Q How do I record from my guitar amp’s headphone out?

Sound Advice : Recording



Hugh Robjohns



These are all jack-to-jack cables, but they’re not wired the same. To record from a headphone output to an audio interface’s input, you’ll usually need the bottom one, often known as a ‘Y–cord’ or ‘insert cable’.These are all jack-to-jack cables, but they’re not wired the same. To record from a headphone output to an audio interface’s input, you’ll usually need the bottom one, often known as a ‘Y–cord’ or ‘insert cable’.



I’m seeking some guidance on why I’m unable to record from my guitar amp straight to my audio interface. I’m working with two small amps, a Marshall and a Crate, and trying to record a feed from the guitar amps’ headphone outputs via a Focusrite Scarlett 2i2 interface. Originally, I thought that the headphone jacks on both amps might be broken, but when I plugged in a pair of Sony headphones, the sound came through perfectly clearly. Then I thought the issue might be that the headphone had a quarter–inch TRS connector instead of TS, so I just picked one of those up and tried to hook the amp up to the interface that way. No dice.



I’m not getting total silence, though — if I turn the amp up loud enough I get very crackly audio that sounds like it has a high–pass filter on it. Your guidance would be greatly appreciated!



SOS Forum post



SOS Technical Editor Hugh Robjohns replies: Although headphone amps aren’t technically the best output source for a recording, you should still capture something reasonable if using the correct cables. But the issue here is that the apparently similar quarter–inch sockets in the amp and interface are wired very differently, and therefore carry/expect differently formatted signals.



The headphone output is unbalanced and is wired to be compatible with stereo headphones, even though the guitar amp produces a mono signal. That means that the unbalanced amp output signal is wired to both the tip and ring contacts in the headphone socket, with a common ground on the sleeve. The interface input expects a balanced line–level signal, which means that it only responds to the difference between the signals on the tip and ring contacts.



With a TRS–TRS cable connecting the amp headphone output to the interface balanced line input, the signals on the interface tip and ring contracts are identical; there is no difference, and so there will appear to be no signal. All you’ll hear, as you describe, is the very small error signal resulting from an imperfectly balanced input amplifier, which is usually a very quiet hissy, spitty, toppy sound (see Figure 1).



With a TS–TS (instrument) cable connecting the amp’s headphone output to the interface balanced line input, there is no ring contact, and so the ring output terminal in the headphone socket is shorted directly to ground by the plug sleeve. Inside the amplifier, I suspect the headphone-socket tip and ring contacts are actually wired directly together (rather than having a true stereo output amplifier) since there is only a mono source. This means that the act of inserting a mono TS plug will actually short the entire headphone amp output to ground, leaving no signal to output to the interface at all! (See Figure 2.)



The only workable solution, if you want to record from the headphone output, is to use a ‘Y–cord’, which comprises a TRS plug at the amp end, and two TS plugs for the interface end. It’s often sold as an ‘insert breakout cable’ or a ‘stereo–to–dual–mono output splitter’ cable. When using this kind of breakout cable, the TRS plug provides the headphone socket tip signal on the tip of one TS plug, and the ring signal on the tip of the other TS plug. If you then plug one (or both) of these TS plugs into the balanced line input(s) of your interface, the balanced input circuitry looks for the difference between the signals on the tip and ring again, but this time the TS sleeve shorts the ring signal to ground, and the wanted headphone output signal is applied between the tip and ground, so all is well. (See Figure 3.) Hey presto! It will all work as you want and expect...    

Published in SOS July 2015

Korg Kronos Music Workstation Video Manual Part 1- Introduction and Navigation

Q How can I export a Pro Tools project for mixing in Reaper?

Sound Advice : Mixing



Matt Houghton



Moving imported WAV files to their time–stamped position in Cockos Reaper.Moving imported WAV files to their time–stamped position in Cockos Reaper.



My band tracked some songs at a local studio using their Pro Tools rig and I want to mix the tracks at home in Reaper. What’s the best way of getting the session data out of Pro Tools and into my Reaper system?



Dave Matthews via email



SOS Reviews Editor Matt Houghton replies: I feel your pain: we often go through this sort of thing with Mix Rescue projects! The best method really depends on the nature of the project. If it’s just a plain multitrack recording session then it should be a simple case of importing all the audio files to different tracks so that they all start at bar 1, beat 1. You’ll see at a glance which were from the same take as they’ll be the same length. Hopefully, the engineer labelled the tracks/files so you can see which sounds are which.



