Q. Can I use two completely different mics in an M/S array?

I have a vintage Neumann U87 mic, a GAP Pre73 preamp and also a Zoom H4N recorder (massive contrast, I know). If I want to, can I use these in a Mid/Side setup with the Neumann as the Mid mic and the H4N for the Side? Or should I use the U87 in a figure-of-eight polar pattern for the Side and another mic — my Shure SM7B or Rode M3 — as the Mid? How, then, would I combine the two into a Mid/Side audio file?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: Using such radically different mics to form an M/S array is never going to give great results; the stereo imaging will tend to be rather variable and frequency dependent because of their mismatched frequency responses.

However, for experimental purposes I'd suggest using the U87 in figure-of-eight mode for the Side mic and either the SM7 or M3 as the Mid mic. I suspect you'll still find quite a lot of image instability, but it should work well enough to give you a feel for the versatility and practicalities of the M/S mic technique.
All you need to do to create a Mid/Side file is allocate the Mid mic to channel one and the Side mic to channel two, and record as a normal stereo file.

Regarding rigging the mics, the normal convention is that the in-phase side (front) of the figure-of-eight mic — the U87, say — faces to the left of the sound stage (as viewed from behind the mic array, looking towards the sound sources).

 
Although our reader may never achieve great results using vastly different mics in an M/S setup, it may still be worth him experimenting with his U87 in figure-of-eight mode as a Side mic and his Rode M3 as a Mid mic.
 
You should also set the preamp channel gains to provide a similar sensitivity for both mics, to optimise the signal-to-noise ratio and make the decoder work properly. The easiest way to do that is to decide how wide you want the sound stage to be — decide where the notional edges of the stereo pickup area are — and then get an assistant to stand at the nominal left-hand edge and make a fairly constant noise, such as humming. Assuming the Mid mic output is already providing an appropriate level (if not, adjust that first), set the gain of the Side mic's preamp to give an approximately equal output level to that provided by the Mid mic's preamp.

 Question of the month

Published March 2012

Pa700 Video Manual Part 4: Song Play

Q. How can I make using headphones less fatiguing?

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 (/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 March 2012

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

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 /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 April 2012

Pa700 Video Manual Part 3: Styles

Q. What are auxes, sends and returns?

Excuse the simplicity of the question, but I'm always coming across these terms in the magazine, and I don't know what they are: auxes, buses, sends and returns. Can you explain to me what are? Are they all part of the same thing or completely unrelated?
Tony Robbins via email
 
The aux sends on a mixer (whether hardware or software) allow you to send independent mixes to performers on stage or in the studio. You can also use them to feed effects processors at mixdown.

SOS contributor Mike Senior replies: All of these terms are related, in that they are all ways of talking about the routing and processing of audio signals. The word 'bus' is probably the best one to start with, because it's the most general: a bus is the term that describes any kind of audio conduit that allows a selection of different signals to be routed/processed together. You feed the desired signals to the bus, apply processing to the resulting mixed signal (if you want), and then feed the signal on to your choice of destination. If that description seems a bit vague, that's because buses are very general‑purpose.

For example, it's common in mixing situations to hear the term 'mix bus', which is usually applied to the DAW's output channel. In this case, all the sounds in your mix are feeding the bus, and it might then have some compression applied to it before the sound is routed to a master recorder or recorded directly to disk within the software. A 'drums bus', on the other hand, would tend to refer to a mixer channel that collects together all the drum‑mic signals for overall processing, routing them back to the mix bus alongside all the other instruments in the arrangement. Other buses are much simpler, such as those that can be found on a large‑scale recording mixer, feeding the inputs of the multitrack recorder, or those which carry audio to/from external processing equipment. Some don't even provide a level control.

An 'aux' is just a type of bus that you use to create 'auxiliary' mixes alongside that of the main mix bus: each mixer channel will have a level control that sets how much signal is fed to the aux bus in question. What you do with your aux buses is up to you: the most common uses are feeding a cue signal to speakers or headphones, so that performers can hear what they're doing on stage or during recording; and sending signals to effects processors during mixing. In the latter case, the aux bus that feeds the effects processor is usually referred to as a 'send', while the mixer channel that receives the effect processor's output will usually be called the 'return'. For more information, check out Paul White's 'Plug‑in Plumbing' feature back in SOS April 2002; you can find it at /sos/feb02/articles/plugins.asp.

