological perspective, being unable to immediately enact changes on the mix during the comparison process encourages me to clarify my own thoughts on the deficiencies of my own production across my available monitoring systems, and I find that this means I go round the houses less often while finalising my mix settings.

Equalising your main output bus at mixdown is pretty common. It’s very easy while you’re working on a mix for your ears to get used to a skewed tonality (they’re very good at adapting), and if this shows up during comparisons with commercial tracks then it’s much easier to deal with using a decent-quality master-bus plug-in than by tweaking the individual EQ settings across dozens of individual tracks.

Multi-band compression, on the other hand, is another thing that I suggest leaving to a separate mastering stage. Again, this is because it’s so fiddly to set up properly, and you’re not normally looking for it to impact hugely on the mix tone or balance; the heavy multi-band compression of the late ‘90s hasn’t aged well, and isn’t very fashionable these days. Also, in my experience, it’s very easy to take your eye off the ball as regards getting the mix balance right when there’s a multi-band compressor in the master bus, because the compression can often counteract your mix settings and disguise subtler balance problems that need addressing. Or, to put it another way, it tempts to you think that the mix is easier than it is somehow, so you work less hard.

Published in SOS February 2013

SONY ACID Pro 7 tutorial part 5 - Working With MIDI Controller Data

Q. Is there a better way of controlling sibilance than a de-esser?

A recording of mine is suffering from excessive sibilance, so I have tried processing it using a de-esser. The problem is that the processing seems to be having a detrimental effect on the sound of the vocal. Is there another way I can get rid of the sibilance, or something I can do differently?

 

Dan Simpkin via email

 

 

The vocal immediately following the Now Time marker (green line) sings ‘Too late a ghost’. The ‘T’ of ‘Too’ is attenuated, after which the automation line rapidly slopes back up to reach the track’s general level. The next attenuation is the ‘T’ in ‘Late’, which almost stands on its own and is easily tackled. Finally, the ‘S’ and ‘T’ at the end of ‘Ghost’ were reduced together.

 

SOS contributor Tom Flint replies:

 

Ideally, a de-esser should automatically attenuate troublesome sibilance without its actions adversely affecting the perceived quality of the audio, but the problem is that a static set of de-esser parameters often doesn’t work well for an entire performance. Not all of the sibilants within a single performance will be equally objectionable, and the energy in different sounds such as ‘S’ and ‘T’ might occupy different frequency ranges. One solution is to automate the threshold and frequency parameters of your de-esser so that it only works as needed, but my feeling is that if you’re going to get into using automation to control sibilance, it makes more sense to directly automate the level of the vocal track within your DAW. It can be time-consuming, but offers precise control over each individual instance of sibilance, and enables you to visualise the problem within your DAW’s arrange page.

 

The first thing to do is to enable level automation on the vocal track, and display the volume parameter for editing with the mouse. The next thing to do is find the first problem instance of sibilance. I use the horizontal and vertical zoom controls to expand the waveform until the contours of the word containing the problem are clearly defined. It’s often possible to identify the syllables in a word just by looking at it in close-up.

 

Once you’ve identified a problem sibilant, create a pair of automation nodes at either end, in effect bracketing that syllable. You can then select the two innermost nodes and drag them down to reduce that sibilant in level without affecting the rest of the track. I very occasionally find it necessary to add a few more nodes while getting the shape right, and delete those that turn out to be superfluous at the end. Having a small number of nodes is good practice as it means that level adjustments can be done quickly by moving a single line. The screen below shows a short phrase in Cakewalk’s Sonar with level automation used to control sibilance in this way.

 

After attenuating the worst instances of sibilance, I listen through to see how natural the overall performance is sounding. It may be the case that after reducing the level of the main offenders, the ear becomes less bothered by the others and a more lenient approach can be taken thereafter. Sometimes as little as a 3 or 4 dB drop in level results in a significant improvement, but it is also very surprising how much reduction can be applied without it sounding at all odd. The most important thing to note is that although you can often identify syllables visually, only your ears can tell you how much attenuation is needed, and how much will sound natural. In other words, what sounds right is right. Manually automating the level provides an opportunity to make alterations that better suit how the brain interprets what it is hearing.

 

One of the great advantages of using automation to control sibilance is that you’re not limited to applying it just to the level of the track. For instance, it’s often the case that sibilance is exaggerated by auxiliary effects such as reverb or delay. If this is the case, try copying your automation curve to the relevant send parameters on your vocal track, then scaling it drastically so that the level of those sends is heavily attenuated for sibilants. Likewise, any high-frequency EQ boost on your vocal track can make sibilance worse, so try copying the same automation curve to the gain parameter on the relevant band of your EQ plug-in.

 

When the level automation is complete, overall level adjustments can be made by sending the channel to an intermediate bus and using its fader to boost or cut by the required amount, thereby leaving the sibilance control automation undisturbed.

Published in SOS February 2013

Mozart: Piano Concerto No. 17 / Pollini · Abbado · Berliner Philharmoniker

Q. How can I warm up my recording without using EQ?

Sound Advice : Recording

I’ve put a lot of effort into creating and editing a recording of solo mandolin — played quite slowly — and although I like the final result a lot, on consideration the tone is too trebly and cold, almost like a photograph with too sharp a resolution. A friend mentioned he thought I could perhaps ‘warm it up’ using compression, perhaps of a type designed for vocals. Can you give me some guidance on how best I might do this? Of course, I realise I can use EQ, but would specifically be interested in any thoughts on how compression/limiting could be used on an existing take to get a warmer result. I’ve used Logic and the recording is clear, undistorted, and free from ambient sound.