If you’ve done edits or punch–ins, that adds a layer of complexity, but is still fairly simple. First, ask the studio to ‘consolidate’ all the clips in Pro Tools, to create files that line up as described above. If that’s not an option (or the studio charges you too much for the privilege!) note that Pro Tools automatically time–stamps the WAV files it records and Reaper can read those time stamps. Drag and drop your files into Reaper, select all, right-click on a clip to bring up the context menu, and select Item Processing/Move Items To Source Preferred Position (BWF). All files should line up in the same positions as they were recorded in Pro Tools. Be aware, though, that some DAWs have a default timecode offset (for reasons relating to audio–to–picture applications that you needn’t understand for this task). So you might find, as I once did, that all your files start as if the bar 1 beat 1 position is at +1 hour on Reaper’s timeline. To remedy this, zoom right out to find the files, select all, and drag all files to your preferred starting point



If the Pro Tools session includes more information, such as pan settings, clip gain, volume automation and so on, you’ll need either to bite the bullet and redo all that work yourself, or use another method to transfer the files. Some of that information can be transferred as OMF/AAF files, but (a) I’ve found that these are unreliable and inconsistently implemented by different DAWs and (b) Reaper doesn’t support them! The best bet in this situation is to invest in Suite Spot Studios’ AA Translator software, which reads and writes more project data in a wider range of formats than any other software I know of. I have a copy here under review and have been most impressed so far. It’s Windows only, but with the manufacturer’s help I have it running in Winebottler on Mac OS 10.9.5.



The trickiest thing is transferring plug-in effect or instrument data from one DAW to another. Your best bet is to print those effects as audio. But you should also be able to get the studio to supply the MIDI data from the session to enable you to rebuild any instrument parts.    

Published in SOS July 2015

Korg's "What Would You Do For A Kronos" Contest Winner: Matt Griffo!

Korg microSTATION Video Manual: Conclusion: 7

Q. Does changing the phase of drums make a big difference?

Sound Advice : Recording




Mike Senior



Although polarity/phase adjustments at mixdown can radically affect the sound of a multimiked drum kit, they might make very little difference at all. The only way to find out is to check...Although polarity/phase adjustments at mixdown can radically affect the sound of a multimiked drum kit, they might make very little difference at all. The only way to find out is to check...



I’ve recently become interested in manipulating phase while mixing recorded drums to see what that would do, as I’ve not messed with that before! The drums in question were recorded onto seven channels with three non-matching mics: front, back and overhead, plus kick, snare and two tom mics. I downloaded the Sonalksis FreeG fader, which has a polarity reverse button, and also the UAD Little Labs IBP variable phase-manipulation plug-in. I tried about every variation I could come up with in terms of phase with the seven channels, starting with polarity flips and then slowly sweeping the 180-degree range with IBP. While there was some variation in sound as a result, it was minimal, and not necessarily better — just different. Certainly not dramatic and nothing that I ended up using. Is that unusual?



Tom Dyer, via email



SOS contributor Mike Senior replies: Not at all. Depending on the mic positions, spill levels, and balance between the drum mics (and whether they were gated during recording), phase can make anything from a massive to a miniscule difference. However, it’s very rare that it makes no discernible difference at all, so it’s always worth checking for the most appealing-sounding combination as a matter of course whenever you’re dealing with multi-miked recordings. Yes, the polarity/phase relationship you choose will be a subjective decision, but then subjective preferences are a lot of what mixing is about, after all!



I’d also add that a final mix is rarely constructed from a handful of huge ‘night and day’ sonic changes, but is almost always the result of hundreds of subtle little tweaks, so even when the effects of a single polarity/phase adjustment are fairly minimal, the cumulative effect of a such adjustments across multiple tracks may nonetheless add up to something more substantial.  

Published in SOS June 2015

Korg microSTATION Video Manual: SD Card Saving and Loading: 6

Q. How do I use two guitar amps with one speaker cabinet?

Sound Advice : Recording




Matt Houghton



There’s more to switching between different guitar speakers than a simple A/B switch, but these boxes from Radial and Palmer should make it easy.There’s more to switching between different guitar speakers than a simple A/B switch, but these boxes from Radial and Palmer should make it easy.



I have a friend who has a problem. He asked me if I had a solution but I’m not that clever, I’m afraid! He uses two different amps on stage, one a Marshall and the other a Hughes & Kettner, with a 2x12 and 4x12 cab, respectively. Is there any way he can just use one cab for both amps, changing amp at the flick of a switch? The music he plays is quite diverse and he swaps several times during a set. I have spoken to a couple of people and they started talking about dummy loads and exploding amps! Any advice would be appreciated.



Owen Armstrong, via email



Q How do I use two guitar amps with one speaker cabinet?