Published September 2011

Pa700 Video Manual Part 2: Sounds

Q. What is side‑chaining, and what do you use it for?

 This might be a very big topic, but I'm hoping that you can help to clear up some confusion. Side‑chaining seems to be something that is used a lot, but I don't really understand what it is. Can you explain?

Kim Nguyen via email

 
Normally, compressors and gates use the signal that's being processed to control the amount of gain reduction taking place, as in the top arrangement in the diagram to the right. Some devices, however, allow you to use a secondary input to control the gain of the first input (below). This allows you to, for example, compress a bass guitar using a kick drum as the trigger, or 'side‑chain' input.

 

SOS Reviews Editor Matt Houghton replies: This is a huge topic, and it would be well worth reading some of the past SOS features about it (see the archive on our web site). Essentially, though, any dynamics processor (for example, a gate, expander, compressor or limiter) uses two input signals: the incoming audio itself and a side‑chain, which feeds the detection circuitry that determines whether or not the processor acts on the material. Simple processors take their side‑chain signal directly from the audio input. A more sophisticated approach is to split that signal, and allow you to process the side‑chain with high‑ or low‑pass filters.

Many professional devices also have a second physical input called the external side chain, so that you can feed the processor's detection circuit with any audio signal, which can be totally unrelated to the main audio input. A common example is ducking, where you might feed the kick‑drum signal into a compressor on the electric bass and set it up with a fast attack and release time so that the bass is attenuated by 1‑2 dB every time the kick exceeds the threshold. Another example would be to use a signal to 'key' a gate: you could place a gate on a synth pad, for example, and use a percussive loop to make the gate open and close rhythmically with the groove of the loop, without ever needing to hear the loop itself!

Published September 2011

Pa700 Video Manual Part 1: Introduction, Setup & Navigation

Q. Should I EQ my drum recordings?

When recording a drum part, do I have to EQ it, or should it be left untreated?

 
There's no hard-and-fast rule when it comes to treating your drum recordings (or not!). If you get a really good-quality source recording, you may not need to EQ it at all.

Samuel Am, via email

SOS Reviews Editor Matt Houghton replies: There's no right or wrong answer to this question! As long as the drum sounds right to you when played through your studio monitors, use whatever you need to get the result you want: compression, EQ, Transient Designer or, in the case of a snare drum, perhaps even a tiny bit of distortion.

Back in the days of big-budget studio recordings, many engineers would — if they felt it was needed — EQ and compress drum mics on the mixing console while recording, and then print the results to multi-track tape for mixing. Today, it's easy to capture a clean recording and add EQ or other processing to taste at the mixing stage, using plug-ins. This leaves more options open to you, and avoids you getting stuck with poor EQ and dynamics decisions that don't suit the mix, but it also means that you're putting off decisions, so the whole process can take longer.

If you're new to this, I'd recommend experimenting first with mic choice and placement to get the best sound possible. Then add EQ and/or compression during the mix stage if you feel it's needed for the particular track in question. In the long run, you'll end up with better results if you start working in this way, and eventually your EQ and dynamics decisions will become second nature, at which point it will be easy to make them during recording rather than mixing.

Of course, depending on the style of the song, it's also perfectly possible to record a natural-sounding drum part with no processing whatsoever, just relying on mic choice and placement (and a good-sounding drum and room, and a good drummer!) to capture a clean sound.

As for which mics to pick, it really depends on the drum(s) you're using: what you use on a kick or snare close-mic can be very different from what you use for overheads or hand-drums.

Published April 2012

KROSS 2 Video Manual Part 8: Global and Media Mode Functions

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

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 /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 April 2012

KROSS 2 Video Manual Part 7: Recording and Playing Audio

Q. Should I buy a stand-alone master clock?

I'd like to know what the advantages and disadvantages are of using stand-alone word clock units, like Apogee, Lynx, Antelope, Mytek and so on, versus the old built-in word clock in a TC Konnekt Studio 48. I don't need many sockets (up to six, maybe) and I'm OK with daisy-chaining my gear as I do now, but would a separate word clock have many advantages over what I have now? I can put around £400 to £500 aside to buy something, if it's worth it. Via SOS web site

SOS Technical Editor Hugh Robjohns replies: If it ain't broke, why fix it? As a general point, running separately buffered clock feeds from a clock distribution unit is technically better (in terms of jitter and overall timing precision) than the daisy-chain technique. However, there's nothing fundamentally wrong with daisy-chaining either. And if it's working reliably now, there's no obvious need to change anything.