Simon Evans via email

SOS contributor Mike Senior replies: 

There are ways to warm up a mandolin sound subjectively using compression, although none of them are likely to make as big an impact as EQ. Fast compression may be able to take some of the edge off a mandolin’s apparent tone, for instance, assuming the processing can duck the picking transients independently of the note-sustain elements. There are two main challenges in setting that up. Firstly you need to have a compressor that will react sufficiently quickly to the front edges of the pick transients, so something with a fast attack time makes sense. Not all of Logic’s built-in compressor models are well-suited to this application, so be sure to compare them when configuring this effect; instinctively I’d head for the Class A or FET models, but it’s always going to be a bit ‘suck it and see’. The second difficulty will be getting the compressor not to interfere with the rest of the sound. The release-time setting will be crucial here: it needs to be fast enough to avoid pumping artifacts, but not so fast that it starts distorting anything in conjunction with the attack setting. Automating this compressor’s threshold level may be necessary if there are lots of dynamic changes in the track, for similar reasons. Applying some high-pass filtering to the compressor’s side-chain (open the Logic Compressor plug-in’s advanced settings to access side-chain EQ, and select the ‘HP’ mode) may help too, because the picking transients will be richer in HF energy than the mandolin’s basic tone.

The Advanced Settings panel in Logic’s built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you’re trying to sensitise (or desensitise!) the compressor to a mandolin’s picking transients.

Another way to apparently warm up a mandolin is to take the opposite approach: emphasise its sustain character directly while leaving the pick spikes alone. In a normal insert-processing scheme, I’d use a fast-release, low-threshold, low-ratio (1.2:1 to 1.5:1) setting to squish the overall dynamic range. Beyond deciding on the amount of gain reduction, my biggest concern here would be choosing an attack time that avoided any unwanted loss of picking definition. In this case, shelving a bit of the high end out of the compression side-chain might make a certain amount of sense if you can’t get the extra sustain you want without an unacceptable impact on the picking transients.

Alternatively, you might consider switching over to a parallel processing setup, whereby you feed a compressor as a send effect, and then set it to more aggressively smooth out all the transients. The resulting ‘sustain-only’ signal can then be added to the unprocessed signal to taste, as long as you’ve got your plug-in delay compensation active to prevent processing delays from causing destructive phase-cancellation. Using an analogue-modelled compressor in this role might also play further into your hands here, as analogue compressors do sometimes dull the high end of the signal significantly if they’re driven reasonably hard, giving you, in effect, a kind of free EQ.

Published in SOS May 2012

SONY ACID Pro 7 tutorial part 4 - Punch In Recording

Q. Are some analogue signal graphs misleading?

Sound Advice : Mixing

I read your feature about ‘Digital Problems, Practical Solutions’ (www.soundonsound.com/sos/feb08/articles/digitalaudio.htm), which said that digital audio can capture and recreate analogue signals accurately, and that the ‘steps’ on most teaching diagrams are misleading. Does that mean that the graph should really show lines, or plot ‘x’s, instead of looking like a standard bar-graph?

Remi Johnson via email

SOS Technical Editor Hugh Robjohns replies: 

Good question! The graphs in that article are accurate as far as they go, but offer a very simplified view of only one part of the whole, much more complex, process.

When an analogue signal (the red line on Graph 1: Sample & Hold) is sampled, an electronic circuit detects the signal voltage at a specific moment in time (the sampling instant) and then holds that voltage as constant as it can until the next sampling instant. During that holding period the quantising circuitry works out which binary number represents the measured sample voltage. This, not surprisingly, is called a ‘sample and hold’ process, and that’s what that diagram is trying to illustrate.

Graph 1: Sample & Hold

So the sampling moment is, theoretically, an instant in time, best represented on the graph as a thin vertical line at the sample intervals (the blue lines in the picture Graph 1: Sample & Hold), but the actual output of the sample and hold process is the grey bar extending to the right of the blue line.

However, the key to understanding sampling is understanding the maths behind that theoretical sampling ‘instant’, and that means delving into the maths of ‘sinc’ (sin(x)/x) functions, which is the time-domain response of a band-limited signal sample. At this point most musicians’ eyes glaze over…

As we know, the measured amplitude of each sample from an analogue waveform is represented by a binary number in the digital audio system. When reconstructing the analogue waveform that number determines the height of the sinc function.

The important point is that we are not just creating a simple ‘pulse’ of audio at the sample point, because the sinc signal actually comprises a main sinusoidal peak at the sampling instant (and of the required amplitude), plus decaying sine wave ‘ripples’ that extend (theoretically for ever) both before and after that central pulse. The reconstructed analogue waveform is the sum of all the sinc functions for all the samples.

The clever bit is that the points where those decaying sinc ripples cross the zero line always occur at the adjacent sampling instants. This is shown in the next diagram (Graph 2: Two Sinc Functions) where, for simplicity, just two sample sinc functions are shown for samples 23 (red) and 27 (blue). You can see that at the intermediate sample points (26, 25, 24 and so on) the sinc functions are always zero.