SOS Reviews Editor Matt Houghton replies: Assuming we’re talking about tube amps here, what you’ve heard about dummy loads is relevant: never switch on a tube amp without a load connected.



I only know of a couple of companies (though I’m sure there are others) who make products designed specifically to do what your friend requires. The first are Radial, whose HeadBone comes in three varieties: one that’s configured for switching between two solid-state amps, another for switching between two tube amps, and a third for switching between one of each. The other company are Palmer, who offer the Tino system, which can cope with any combination of tube and solid-state amps.



The thought occurs, though, that if portability is the problem, perhaps your friend could look at an alternative amp as well — a Kemper Profiling Amp, perhaps, with which he could ‘capture’ the sound of his existing setup?  

Published in SOS June 2015

Korg microSTATION Video Manual: Sequencer: 5b

Q. What is the ‘Ground Compensated’ output on my mixer?

Sound Advice : Mixing




Hugh Robjohns



A ground-compensated system works in exactly the same way as an impedance-balanced system when connected to a balanced input. However, when connected to an unbalanced input, any noise on the destination ground, due to a ground loop, is added to the send signal in compensation.A ground-compensated system works in exactly the same way as an impedance-balanced system when connected to a balanced input. However, when connected to an unbalanced input, any noise on the destination ground, due to a ground loop, is added to the send signal in compensation.



On my Soundcraft Ghost mixing console the aux outputs are listed as ‘Ground Compensated’ in the manual. I’ll be connecting these aux outputs to the balanced inputs of my effects gear, which is mostly on XLR and some on TRS. I’ve been using Rane Note 110: Sound System Interconnection as my guide but I can’t determine exactly the options for wiring up these so-called ‘Ground Compensated’ outputs. Do you have any suggestions?



Russell, via email



SOS Technical Editor Hugh Robjohns replies: The ‘ground-compensated’ or ‘GC’ output is a very clever circuit design dating from the 1980s, and Douglas Self, who was a designer at Soundcraft, explains it very well in his book Small Signal Audio Design (reviewed in SOS January 2015: http://sosm.ag/dself-ssad-2nd-ed). The ‘ground compensated’ output is actually a ‘single-sided’ output, but one cleverly designed to work equally well with both balanced and unbalanced destinations. However, its real strength is that it is superb at preventing ground-loop hum problems with unbalanced destinations.



As far as the cable wiring for your balanced destinations is concerned, you simply wire the cables in exactly the same way as for any normal balanced configuration. XLR pin 2 (hot) goes to XLR pin 2 or TRS tip; XLR pin 3 (cold) goes to XLR pin 3 or TRS ring; and XLR pin 1 (screen) goes to XLR pin 1 or TRS sleeve.



If you’re interested in the electronic technicalities, you’ll know that a conventional balanced output provides half the signal on the ‘hot’ connection and the other half, polarity-inverted, on the ‘cold’ connection. As a differential input looks for the voltage difference between the hot and cold lines, it sees and extracts the full-level signal.



This arrangement is often called a ‘symmetrical’ output because the signal is carried symmetrically on both sides, but note that this symmetry is purely a convenience; it has nothing whatsoever to do with the rejection of interference! The interference-rejecting properties of a balanced interface employ the differential nature of a balanced input to cancel out any interference, but that critically relies upon the hot and cold lines having equal (ie. balanced) impedances to ground. Only then will any interfering signals develop exactly equal voltages on both the hot and cold lines, and thus cancel out accurately in the differential receiver.



A GC output doesn’t send a symmetrical signal but instead provides a ‘single-sided’ signal — at full level — only on the ‘hot’ pin of the output XLR (pin 2). There is no output signal on the cold pin, but as a differential input looks for the voltage difference between the hot and cold lines, it will still see and extract the full-level signal. The unusual aspect of the GC design is that not only does the cold line not carry any signal, it’s also not even an output: it’s actually configured as a ‘sensing’ input! The idea of an output socket having an input connection might appear confusing at first, and I’ll return to that in a moment, but first it’s worth considering the simpler but related ‘impedance balanced’ output configuration, which is also commonly employed on a lot of mixer and interface outputs.



The impedance-balanced output also sends a full-level signal on the hot side of the output XLR, and the cold side is connected directly to ground through a resistor, which presents exactly the same impedance as the hot side output’s impedance. This ensures that the system appears as a properly balanced source, and thus retains the normal balanced line interference-rejecting capabilities (see diagram).