In general, converters (A-D and D-A) work better and deliver better technical performance if they run from their own internal clocks. Almost without exception, the measurable performance of most converters driven from external clocks is degraded, and the best you can hope for is that the degradation is negligible or minimal. Devices that process and pass only digital signals are not particularly critical of the clocking arrangements and quality is totally unaffected by moderate clock jitter.

So my usual recommendation is to use the internal clock of your 'master' A-D converter as the system's master clock, and distribute that via a dedicated clock distribution unit. The Drawmer D-Clock provides good value for money, for example.

  
A master clock may well become necessary if working with external video machines because of the need to synchronise video and word clock. In this case, a good-value option would be the Mutec Iclock, shown here.

If you're working with external video machines, then a master clock usually becomes a necessity because of the need to synchronise video and word clock, and in that situation I think the best value for money comes from something like the Mutec Iclock or Audio Design SynchroGenius. For the very few audio-only installations where a master clock is beneficial for practical reasons then, again, the Drawmer M-Clock boxes provide excellent value for money.

As I demonstrated in the article 'Does Your Studio Need A Digital Master Clock?' [go to /sos/jun10/articles/masterclocks.htm for the full article], the more expensive options like the Big Ben and the Antelope offered no detectable advantages in terms of audio quality, and few, if any, facilities that aren't available elsewhere for less.

 If I were you, I'd invest that money in something else that would make a real, practical and tangible difference to your music-making activities.

Published December 2011

KROSS 2 Video Manual Part 6: Recording a MIDI Sequence

KROSS 2 Video Manual Part 5: Performing With The Pad Sampler

Q. Is asymmetry in monitors a problem?

I'm considering AVI's Pro Nine Plus system for my main nearfield monitoring, partly based on the great review they got from Paul White back in September 2005 (/sos/sep05/articles/avipronine.htm). Should I be worried about the fact that the tweeter mounting and porting are both asymmetrical in relation to the main driver? Will this cause slight delay/phasing issues when placed in the classic equilateral-triangle stereo setup?

 
AVI's Pro Nine Plus monitors (left) have their tweeter mounting and porting asymmetrical to the main driver. This is actually relatively common, as you can see from the Acoustic Energy AE22s and the Dynaudio BM15As (below).
 

Via SOS web site

SOS contributor Mike Senior replies: Opinions differ about a lot of aspects of speaker design, as you can easily see even just from comparing the external appearance of a selection of similarly priced monitors. One such moot point is how important symmetrical driver placement is, and the Pro Nines are by no means the only speakers that have their tweeters skewed to one side like this. The Acoutic Energy AE22s and Dynaudio BM15As both feature this kind of setup, and are both nonetheless well-regarded. Although I've not tried these specific speakers myself, the main thing I'd be wary of in principle is that the size of the stereo sweet spot may be reduced. No matter which way you move your head (forward/back, side to side, or up/down) the potential for inter-driver phasing in the mid-range appears to me to be greater than with a more traditional vertically stacked driver configuration. Even if this theoretical concern is borne out in practice, though, the real question is how much it'll matter to you. If you're happy to stay in the sweet spot most of the time, and can check the mid-range balance with a single-driver speaker such as an Auratone (or similar), then it may not be a huge practical concern. Personally, I'd say that if you like the speakers otherwise, don't let the asymmetry be a deal-breaker.

As for the ports, again I don't think their asymmetry should really put you off, and although I find that porting in budget-level monitors can cause all sorts of low-end monitoring problems, I would imagine that these speakers are probably getting into the kind of price range where the potential problematic side-effects of the porting are kept well enough under control that you can work with them for mixing purposes. Certainly, the 90Hz low-end boundary on the published frequency-response figure leads me to suspect that the port hasn't been overhyped, as it seems to be on many budget models, and that counts for a lot in terms of accuracy.

Published February 2012

KROSS 2 Video Manual Part 4: Using The Performance Tools

Q. Do mixes benefit from low-pass filtering at mixdown?

I've heard a lot about high-pass filtering tracks to reduce clutter at mixdown, but not as much about low-pass filtering in this context. Would mixes suffer or benefit from doing the same at the opposite end? For example, would it be easier to bring out 'air' in a vocal if other parts were low-passed?