Graph 2: Two Sinc Functions

That means that the ripples don’t contribute to the amplitude of any other sample, but they do contribute to the amplitude of the reconstructed signal in between the samples, with the adjacent sample sinc functions having the greatest influence, and lesser contributions from the more distant samples. This is shown in the next diagram (Graph 3: 3kHz Sinc Addition), in which the sinc functions of a number of adjacent samples are shown, and when summed together produce the dotted line that is a sampled 3kHz sine waveform

Graph 3: 3kHz Sinc Addition

These last two diagrams have been borrowed from a superb paper by Dan Lavry (of Lavry Engineering), which explains sampling theory extremely well, and can be found here: www.lavryengineering.com/documents/Sampling_Theory.pdf.

Published in SOS May 2012

SONY ACID Pro 7 tutorial part 3 - Basic In Line MIDI Editing

Q. What’s the best system for backing up my work?

Having recently started making and recording my own music, I need to start thinking about backing it up. At the moment, I’m just keeping everything on my hard drive, which I’m somewhat nervous about (I’ve often heard people say that digital data doesn’t exist at all unless it exists in at least three places!), so I need to sort out a system quickly. What procedure/system would you recommend?

Julia Webber via email

SOS contributor Martin Walker replies: 

It’s very refreshing to find a musician who even thinks about backing up data at such an early stage: often people only consider the options having dried their eyes after losing a lot of irreplaceable songs. Hard drives can and do go wrong, and catastrophic failures can happen in a microsecond, leaving you unable to retrieve any of your previous files (companies do exist that specialise in bringing data back from the dead, but they tend to be expensive).

So it pays all of us to make regular backups, then we can laugh when disaster strikes and restore our most recent backup rather than lose any data: even if the very worst happens and the entire hard drive goes belly-up, it’s entirely possible to plug in a replacement drive and be up and running again within a couple of hours.

First, you need to decide how often you need to back up. To answer this question, just decide how much work you are prepared to lose. Many hobbyists and some professionals are happy to back up once a week, but always back up immediately you’ve finished an important session as well, just in case. Second, decide how best to organise your data to make each backup as easy as possible: after all, the easier it is, the more likely you are to do it, and consequently the less data you are likely to lose if anything does go wrong.

As long as your hard drives are well organised, even a freeware utility like Paragon’s Backup & Recovery can be ideal for backup purposes.

I prefer to organise my hard drives by dividing them into various partitions, each devoted to a specific subject such as Operating System + Applications, Audio Projects, Samples, Updates, My Data and so on. Most modern operating systems let you partition your drives in any way you wish. Although this takes a little more effort at the start of your backup regime, for me the huge advantage of separating your data from the operating system and applications is that you can take global backups of entire partitions using a Drive imaging utility such as Acronis True Image or Norton Ghost. This way, you’ll know that absolutely everything on that partition will be contained within each backup file (even those plug-in presets you create that get tucked away somewhere safe and then forgotten!).

The alternative is to leave all your data spread across the one huge default partition for each drive, and use backup utilities that let you specify which files to back up and which to ignore, such as Mac OSX Time Machine and Windows 7 Backup. Some audio applications, such as Wavelab, also offer dedicated backup functions. Once again, this takes time to set up initially, and this approach also relies on you specifying a comprehensive list of files to save, so if you forget something vital, you may come a cropper later on.

Whether you choose drive imaging or a dedicated backup utility, you can create a global backup file but, to save time and storage space later on, both may also offer the subsequent option of much smaller incremental backup files that only contain files that have been added or changed since your most recent backup.

The final choice is where to store your backups. The most important thing is to store them separately from the original data, so that they are unlikely to be damaged with the originals. If your computer has multiple hard drives, a very quick and easy regime is to store backups of one drive onto the other: this protects you if one drive becomes faulty, but not if your entire computer goes up in a puff of smoke.

For greater security, another set of backup data should be stored away from your computer, either on removable media such as USB sticks, CD-Rs, DVD-Rs, or removable or Firewire/USB hard drives. It also makes more sense to store these backups in a completely different location, so that even if your house burns down your data remains intact. Cloud-based online backups, such as Dropbox or Amazon S3 (Simple Storage Service), are very handy if you have a fast connection, although uploading speeds can be cripplingly slow compared to downloads. A much quicker and easier alternative may be to swap backups with local friends or family: you keep a regular copy of their backups and they keep a copy of yours.  

Published in SOS February 2012

SONY ACID Pro 7 tutorial part 2 - Sections

Q. Can I output my final mix one channel at a time?

Sound Advice : Mixing

I have recently purchased a Golden Age Project Pre 73 MkII and Comp 54 on the recommendation of someone from the SOS forums, and I am so pleased. I use an RME Babyface and wondered, with my limited hardware, would it be possible to output my final mix one channel at a time through the Comp 54? The reason I ask is that the hardware adds something that no VST seems to be able to do. If someone knows how I could do this it would be great. If it matters, the DAW I am using is Reaper.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

The answer is yes, but it’s not as straightforward as it might appear and you need to be careful.

The basic problem is that when you’re working with a stereo mix the stereo imaging is determined by the subtle level differences of individual instruments in the two channels. A compressor exists to alter the level of whatever you pass through it dynamically, depending on its own level.

Imagine an extreme situation where you have some gentle acoustic guitar in the centre of your mix image, and some occasional heavy percussion panned hard left. If you process those two channels with separate unlinked compressors, the right channel compressor only sees a gentle guitar and does nothing, while the left channel compressor will feel obliged to wind the level back every time the mad drummer breaks out.