In an impedance-balanced system, the entire signal is sent down the hot side. The cold side is arranged to have the same source impedance to maintain accurate interference rejection when connected to a balanced input. The full signal level is maintained if connected to an unbalanced input.In an impedance-balanced system, the entire signal is sent down the hot side. The cold side is arranged to have the same source impedance to maintain accurate interference rejection when connected to a balanced input. The full signal level is maintained if connected to an unbalanced input.



The benefits of this design are that it is cheaper to implement (needing only one active line driver instead of two), and the full signal level is received at an unbalanced input instead of being 6dB lower than expected (which would be the case if connecting just one side of a symmetrical output).



However, in a GC output the cold-side impedance-matching resistor is typically split in two to act as a voltage divider. The junction between the resistors is then used to feed part of whatever voltage appears on the cold line back into the output driver where it is added to the wanted output signal (see diagram). In a symetrically balanced system, half the source signal is sent down each side of the balanced lines, with opposite polarities. The differential receiver re-combines them to provide the full output signal. Both sides of the balanced line have identical impedances to ground, so any interference generates identical voltages on both sides, and is thus cancelled out at the differential receiver.In a symetrically balanced system, half the source signal is sent down each side of the balanced lines, with opposite polarities. The differential receiver re-combines them to provide the full output signal. Both sides of the balanced line have identical impedances to ground, so any interference generates identical voltages on both sides, and is thus cancelled out at the differential receiver.



Now, if this GC output is connected to a balanced input, there will be nothing on the cold line and so the system works exactly like a normal impedance-balanced output. But, if connected to an unbalanced input, the cold line will reflect the destination equipment’s ground, which may well be at a slightly different voltage to that of the source equipment’s ground because of a ground-loop. Normally, that difference would result in a ground-loop hum, but the GC output automatically adds any ground voltage difference to the source signal, thus neatly cancelling out any ground-loop hum.



I’m surprised the GC output form is not more widely employed because it is just as simple to implement as the impedance-balanced arrangement, requiring only a slightly different circuit. It’s just as cost-effective and easy to implement, but it provides built-in and extremely effective compensation for ground-loop problems that would otherwise require isolating transformers or other shenanigans to resolve.  

Published in SOS June 2015

Q Should I worry about the readings on my vectorscope?

Sound Advice : Maintenance




Mike Senior



I’m using the MeldaProduction Mixing Bundle plug-ins for mixing and I noticed that certain instrument patches (especially from Spectrasonic Omnisphere and the Access Virus) sound very appealing and wide, but when I check them on Melda’s MStereoScope [a vectorscope display] they appear fairly heavily out of phase. Certain drum-machine kits (such as Steinberg’s Groove Agent 4) also seem quite out of phase too. I’m not talking totally out of phase, but above 66 percent on the display, such that the MStereoScope manual suggests that “phase coherence may cause problems”. However, I also noticed that when I use these kinds of sounds in my full mix (ie. mixed with mono bass, mono kick, and so forth) an instance of MStereoScope on the master bus shows a perfectly healthy phase reading, and checking the mix in mono I don’t notice sudden sonic changes. So should I worry about the phase readings on the individual tracks/sounds, assuming that (for me) mono-compatibility isn’t that high a priority?



Balazs Ita via email



Although vectorscope plug-ins such as that in Melda Productions’ MStereoScope are extremely useful for avoiding mono-compatibility problems while recording and mixing, they’re no substitute for using your ears.Although vectorscope plug-ins such as that in Melda Productions’ MStereoScope are extremely useful for avoiding mono-compatibility problems while recording and mixing, they’re no substitute for using your ears.SOS Contributor Mike Senior replies: I’m also a big fan of Melda’s MStereoScope but I think you’re drawing too many conclusions from its display — like all audio metering, it should only really be used to assist your ears, not to somehow dictate what ‘should’ sound right. There is no standard stereo mix width that everyone agrees on, and it can vary a great deal depending on what kind of music you’re working on. So the first thing I’d say is to disregard the manual’s width recommendations, and instead study how some of your favourite productions look on the scope. That’ll give you much more useful information about the kinds of things you should be seeing on the display for the music you’re doing.



To get back to the specific question, it’s no coincidence that the patches that sound so appealingly wide have a strong out-of-phase component in their stereo signal, because it’s the out-of-phase component that’s directly responsible for stereo width. Indeed, synth-preset designers often deliberately hype out-of-phase components to deliver a ‘wider than the speakers’ stereo image, because that’s a good way to impress a lot of potential purchasers. The spaced microphone techniques often used for pop/rock drum recordings also tend to be rich in out-of-phase components. The downside of super-wide stereo signals, though, is that their out-of-phase component disappears in mono, which can cause their subjective level and tone to suffer if the mix is played on many smaller portable playback devices or on large distributed speaker systems in public spaces. Try switching to mono while soloing any of those synth/drum patches you mentioned and you should hear an appreciable difference.