 Via SOS web site

SOS contributor Mike Senior replies: Particularly in small-studio environments where the low-frequency monitoring fidelity is questionable, there's a lot to be said for high-pass filtering in a fairly systematic way to head off problems at mixdown. However, widespread low-pass filtering offers fewer benefits, simply because so many instruments in a mix will have harmonics and noise components that extend right up the spectrum. In practice, I find peaking/shelving cuts are, therefore, more appropriate for dealing with typical mixdown tasks, such as frequency-masking problems. Yes, in theory you could make your lead vocal sound airier by low-pass filtering the other parts, but you'd still have to consider how the mix as a whole will sound during moments when the vocal isn't active, so achieving an airy vocal in practice isn't usually as simple as this.

 
Although fairly systematic high-pass filtering is very sensible in home-studio mixing, as you can see in this screenshot from a recent Mix Rescue project, it's rarely beneficial to apply low-pass filtering in a similar way.

Having said that, there's nothing wrong with low-pass filtering if you really want to kill the high frequencies of an instrument for balancing reasons. I would most commonly do this with amped instruments, such as electric guitars, which are capable of contributing a lot of undesirable amplifier noise in the top two octaves of the audible spectrum. However, this has to be evaluated on a case-by-case basis, because it's very easy to dull the overall mix if you're not careful.


Published January 2012

KROSS 2 Video Manual Part 3: Combining Sounds and Using Favorites

Q. Is flutter echo a problem in a well-treated room?

My daughter managed to play a tough piece she's been practising on the keyboard this weekend. She played it so well that we clapped our hands... then we noticed how strange the clapping sounded. It rang on but died very quickly, and for the time it rang on, it sounded very metallic and almost robotic.That was close to the middle of the room. The room is partially treated at the moment, with panels at the side-wall reflection points and ceiling, one on the ceiling, and three corner superchunks. I tried clapping again with some further panels on the side walls directly to the left and right of where I was sitting, and the noise disappeared. I understand enough to realise the sound is the clap bouncing back and forth between the two walls, and I'm guessing that this is what folk refer to as flutter echo. What I'm a little less sure about is whether it is a problem, and what — generally — a hand clap should sound like in a well-treated room.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: If we're talking about the sound in a control room, the point is what the room sounds like when listening to sound from the monitor speakers. It is conceivable that, by design (or coincidence), the acoustics could well sound spot on for sounds from the speakers, but less accurate or flattering for sources elsewhere. And, unless you're planning on recording sources in the control room at the position you were clapping your hands, those flutter echoes might not represent a problem or require 'fixing'.

However, in general, strong flutter echoes are rarely a good thing to have in a control room and I'd certainly be thinking about putting up some absorption or diffusion on those bare walls to prevent such blatant flutter echoes.

 
Flutter echoes in a studio can be distracting and fatiguing, so it's often worth putting up some absorbent foam on bare walls to reduce them. Don't overdo it, though: you need to maintain a balanced acoustic.

You shouldn't go overboard with the room treatment, though, because while working in a control room that has 'ringy' flutter echoes or an ultra-live acoustic can be very distracting and fatiguing, so too is trying to work in a room that sounds nearly as dead as an anechoic chamber!

Of course, traditional control rooms are pretty dead, acoustically speaking, and that is necessary so that you can hear what you are doing in a mix without the room effects dominating things. But the key is to maintain a balanced acoustic character across the entire frequency spectrum. The temptation in your situation might simply be to stick a load of acoustic absorbers on the walls, and that would almost certainly kill the flutter echoes, but in doing so there is also a risk that you'd end up with too much HF and mid-range absorption in the room (relative to the bass-end absorption).

That situation would tend to make the room sound boxy, coloured and unbalanced, and that's why a better alternative, sometimes, is to use diffusion rather than absorption; to scatter the reflections rather than absorb them. The end result is the same, in that the flutter echoes are removed, but the diffusion approach keeps more mid-range and HF sound energy in the room.

The question of which approach to use — diffusion or absorption (or even a bit of both) — depends on how the rest of the room sounds, but from your description I'd say you still had quite a way to go with absorption before you've gone too far.

To sum up, I'd suggest that you're not worrying unnecessarily, and that it would help to put up some treatment to reduce those flutter echoes.

Published February 2012

KROSS 2 Video Manual Part 2: Selecting & Playing Sounds

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

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 September 2011

KROSS 2 Video Manual Part 1: Navigating The Interface

Q. What is side‑chaining, and what do you use it for?

This might be a very big topic, but I'm hoping that you can help to clear up some confusion. Side‑chaining seems to be something that is used a lot, but I don't really understand what it is. Can you explain?