While you may like the effect a certain piece of gear (like this Golden Age Project Comp 54) has on your recordings, passing your left and right channels through it separately is not a good idea. The reason for this is that the compressor can only react to what it is fed at any given time. So when the left and right channels are heard together — after being run through the Comp 54 — the sound will be very uneven. You can get around this by setting up an external side-chain input, which will cause the compressor to react to what it gets from the other channel, but with the Comp 54 this is not possible, so another approach altogether might be in order.

Listen to the two compressed channels afterwards in stereo and the result will be a very unsettled guitarist who shuffles rapidly over to the right every time the percussionist breaks out (probably a wise thing to do in the real world, of course, but not very helpful for our stereo mix).

If you process your stereo mix one channel at a time through your single outboard compressor, that’s exactly what will happen. The compressor will only react to whatever it sees in its own channel during in each pass, and when you marry the two compressed recordings together again you will find you have an unstable stereo image. The audibility of this, and how objectionable you find it, will depend on the specific material (the imaging and dynamics of your mix), but the problem will definitely be there.

Stereo compressors avoid this problem by linking the side chains of the two channels, so that whenever one channel decides it has to reduce the gain, the other does too, and by the same amount. In that way it maintains the correct level balance between the two channels and so avoids any stereo image shifts.

You can achieve the same end result if your single outboard compressor has an external side-chain input, but sadly I don’t think the Golden Age Project model does. If it did, what you’d need to do is create a mono version of the stereo mix in your DAW and feed that mono track out to the compressor’s external side-chain input, along with one of the individual stereo mix channels (followed by the other). That way, the compressor will be controlled only by the complete mono mix when processing the separate left and right mix channels, so it will always react in the same way, regardless of what is happening on an individual channel, and there won’t be any image shifting.

That’s no help to you with this setup, of course, but don’t give up yet, as there is another possibility. You could take an entirely different approach, and that’s to compress the mix in a Mid/Side format instead of left-right. It involves a bit more work, obviously, as you’ll need to convert your stereo track from left-right to Mid/Side, then pass each of the new Mid and Side channels separately through the compressor, and then convert the resulting compressed Mid/Side channels back into left-right stereo. Using an M/S plug-in makes the task a lot easier than fiddling around with mixer routing and grouping, and there are several good free ones around.

The advantage of this Mid/Side technique is that, although the Mid and Side signals are being processed separately and independently, the resulting image shifts will be much less obvious. The reason for this is that instead of blatant left-right shifts, they will now be variations in overall image width instead, and that is very much less noticeable to the average listener.

Sorry for the long-winded answer, but I hope that has pointed you in the right direction.

SOS Reviews Editor Matt Houghton adds: I agree with Hugh’s suggestion of M/S compression. I regularly use that when I want to deploy two otherwise unlinkable mono compressors, and there’s no reason why you can’t process the Mid and Side components one at a time. The only issue here will be your inability to preview what you’re doing to a stereo source, so be careful not to overwrite your original audio files! However, I sense that it’s the effect of running through the compressor’s transformers that you’re hoping to achieve. In that case, just set to unity gain and set the threshold so that the unit isn’t compressing, and then run the signal through it. If it is standard L/R compression you want, you could always get another Comp 54, as although they’re mono processors they’re stereo-linkable with a single jack cable.

In Cubase, I find that the best approach to incorporating such outboard devices into my setup is to create an External FX plug-in for each device, and then insert that on each channel and print the result. In Reaper, the equivalent tool is the excellent ReaInsert plug-in. This approach not only makes the process less labour intensive in the long run, but means that you can drag and drop the processor to different points in the channel’s signal chain, should you want to.

Published in SOS February 2012

SONY ACID Pro 7 tutorial part 1 - Screen Walk

Q. Does it make sense for me to mix with a mono 'grot box'?

Sound Advice : Mixing

I realise that there are advantages to monitoring on a single ‘middly’-sounding small speaker (such as an Avantone MixCube) from time to time while mixing, to get an idea of what the music might sound like on typical cheap consumer playback systems. However, I mix mainly deep house and lounge, which is quite rich in high and low frequencies, and these are easily conveyed by the full-range playback systems in trendy restaurants, cafes and clubs, but not by these small monitors. Does using one while mixing, therefore, actually make any sense for me? Also, if I did decide to get a single MixCube, I guess the best place to put it would simply be in the middle of the desk, but the problem is that that’s where my computer screen is! Should I buy a higher speaker stand to go above my screen and angle the Avantone down towards me?

Nicolas Issid via email

SOS contributor Mike Senior replies: 

If your music were only ever played on larger full-range systems like those you mention, the usefulness of limited-bandwidth referencing would indeed be reduced. However, I’d personally think twice about targeting the sound too narrowly for one type of playback system, and would be inclined to prepare my music for lower-resolution playback in case it, for any reason, gets transmitted for wider consumption — on the Internet, say, or as part of a TV programme, radio advert or computer game.

Rather than buying a mixer with lots of outputs and manually routing sound effects to different speakers for a theatre production, why not use a dedicated computer-based ‘playout’ system, in conjunction with a multi-output soundcard or audio interface? Some suitable software is even available free.

Even if you mix music primarily aimed at full-range venue playback systems, there’s still something to be gained at mixdown from checking your mix on a single small speaker.