Now, if you put the vectorscope display over your whole mix, it’ll be showing the overall balance between in-phase and out-of-phase components for all the sounds, so it stands to reason that if your most important instruments are mostly mono sources and centrally panned, they’ll dominate over your superwide sounds to give a fairly in-phase-looking result. This doesn’t take away from the fact that those super-wide tracks will still suffer in mono-playback situations, in exactly the same way they did in solo. If those tracks aren’t important to the mix, the sonic damage is unlikely to be a big deal; but if one of your lead riffs drops 3dB in level and changes timbre in mono, say, then that may be a more serious issue. As such, I wouldn’t rely on the vectorscope to troubleshoot this issue. The only foolproof way to reassure yourself that all is well is by switching back and forth between mono and stereo playback and listening carefully.



Of course, all that has to be tempered by how important mono-compatibility is to you, and in your case you may decide that you’re not too worried about mono losses. In which case, the question of stereo width becomes much more of a pure aesthetic concern. That doesn’t completely remove any need for restraint, though, as stereo width provides an important element of contrast in a lot of music productions, differentiating different instruments and musical sections from each other.  

Published in SOS May 2015

Korg microSTATION Video Manual: Sequencer: 5

Q How do I use phantom power with a passive mic splitter?

Sound Advice : Miking



Hugh Robjohns



I love your mag and always look forward to reading new issues on my iPad! I have a question concerning phantom power that I hope you can help me with: I have some of my condenser mics running into more than one preamp, via a Radial JS2 [passive mic splitter], with only one of the preamps applying phantom power. Is this the way it should be, and could it damage the mic or preamps in any way if both preamps are set to supply phantom power at the same time?



Martin Yap, via email



When using a passive mic splitter, you’ll do no damage to mic or preamp by applying phantom power from two preamps, but usually only one phantom source is needed.When using a passive mic splitter, you’ll do no damage to mic or preamp by applying phantom power from two preamps, but usually only one phantom source is needed.SOS Technical Editor Hugh Robjohns replies: You are using the device correctly: the preamp supplying phantom power should be connected to the ‘Direct Thru 1’ output, and the second preamp should connect to either the ‘Direct Thru 2’ or the ‘Isolated’ output. The ‘Direct Thru 1’ XLR passes phantom power directly to the mic.



There is no risk of any damage being caused (to mic or preamps) should you switch on the phantom power on both connected preamps, but doing so is pointless and you might forget to turn off both when connecting/disconnecting mics, resulting in an unexpected crack through your monitor speakers!



The Radial JS2 has one transformer-isolated output, which cannot pass phantom power to the mic at all and is intended to provide an electrically isolated feed for a separate recording system. The second (‘Direct Thru 2’) can pass phantom power, but only if the ground-lift switch is not activated. As I said above, the ‘Direct Thru 1’ output is intended to connect with the source of phantom power.  

Published in SOS May 2015

Korg microSTATION Video Manual: Combination Mode: 4

Q Are Perspex drum screens effective?

Sound Advice : Recording




For tracking a band in the same room, what are the pros and cons of using a Perspex drum screen? I imagine they reflect a lot of high frequencies back at the drummer, so does anyone have any experience with this colouring the sound in a good or bad way? How effective are they at isolating the drums from the other instruments? Would I be better off looking into absorptive panels?



Mike Senior



SOS Forum post



SOS Contributor Mike Senior replies: The main advantage of a clear Perspex screen is obviously that you can see through it! Beyond that, the question is what you’re trying to achieve. The main situation where I would consider a hard, reflective screen like that on sonic grounds would be when trying to increase the amount of early reflections captured by the drum mics, in particular the overheads. Such reflections do indeed colour the sound but can certainly do so in a nice way — especially when overheads are placed above the kit, because the screen can be positioned to reflect into those mics more of the rich timbre that emanates horizontally from the snare and toms, and which otherwise wouldn’t be that well represented. You might also use a reflective screen like this if your overheads are quite tight in for spill-reduction purposes, because that might help retain a more holistic impression of the kit from close up. (Alternatively, I’ve had good results with using a second pair of overheads in such circumstances instead.)