Kim Nguyen via email


 
Normally, compressors and gates use the signal that's being processed to control the amount of gain reduction taking place, as in the top arrangement in the diagram to the right. Some devices, however, allow you to use a secondary input to control the gain of the first input (below). This allows you to, for example, compress a bass guitar using a kick drum as the trigger, or 'side‑chain' input.

 

SOS Reviews Editor Matt Houghton replies: This is a huge topic, and it would be well worth reading some of the past SOS features about it (see the archive on our web site). Essentially, though, any dynamics processor (for example, a gate, expander, compressor or limiter) uses two input signals: the incoming audio itself and a side‑chain, which feeds the detection circuitry that determines whether or not the processor acts on the material. Simple processors take their side‑chain signal directly from the audio input. A more sophisticated approach is to split that signal, and allow you to process the side‑chain with high‑ or low‑pass filters.

Many professional devices also have a second physical input called the external side chain, so that you can feed the processor's detection circuit with any audio signal, which can be totally unrelated to the main audio input. A common example is ducking, where you might feed the kick‑drum signal into a compressor on the electric bass and set it up with a fast attack and release time so that the bass is attenuated by 1‑2 dB every time the kick exceeds the threshold. Another example would be to use a signal to 'key' a gate: you could place a gate on a synth pad, for example, and use a percussive loop to make the gate open and close rhythmically with the groove of the loop, without ever needing to hear the loop itself!

Published September 2011

KORG Pa1000: more power - new arranger keyboard from KORG

Q. What are auxes, sends and returns?

Excuse the simplicity of the question, but I'm always coming across these terms in the magazine, and I don't know what they are: auxes, buses, sends and returns. Can you explain to me what are? Are they all part of the same thing or completely unrelated?

Tony Robbins via email

 
The aux sends on a mixer (whether hardware or software) allow you to send independent mixes to performers on stage or in the studio. You can also use them to feed effects processors at mixdown.

SOS contributor Mike Senior replies: All of these terms are related, in that they are all ways of talking about the routing and processing of audio signals. The word 'bus' is probably the best one to start with, because it's the most general: a bus is the term that describes any kind of audio conduit that allows a selection of different signals to be routed/processed together. You feed the desired signals to the bus, apply processing to the resulting mixed signal (if you want), and then feed the signal on to your choice of destination. If that description seems a bit vague, that's because buses are very general‑purpose.

For example, it's common in mixing situations to hear the term 'mix bus', which is usually applied to the DAW's output channel. In this case, all the sounds in your mix are feeding the bus, and it might then have some compression applied to it before the sound is routed to a master recorder or recorded directly to disk within the software. A 'drums bus', on the other hand, would tend to refer to a mixer channel that collects together all the drum‑mic signals for overall processing, routing them back to the mix bus alongside all the other instruments in the arrangement. Other buses are much simpler, such as those that can be found on a large‑scale recording mixer, feeding the inputs of the multitrack recorder, or those which carry audio to/from external processing equipment. Some don't even provide a level control.

An 'aux' is just a type of bus that you use to create 'auxiliary' mixes alongside that of the main mix bus: each mixer channel will have a level control that sets how much signal is fed to the aux bus in question. What you do with your aux buses is up to you: the most common uses are feeding a cue signal to speakers or headphones, so that performers can hear what they're doing on stage or during recording; and sending signals to effects processors during mixing. In the latter case, the aux bus that feeds the effects processor is usually referred to as a 'send', while the mixer channel that receives the effect processor's output will usually be called the 'return'. For more information, check out Paul White's 'Plug‑in Plumbing' feature back in SOS April 2002; you can find it at /sos/feb02/articles/plugins.asp.

Published September 2011

Q. How much power does my stage system need?

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 October 2011

KORG Pa1000: more power - new arranger keyboard from KORG

Q. How can I connect hardware synths to my setup?

Currently, I have a MIDI keyboard, a Mackie Spike audio interface, an Apogee Duet interface, a UA Solo 610 preamp and a Neumann TLM103 mic. I use the Spike as a soundcard and run my MIDI through it, and the Duet for recording vocals.

I'm looking to get some hardware synths in the near future and need some advice. In preparation for the synths, I've bought a MOTU Express 128 so that I can have up to eight synths at once hooked up for MIDI. As both the Spike and Duet only have two audio inputs each, I am also looking to do away with those and get a better audio interface. However, if I get rid of them, I do not have a soundcard to produce sound via my monitors.