That apart, though, I think you’re slightly underestimating the value of something like the Avantones, because they’re not just about the ‘middly’ frequency response. Their small-scale, single-driver, portless design makes them much more revealing as far as simple mix balance issues are concerned (ie. for deciding what level each instrument should be at) than almost any even marginally affordable full-range nearfield/midfield monitoring system. This is even more the case if you use only one such speaker, rather than a stereo pair, as you also avoid inter-speaker phasing issues. Overall, I think you’d still benefit a great deal from this kind of speaker even if you mix primarily for larger speaker systems.

As far as speaker placement is concerned, in my opinion it doesn’t really matter much where you put it, as long as it’s pointing roughly in your direction and you’re not getting acoustic reflection problems from a nearby room boundary or other hard surface. The only disadvantage of mounting a single speaker off centre is that it may temporarily skew your stereo perception to one side after you’ve been listening for a while. Not that this is actually a significant mixing problem in practice, though, because it’s very easy to work around.  

Published in SOS March 2012

KORG iMS-20 v1.1 Demonstration Movie

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Q. How important are microphone self-noise and SPL figures?

Sound Advice : Miking

I am interested in the Shure SM7b mic and have been looking at its specifications, but the Shure web site seems to be missing information for self-noise and maximum SPL levels. I’ve heard people saying that the SM7 can handle up to 180dB SPL! I’m curious as to whether or not that is true (probably not) and if it is anywhere near that, I’m assuming it’s because it’s got some kind of -30dB switch on it or something crazy like that. Can you shed any light on this?

You won’t find self-noise specifications for the Shure SM7b, as it is a dynamic (moving-coil) microphone. The only self-noise generated is the thermal noise from its own output impedance.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: The reason you can’t find those specific specifications is because the SM7 is a dynamic (moving-coil) microphone. In fact, you probably won’t find those specs for any dynamic mic from any manufacturer (other than dynamic mics with built-in buffers or gain stages), because they are largely meaningless and pointless figures.

The self-noise generated by a moving-coil microphone is only the thermal noise generated by the mic’s own output impedance, which is essentially just the DC resistance of the moving coil itself, plus that of a humbucking coil (if employed) and the output transformer (if present). This noise contribution is negligible, and will be utterly swamped by the receiving preamp’s own electronic noise.

The maximum SPL level for a dynamic mic is determined mainly by the range of mechanical movement afforded to the coil, and that will be more than high enough for any conventional application. So it’s not unusual to find professional dynamic mics that are capable of over 150dB SPL (for one percent THD), albeit with rapidly increasing distortion towards the limits, and with mechanical clipping occurring when the diaphragm and/or coil hits the end stops at 170 or 180 dB SPL.

In contrast, the self-noise and maximum SPL figures are quoted for all electrostatic mics (capacitor and electret) because the impedance converter electronics built into the microphone determine the mic’s dynamic range capability, the lower limit being set by the amplifier’s self-noise, and the upper limit by the amplifier’s distortion or clipping. 

Published in SOS April 2012

KORG iELECTRIBE v1.6 New Function "Beat Flutter"

Q. Can you recommend a low-cost heavy-duty mic stand?

Sound Advice : Miking

I have the usual selection of Stagg and anonymous mic stands, which are fine most of the time, but I now have some mics that are really pretty heavy (SE Electronics’ Gemini III, for instance) and none of my present stands really cut it. Of course, all mic stands are described as ‘heavy duty’, but I’m looking for something that can hold really heavy microphones reliably and with the minimum of hard twisting of small knobs and so on.Of course, SE make a suitable stand, but I’m not sure I could justify $500 on one mic stand. Can you suggest anything usable below, say, $150?

An expensive mic stand might seem like a waste of money, given that most still suffer from ‘droop’, but some very well-engineered stands exist that do not suffer from this problem. This stand from Sontronics, for example, is more than worth its cost, given that it is protecting far more valuable mics that could last you a lifetime if well looked after.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

If you can use a mic stand without a boom arm — so, just the vertical pole — there shouldn’t be any problem, because even budget mic stands should be able to support the heaviest microphone without too much trouble. The real problem comes when trying to hang a heavy mic on a boom arm, because most ordinary mic stands don’t have anything like a sufficient counterweight mass to properly balance even moderate mics, let alone big, heavy ones. As a result, the boom arm clutch has to resist almost all of the rotational force created by the leverage of the heavy mic at the end of the boom and, frankly, most just aren’t up to the job. The inevitable consequence is the annoying ‘droopage’, and the more you try to tighten the clutch to prevent it, the quicker the whole thing wears out (or breaks), and quickly becomes droopy even when supporting light microphones!

The correct engineering solution is to properly counterbalance the weight of the microphone so that there is no net rotational force at the boom clutch. That then allows the clutch to do what it was intended to do — stop the boom arm from moving — rather than have to accommodate the entire rotational leverage. The cheap and cheerful solution is to tape or affix some additional weight to the end of the boom arm; you need enough to balance your heaviest mic at the maximum boom extension you plan to use. However, this will be ugly and may not be as safe as it should be, and you certainly don’t want the weight to fall off onto someone’s foot... or the mic to crash onto the floor shortly afterwards!

I know the idea of spending $500 on a mic stand seems silly, but, to be honest, I think it’s worth it for peace of mind when you’re working with mics that cost $1500 and potential personal injury insurance claims! Moreover, mic stands in this cost bracket generally live forever, because they are so well designed and rugged, which means that the amortised investment is actually very low.

The SE mic stand is surprisingly stable, but it is a kind of hybrid of a reverse-engineered Keith Monks boom arm and clutch from the 1970s and a drummer’s cymbal stand. It does have a heavier counter-weight than most budget stands, but it’s still not an ideal solution, to my mind.