Perspex drum screens (such as this one by UK company Drumscreens) offer the advantage of a clear line-of-sight, but can they also offer sonic advantages in the studio?Perspex drum screens (such as this one by UK company Drumscreens) offer the advantage of a clear line-of-sight, but can they also offer sonic advantages in the studio?Photo: Clive Wilson / DrumscreenscoukThe main pitfall is that you may end up with nasty comb-filtering artifacts on any mics that are within a couple of feet of the screen itself — listen for that characteristic hollow, metallic timbre. If you’re not sure what I mean here, there’s an example in the audio resources accompanying January 2015’s Session Notes column (http://sosm.ag/sessionnotes-BHLH), specifically the BucketBaffle04 file.



Dealing with this is usually just a case of ‘suck it and see’. Solo each mic to check for tell-tale timbral pecularities, repositioning as necessary. In this context, coincident or near-coincident stereo pairs over the kit may offer an advantage over more common spaced-pair rock-miking techniques, as they allow you to place the mics over the centre of the kit, rather than close to the screen(s). You might also consider using boundary mics mounted on the screen itself instead, as these are able to side-step the comb-filtering problem by virtue of their design.



Mind you, my experience is that most project-studio ensemble-recording sessions suffer from too much reflection from hard surfaces in the recording room, rather than too little. As such, I typically find much more call for absorptive screens and blankets/duvets than reflective panels when recording bands in budget-conscious situations. As far as isolation is concerned, the problem with a Perspex screen in a small room is that it doesn’t take much energy out of the room, so unless you create some kind of close-fitting Perspex booth around the drummer (rarely a practical proposition!) you’re not going to make that much difference to the spill levels — in fact, you’ll delay the spill more, which often makes it sound more washy in the mix. With using absorbers, on the other hand, you effectively end up deadening the room as a whole to some extent, so even though they may not obstruct direct sound paths as effectively, the results are likely to be more mixable because the spill will probably be less delayed and have a shorter decay tail.



I most often use reflective panels in ensemble sessions for monitoring purposes, not for isolation, bouncing a player’s instrument sound back at their own ears so that they can better hear what they’re doing without resorting to headphone foldback. This situation seldom arises with drums, as they’re usually the loudest thing in the room as it is so there’s no trouble with anyone hearing them!  

Published in SOS May 2015

Korg microSTATION Video Manual: Program Mode: 3b

Q What does an API 2500 compressor’s ‘Thrust’ control do?

Sound Advice : Mixing



I don’t understand what the ‘Thrust’ control on my Waves API 2500 compressor plug-in does. I assume it’s the same as the hardware, but is it just marketing-speak for ‘side-chain EQ’ or is there more to it?



Hugh Robjohns



Jim McEwan via email



SOS Technical Editor Hugh Robjohns replies: The short answer is ‘yes’ in that it is just a side-chain EQ, but there’s actually a little more to it than that, as the detail of the equalisation is critical to its success and usefulness.



In essence, API’s Thrust circuit places a low-cut and high-boost equaliser into the audio signal that feeds the side-chain detector circuit. As you suggest, this is not an unusual concept; many compressor manufacturers include some kind of high-pass filter in the side-chain, with the specific aim of reducing the tendency of high-energy, low-frequency elements to dominate the control of the compressor too much. The benefit of this simple high-pass filter approach is to allow a louder overall volume with a stronger mid-range, while still retaining a well-controlled bass. In contrast, if the side-chain is not ‘desensitised’ to the high-energy bass instruments, the risk is audible pumping and a perceived loss of mid-range punch, because only the bass instruments tend to determine the compression level, rather than the mid- or high-frequency elements.



The Loud setting for the API 2500’s ‘Thrust’ EQs the side-chain signal to add 3dB per octave — which is the inverse of the response of pink noise. To find out what’s so special about this response, check out Eddie Bazil’s article on mixing to a pink noise reference in SOS December 2014.The Loud setting for the API 2500’s ‘Thrust’ EQs the side-chain signal to add 3dB per octave — which is the inverse of the response of pink noise. To find out what’s so special about this response, check out Eddie Bazil’s article on mixing to a pink noise reference in SOS December 2014.Q What does an API 2500 compressor’s ‘Thrust’ control do?API’s Thrust filter is an extension of the same idea, but the cleverness is in choosing the precise nature of the side-chain equalisation. The Loud Thrust filter option is the simplest to understand, as it is a linear equalisation slope that rises at +10dB per decade (which is +3dB per octave) from below 20Hz to above 20kHz. That +3dB per octave number is important because it is the precise opposite of the falling -3dB per octave spectral balance of pink noise. Pink noise is ‘special’, of course, because its spectral balance maintains equal energy in each octave band — a condition that gives a very natural-sounding spectral balance to our ears. (See Eddie Bazil’s article on setting a rough mix by comparing source tracks with pink noise: www.soundonsound.com/sos/dec14/articles/pink-noise.htm).