This is where I'm getting confused. How do I set up, say, three hardware synths via audio and MIDI (I believe you need both connected to get sound in your DAW?) and also get sound from my monitors out of my DAW? Can I get an audio interface that I can record vocals through and plug hardware synths into?

Via SOS web site

SOS Editor In Chief Paul White replies: You have a couple of options, one of which is to use an external analogue mixer to combine the output of your DAW (stereo) with your hardware synths. When the mix is sounding right, you record the output of the mixer back via your audio interface onto a new stereo track, but with the playback fader turned down during recording so the signal doesn't feed back on itself. Speaker and headphone monitoring would be done from the output of the mixer. I used to work in this way and got really good‑sounding results.

The other option is to buy an interface with plenty of spare inputs, ideally one that can be further expanded using an ADAT‑compatible preamp. MOTU's interfaces are generally reliable and straightforward (most include volume controls for your monitors) and I've also used M‑Audio with no problems. Expanders are available from under , such as Behringer's ADA8000, which will give you eight more inputs if you need more. You'd then connect your synths up to pairs of inputs (for stereo) and record their outputs just as you'd record any other audio. Most DAWs now have the ability to set up live inputs in permanent monitor mode, so you can always hear them even when they're not set to Record Ready. Working in this way, each synth would have both a MIDI track to control it and a stereo audio track to record it.

  
Expanding the number of inputs in your setup can be done at a relatively low cost. This Behringer ADA8000 can be found for well under £200$250 and will give you an extra eight inputs to play with.

The advantage of working like this, rather than using an external mixer, is that you can apply plug‑ins to the synth channels if you need more effects. You can also come back to your mixes years later when the synths have been disconnected or sold.

The MOTU multi‑port MIDI interface will enable you to handle up to eight multitimbral synths at once without running out of MIDI channels, so that seems a practical choice.

Published October 2011

KRONOS LS artists first impressions

Q. Should I mix an album as I’m writing it, or all at once?

I'm in the long process of trying to write enough material to put a cohesive, album-length bunch of stuff together. I have a few ideas in 'semi-baked' state, and have got to the point where I have one track written, structured and recorded, and am ready to make a proper mix (I've already made a rough mix).My decision now is whether to go to town on mixing that one track, and then get on with the rest of the writing and recording at a later date, or to keep it at the rough mix stage, finish the rest of the material, then mix the whole lot afterwards.I'm guessing the second approach would lead to greater overall consistency, but this is my first real stab at 'doing an album', if you want to call it that. My output up to now has been rather discontinuous, so it hasn't mattered before.What approach would you take, and how do you think it could help your progress?

Via SOS web site

SOS Reviews Editor Matt Houghton replies: Consistency is great if it's consistently good. Otherwise it's not such a laudable aim! There's no harm in still writing and recording stuff while you're mixing other stuff, but I would rather mix one track at a time, so that any lessons I learn can be applied to the next mix, and so on.
Also, bear in mind that, while mixing the first or second tracks, you might have one of those dawning "Oh, that would have been so much easier if only I'd recorded it like that!” moments, and that would be a bugger if you'd already tracked everything else.
 
There's no particular reason not to continue writing while you're mixing other tracks, but it makes sense to complete a couple of mixes before getting stuck into the rest of a project if you're, say, recording an album. This means that you can apply what you've learnt from your first mix(es) to the rest of the material. It also means that any recording issues you pick up during the mixing stage won't appear in all tracks.

SOS contributor Mike Senior adds: I'd second Matt on that one. It may mean that you end up redoing the first couple of mixes with the benefit of hindsight, but I think, overall, it's probably the best option if you're still feeling your way though a little bit with the mixing side of things.

It's no different from when you're mixing anything: you have to reference your work against any other material you want consistency with. Often that will be commercial releases with which you want your work to compete, but it can just as easily be other mixes you've done, which are destined for the same record. If you make sure to do that, then everything else should sort itself out in the long run.

I do tend to keep the main send effects I used for the first mix available for the second if I'm working on several things for one artist, as long as those effects met with their approval first time round! That does help to give some conformity to the sound. However, there are perfectly valid aesthetic reasons for not wanting to make all the tracks sound the same, so you should still try to make each track shine on its own terms. If that means using completely different mixing strategies, then so be it.

Published November 2011