The most cost-effective and properly engineered stand I’ve come across to date is the Sontronix Matrix 10. 

It’s not the prettiest or most compact stand on the planet — it’s basically a modified photography lighting stand — but it has cogged clutches that definitely won’t slip, a very sensible counterweight, removable wheels, and a handy drop-arm. It’s very secure, totally reliable, and there’s nothing to break, so it will live forever. I reviewed it in the August 2010 edition of Sound On Sound (see the full review at  

www.soundonsound.com/sos/aug10/articles/sontronics-matrix-10.htm).

If you want something in matt black and with a much smaller footprint, I’ve just been reviewing the Latch Lake MicKing stands, which I have to say are utterly brilliant. However, they are also pretty expensive, because they are very well engineered, and imported from the US. The review is soon to appear in Sound On Sound, but these stands have a sensibly massive counterweight on the boom arm, a very heavy, but compact, base (with transport wheels to make it easy to move the stand to a storage area), a nice drop-arm system, and really ingenious lever locks and clutches that are adjustable for both tension and ease of use. These are very solid and impressive stands and well worth the investment, in my view.

Published in SOS April 2012

Q. How can I make using headphones less fatiguing?

Sound Advice : Mixing

I have been making music for years now, and although I have a set of Genelec 8040s that I use during the day (when I’m home), I have been using a set of Audio-Technica M50 headphones for writing at night, when I usually have the ideas and desire to write, but am unable to, due to neighbours and a sleeping wife.However, lately I have been unable to use the cans, as I’ve been experiencing discomfort and what I believe is the onset or warning signs of tinnitus. It’s been a nightmare trying to adapt to not using cans at night, and I find it almost impossible to get anything other than sequencing done at this low volume!I’m wondering whether there are any miracle headphones or bits of kit that would minimise hearing damage or discomfort while still being (relatively) accurate and enjoyable to use.

Via SOS web site

 

SOS Technical Editor Hugh Robjohns replies: 

Firstly, regarding the tinnitus: it’s very common, often temporary and may be nothing to worry about. It can be brought on by something as simple as drinking too much coffee or suffering a mild ear infection, but don’t ignore or neglect it. Go and see a medical professional and get checked out! If there is a problem, early intervention could make all the difference.

I don’t think there are any ‘miracle’ solutions in headphones. Basically, it comes down to self-control in establishing the most appropriate maximum level for those particular headphones and sticking to it. The simplest solution is to put a mark on the headphone volume control and exercise enough self-discipline to never turn it up past that. If you reach a stage in your mixing when you’re finding that maximum level is too quiet, take a break. Give the ears a little time to relax and reset, and then start again.

More volume is not the answer, though. It might seem more exciting and involving, but it doesn’t really help to make better mixes — in fact, it usually makes them worse! The reason is that greater volume allows you to hear through a bad mix more easily, and poor balances aren’t perceived as such. Working at more moderate levels — the kind of volume that most end listeners will use — encourages a far more critical approach to the mix, as poor balances sound obviously awful! Mixing becomes much harder, certainly, but also much more accurate and with far better end results. This is true of both speakers and headphones.

By all means turn the volume up if you need to check low-level background noises and so on, but do so only briefly. Try to mix at a modest level, and keep that level fixed. If you continually change your monitoring level, your mix will change continually too!

However, the fatigue you’re experiencing may involve more than just sheer volume. The M50s are pretty good for the money, but I think you might find it easier to work with a pair of good open-backed headphones that are more revealing. You might find it helpful to read the comments and suggestions for different models in a headphone comparison article we ran in the January 2010 issue (www.soundonsound.com/sos/jan10/articles/studioheadphones.htm). If possible, try different models before buying, to make sure the weight, headband pressure and size of the ear cups suit your head and are comfortable. Open-back headphones do ‘leak’ more sound than closed headphones, though, and that may be an issue for your wife!

The M50, being a closed-back design, tends to be less revealing of mid-range detail than a good open-backed headphone, and a consequence of this is a natural tendency to keep cranking the level to try to hear further into the mix, but more volume still doesn’t quite reveal what you want to hear! Headphones that exert a strong pressure on the sides of the head can also add to the sense of physical fatigue, and the sealed nature of the earpieces quickly makes your ears hot and uncomfortable, which also doesn’t help.

I’d recommend trying some good open-back headphones, like the AKG 702s, Sennheiser HD650s or the Beyerdynamic DT880 Pros. They are expensive, but I think you’ll find it far easier to mix with them and you’ll be much less tempted to wind the level up, although it is still very important to take frequent breaks to allow your perception of volume to reset! Headphones of this calibre provide a top-notch monitoring system that will last for decades if well looked after, and you’ll probably hear all sorts of details that your Genelecs don’t reveal, too.

If you find that your closed-back headphones are quite fatiguing, it may be a good idea to try some open-back headphones, such as these AKG 702s. Decent open-back headphones are often more revealing than closed-back models, and may therefore reduce the temptation to increase the volume.

Obviously, though, there is no physical sensation from the low frequencies when using headphones, as there is when using speakers and that can also be a factor in the continual desire to turn the level up, especially if you’re producing music that demands strong bass content. The only way around that is self-discipline and learning to trust your headphones.