So, the effect of API’s Loud Thrust filter is not only to ‘desensitise’ the compressor to low-frequency elements so that they don’t dominate its action, but to do so in such a way that each octave band across the entire audio spectrum actually ends up with an equal amount of energy. This ingenious approach ensures that the naturally lower-energy mid- and high-frequency elements are given a ‘much fairer’ opportunity to determine the action of the compressor, with the result being a far more ‘punchy’ sound with a powerful mid-range and well-balanced high frequencies. It also tends to be perceived as a less compressed sound (for a given amount of overall gain-reduction) than that of a typical compressor working with a flat-response side-chain.



The Medium Thrust option on the API 2500 is a more complicated variation on the same theme, the key difference being that is has a flat response across the mid-range (between about 200Hz and 3kHz), but with the same +3dB per octave equalisation slopes below and above those corner frequencies. Naturally, the flat section reduces the ability of the mid-range to control the compressor slightly (compared with the Loud mode), although the compressor is still desensitised to high-energy low frequencies, and has increased sensitivity to the high-frequency elements. The third option is labelled Normal Thrust and this mode simply removes all of the equalisation from the side-chain, so that the compressor has a perfectly flat response and equal sensitivity at all frequencies — the same as all standard compressors. The choice of using Loud or Medium Thrust mode is primarily one of personal preferences, but also depends on the nature of the material being compressed and, in particular, the balance of mid-frequency components in comparison to the highs and lows.

Published in SOS April 2015

Korg microSTATION Video Manual: Set Up & Navigation: 2

Q When are diffusors a good idea?

Sound Advice : Recording




Diffusers, as pictured at the rear of this studio, come in many shapes and sizes but they’re only really useful in larger rooms — roughly the size of a double-garage conversion or bigger. In typical domestic spaces they’re usually too near to the listening position to have a positive effect, and can even cause problems.Diffusers, as pictured at the rear of this studio, come in many shapes and sizes but they’re only really useful in larger rooms — roughly the size of a double-garage conversion or bigger. In typical domestic spaces they’re usually too near to the listening position to have a positive effect, and can even cause problems.



I’ve seen some advice on building quadratic diffusors for peanuts, based on an old BBC paper. It seems a great idea, and much better looking than foam! But what I’m not clear on is when it’s a better option to use such a diffusor than to use absorption — presumably it depends on the size of room? Also, is there an optimum frequency range that you’d build these things to tackle? (Obviously they get rather big if you try to use them for bass, so I assume you’d use them alongside bass traps at the very least!)



Paul White



Chris Smith via email



SOS Editor In Chief Paul White replies: The main purpose of normal-sized diffusors (ones that would fit in a typical small studio) is to break up hard reflections by diffusing the reflected energy over a wide area, rather than bouncing it straight back like a mirror. That stops things sounding too ‘dead’ at higher frequencies. However they don’t do much of any use in smaller rooms, simply because they need space to allow the diffused energy to spread out. In short, while they’re potentially useful in professional studios and large rooms, I don’t see them as being very useful in most home-sized studios — and they’re certainly not going to be any use in controlling the low end.



SOS Technical Editor Hugh Robjohns adds: If you’re into basic DIY, all forms of room treatment can be done for peanuts. The visual aesthetics aren’t as important as the resulting acoustic benefits. Diffusors may look great, but they are sound reflectors not sound absorbers, and the latter are usually required far more!



When trying to optimise a domestic room or garage space for use as a control room, the most common problem is that it is too reverberant, with sound bouncing strongly off the parallel boundary surfaces. The easiest way of dealing with this is to install broadband absorber panels which ‘soak up’ any sound waves hitting them, thus preventing the sound from bouncing back into the room and so reducing the overall reverberation time, making the room less ‘live’ sounding.



The challenge with such an approach is in trying to maintain a nice, even balance of reverberation across the entire frequency spectrum. It’s incredibly easy to soak up too much of the high frequency sounds, but considerably harder to do the same for mid and especially low frequencies. It is consequently all too easy to achieve a reasonable reverberation time at mid and low frequencies, but with an excessively damped high end, resulting in a dull-sounding and unpleasant working environment.



In such circumstances, installing some reflective panels of various types can help to restore the balance of energy in the room at the high and upper-mid end of the frequency spectrum. Obviously, since a reflective panel absorbs negligible sound the energy is kept in the room, but we don’t want that reflected energy to create standing waves and slap-backs — and that’s where the diffusor designs come in.



Different diffusor designs operate over different frequency ranges, and bounce the sound in different planes (horizontally, vertically, or both), so some careful consideration is required when choosing and installing the panels, to determine where the reflected sound energy is directed.