As a last resort, if you don’t think you have the self-discipline to leave the volume control alone, it might be wise to consider investing in a suitably calibrated headphone limiter. Again, it’s an expensive option, but I’d suggest that it’s well worth it to protect your priceless ears! There’s some useful background information here: www.tonywoolf.co.uk/hp-limiters.htm. Also, Canford Audio offer various types of headphone level limiter that can be installed inside headphones or wired into the cable. These are based on a clever BBC design, which is now mandatory within the corporation to ensure that BBC staff don’t expose themselves to excessive SPLs through their phones, and it works extremely well. You can read more about it here: www.canford.co.uk/technical/PDFs/EarphoneLimiters.pdf.  

Published in SOS March 2012

Side Chain Compression Emulation Sony Acid Pro and Vegas Pro

Q. How can I use a figure‑of‑eight mic with the mid/side miking setup?

Sound Advice : Miking

I have two figure‑of‑eight Golden Age ribbon mics that I want to use as overheads for drum recording. I’ve read about Blumlein pairs and will try that, but I also wondered if I could try Mid/Side recording techniques. Can you use a figure‑of‑eight mic for the centre mic in that setup, and, if so, how do I get rid of the ‘rear’ sound from the centre mic? While I’m at it, can I ask if you know of any other neat recording tricks for using two figure‑of‑eight mics together?

Connie Buck via email

SOS Technical Editor Hugh Robjohns replies: 

Yes, you can certainly use the M/S approach if you want to, and that does provide the potentially useful advantage of being able to adjust the stereo recording angle remotely to set the required image width.

Recording a vocalist with two figure‑of‑eight mics can produce very good results. If the performer wants to play and sing at the same time, careful positioning to reject the unwanted sound from both mics (the vocals from the guitar mic and the guitar from the vocal mic) can achieve excellent separation.

However, the left‑right decoded signal from an M/S array comprising two figure-of-eights is essentially two figure-of-eights in an X-Y format: basically the same Blumlein array you are already familiar with. Altering the ratio of Mid and Side changes the equivalent mutual angle of the decoded X-Y mics, and distorts their polar patterns slightly. However, for matched Mid and Side levels, M/S with a pair of figure-of-eights decodes as a perfect Blumlein array. Indeed, this is precisely what Blumlein discovered and experimented with 80 years ago!

If you need to ‘get rid’ of the rear pickup of the Mid mic, you will have to place an acoustic absorber behind the mic; for example, the infamous SOS duvet, foam absorbers, or even some kind of reflection filter: anything to capture sounds that would otherwise head back into that rear pickup zone.

As for other neat tricks with dual figure-of-eights, there is a technique called the Faulkner Array that uses two figure‑of‑eight mics spaced about eight inches apart and facing forward. The idea is to capture a normal stereo sound‑stage in much the same way as an ORTF arrangement, but with significantly reduced sensitivity to reverberant sounds from the sides and above. It was a technique devised to deal with the acoustics of a church that had nasty side‑wall slapback issues.

Another situation in which I often use two figure-of-eights is capturing a singing guitarist. By careful placement and angling of the mics, it’s possible to arrange their deep side nulls to provide a significant amount of rejection of the unwanted source: the guitar mic rejects much of the voice, and the voice mic rejects much of the guitar. If you do this carefully (and assuming the guitarist can sit still and not sway about!), you can achieve 20dB of separation or more, which is a major improvement on the usual dual-cardioid approach!

You can read more about this at www.soundonsound.com/sos/1996_articles/dec96/singingguitars.htm.

Published in SOS September 2011

Korg All Access: On Tour With Scott Chesak and the All-American Rejects

Q. How do I record a double bass alongside other instruments?

Sound Advice : Recording

Having been a bass player for years, I’ve recently come into possession of an acoustic double bass. I seem to be getting a decent enough sound out of it that I think I’m ready to use it with my band. We’re going to be recording soon, but will all be playing together in the studio. How can I record the bass alongside other musicians, reducing as much spill as possible?

The ‘modern’ method of recording a double bass in the studio is to ‘bug’ it, often with a pickup fitted on the instrument’s bridge. Any ‘character’ lost in the sound is then usually EQ’d back in. However, the ‘vintage’ way would have been to use careful mic and instrument placement, in conjunction with carefully placed acoustic treatment, to provide a degree of separation.

Bradley Culshaw via email

SOS Technical Editor Hugh Robjohns replies: 

The obvious ‘modern’ solution is to fit a ‘bug’ — a bridge pickup or an internal mic — to the bass, which will provide a pretty high degree of separation. The sound character might not be entirely ‘natural’, but a little EQ should deal with that. The ‘vintage’ alternative is to use acoustic screens or gobos in the studio and thoughtful instrument and mic layout, with the aim of minimising spill and helping to provide some sound shadowing for mics, especially the double-bass mic, thus reducing the spill and providing a workable degree of separation from the other instruments playing in the studio. This is a well‑proven historic technique, and the remaining spill generally helps to gel the mix together and provide a great ‘live’ character to the mix. Of course, such spill makes it almost impossible to overdub replacement parts, but that’s what practice and an unlimited number of takes are for!

Published in SOS September 2011

KAOSSILATOR PRO & KP3 Comparison - In The Studio with Korg

Q. How much power does my stage system need?

Sound Advice : Mixing

I’m trying to work out how much power a PA system I work with draws, and I also need to come up with a sensible ‘plug‑it‑all‑in’ type of procedure. (I’ve read the Sound On Sound December ‘05 article ‘PA Basics’.) It’s mainly small venues we play in, such as function rooms and town halls. Looking at the manual for my Mackie SA1530z, I’m kind of baffled. It says:

Line Input Power Europe: 230V, 50Hz

Recommended Amperage Service: 16 amps

Is this saying that a 16‑amp circuit is recommended? The spec sheet doesn’t seem to list how much current the box will draw. Also, it’s often stated that FOH, mixer and racks, lights and backline should be powered from their own separate sockets (three in total). Is it acceptable to power from both sides of a double socket and another adjacent socket, therefore, all being powered from the same ring main?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

The 16‑amp thing looks like a generic suggestion to me. In the UK, standard domestic outlets are nominally 13A anyway!

Essentially, what they are saying is that it needs to be plugged into a sensible supply. The typical average current will be a few amps at most, but the initial inrush current on switch‑on will be considerably higher, so don’t try to turn everything on in one go!

If you need to know the real current and power‑consumption figures, invest in something like an energy monitor, such as the one I’ve found here: www.maplin.co.uk/plug-in-mains-power-and-energy-monitor-38343. This one is marketed by Maplin in the UK, but I’m sure you’ll find similar devices from all the usual suppliers. You simply plug in the device you want to know about, and the display will give you the current and power being consumed, as well as the supply voltage and frequency. It’s a really handy device and I use mine a lot when testing and checking equipment.

Regarding the use of wall sockets, assuming that you’re working with a PA and backline system that is consuming less than about 4kW in total (which would be most systems for a modest‑sized venue), use a double socket to run all the audio equipment. That minimises any problems with ground loops

If you need to know how much current your setup is using, a simple energy monitor like this should do the trick: plug in whatever you’d like to measure and its power consumption will be displayed.

Run all the backline from one side of the double outlet, and all the PA (FOH, racks, PA and monitors, for example) from the other side. Supplying the two systems from their own RCDs (Residual Current Devices) is essential too, particularly from the point of view of preventing a backline fault from taking out the PA. If the musicians want to use their own RCDs for their gear, that’s fine too!

Running the FOH on a long mains extension from the PA power‑supply socket (or distribution board) continues the theme of ‘star grounding’ and will minimise the potential for ground loops in the PA system. Run lighting from a different socket (or sockets) and try to keep the dimmer racks and cabling well away from the audio cables.

Published in SOS October 2011

KORG iMS-20 sings

Q. Are there any panning rules for maintaining mono compatibility?

Sound Advice : Mixing

With regard to stereo image width, is there typically a ‘cap’ you would place on tracks to maintain a good mono sound? Perhaps there’s some kind of relatively hard‑and‑fast ground rule (assuming a typical sort of track layout), such as ‘never go beyond 50 percent either way’?

Via SOS web site

SOS contributor Mike Senior replies: 

There are two basic issues regarding mono compatibility. The first is that panning any mono track off‑centre reduces its level in the mono balance by a maximum of around 3dB when panning hard left or right. From this perspective, the only ground rule I’d apply there is to make sure that the balance continues to function correctly in mono. If your main guitar power‑riff is panned hard left, it may struggle to fulfill its musical function in mono, simply by virtue of losing a lot of ground against things like the bass, kick, snare and lead vocal (which all typically reside close to the centre).

The second issue to be aware of is that any stereo recording or stereo effect return in your mix may contain elements in one channel that are out of phase or polarity-inverted compared to the other channel. These can phase‑cancel when summed to mono, and although this might simply result in a subjective level drop (as in the case of some M/S‑based widening effects), typically the cancellation is frequency‑selective in some way, so the tone of affected parts suffers as well. Stereo drum-overhead mics and stereo piano recordings commonly fall foul of this to some extent, on account of the widespread use of spaced‑pair recording techniques on these instruments, but almost any multi‑miked part can potentially come a cropper if you pan the individual mics independently in the stereo field

Flux Audio’s Stereo Tool is the pick of the freeware stereo vectorscope displays currently available, and can help you head off mono-compatibility problems, especially if you’re working on headphones.

The cast‑iron remedy to uncertainty here is to make a point of comparing your mix against commercial productions in mono. Conventions on stereo imaging vary a lot between styles, and even between engineers, so it’s tricky to generalise with any validity. However, what may help you is to get hold of a stereo vectorscope display for your DAW, such as Flux Audio’s fantastic freeware Stereo Tool plug‑in. Once you get used to how things look on there, it can tip you off to impending mono phase‑cancellation problems, especially if you’re working on headphones, which don’t give the same funny ‘outside the speakers’ stereo effect that’s usually a clear warning sign on nearfields.

All that said, there is one little panning‑width rule of thumb that I do tend to follow personally, but this isn’t as much related to mono compatibility as it is headphone listening. When you pan something hard to either side in headphones, it gives the impression that it’s right by that ear, because there’s no crosstalk between that earcup and the opposite ear. I’ve always found this a bit distracting myself, and it can make it tricky to blend the sounds in your mix convincingly, in my experience. For this reason I rarely pan mono sources beyond about 85 percent either way, because this makes them a little less dislocated in headphones and actually affects the stereo presentation very little, especially if you’re feeding a selection of stereo effect returns into your mix anyway, which will still guarantee that the stereo picture is painted right out to the edges. Bear in mind, though, that this is very much an issue of personal preference, and there are lots of very famous engineers who actively prefer the extreme‑panned presentation. The only way to make up your own mind is, again, to compare your mix to your favourite records on headphones and decide which sounds best to you.

Published in SOS October 2011