The other issue to beware of is that with most diffusor-panel designs, reflected sound only becomes genuinely ‘diffuse’ a certain minimum distance from the panel, and if the panels are placed too close to the listening position you can hear an unpleasant phasey colouration in some cases. For that reason I would suggest a minimum of 1.5 metres between diffusors and listening position, and ideally more than that.



In my experience, diffusor panels are of limited use in small domestic rooms (floor areas below about nine square meters), but they become much more useful in larger rooms — so are perhaps worth considering if your studio is in a converted double garage or similarly large space. Bear in mind, though, that they take up considerably more space the lower the frequencies are that you wish them to diffuse.  

Published in SOS May 2015

Korg microSTATION Video Manual: Intro: 1

Q Why use two top-snare mics?

Sound Advice : Recording




I just finished reading SOS January 2015’s ‘Session Notes’ column and noticed that Mike Senior used two mics to capture the snare. Normally you would use one on top and the other underneath, but in this instance both mics were used on top. I assume the dynamic is to capture the ‘meat’ and the condenser is for the ‘snap’. This makes sense to me but I’ve never heard of this technique before — is it commonly used?



Mike Senior



The multi-miking setup Mike Senior used in January 2015’s ‘Session Notes’ column was designed to let him swiftly adjust the snare-drum sound by rebalancing the mic signals against each other.The multi-miking setup Mike Senior used in January 2015’s ‘Session Notes’ column was designed to let him swiftly adjust the snare-drum sound by rebalancing the mic signals against each other.SOS forum post



SOS contributor Mike Senior replies: Although multi-miking the top of a snare drum isn’t perhaps the most common approach, it’s by no means unheard of, and has apparently been used on a number of very high-profile albums. For example, Butch Vig set up both a Shure SM57 and an AKG C451 when miking up the top of Dave Grohl’s snare for Nirvana’s Nevermind; Dave Eringa mentioned using an identical setup for the Manic Street Preachers’ only number-one album This Is My Truth, Tell Me Yours; and Jim Scott combined a Shure SM57 with a Neumann KM84 for the Red Hot Chili Peppers’ Californication. (To read more about those sessions, check out our producer interviews in the March 1997, April 1999, and December 1999 issues of SOS, all of which are freely available to read in the magazine’s online articles archive.)



You’re right in surmising that the reason I chose to combine two close-mics was that I wanted the option to blend their different tonal characteristics during the session. In general, I much prefer to put together an ensemble sound by tinkering with mic positions and balances, rather than by dialling in processing, mainly because I tend to get the results I’m after quickest that way — and speed is a crucial factor on the kind of ‘smash and grab’ full-band location session I wrote about in that article. I’m far from the only one who eschews an under-snare mic. John Leckie, Elliot Scheiner, and Alan Parsons, for example, have all expressed reservations about it themselves. For my part, I rarely feel the need for extra level out of the snare wires, probably because I usually use a lot of overheads signal in my mixes and deliberately don’t put the overhead or over-snare mics too close, both of which tactics usually help the snare rattle come across fairly naturally without additional help.  

 Published in SOS April 2015

Korg microSTATION Video Manual: Program Mode: 3

Q Can I switch all my gear on at once?

Sound Advice : Maintenance




It’s fine to switch on gear from a mains switch, but some things are better switched on in a certain sequence. Thus, power distribution systems with front-panel switches — such as the Bryant one pictured — can be really handy.It’s fine to switch on gear from a mains switch, but some things are better switched on in a certain sequence. Thus, power distribution systems with front-panel switches — such as the Bryant one pictured — can be really handy.



Is it wise (or even safe!) to power up certain hardware — in this case a Nord Rack 2X — directly from the mains rather than its own on/off switch? The reason I ask is that the Nord Rack has its power button on the rear and this is inaccessible if you have another unit above it. Thanks.



Hugh Robjohns



SOS Forum post



SOS Technical Editor Hugh Robjohns replies: I answered a similar question a few years ago, which you can find at http://sosm.ag/qa-0802-power, but the answer is a resounding yes! I’ve never had any problems working this way... and I do so every day. The only caveat I would mention is that I don’t power everything in the studio all at once from just a single switch. Instead, I split the load and turn things on in small groups from separate switches, the reason being that the in-rush current from everything going on at once can generate enough earth leakage current to trip the fuse box RCDs! But for a similar reason to you, I turn on my Nord Stage from a mains supply switch that I can reach easily, rather than the on-off button at the back of the keyboard itself, and there is no problem with that whatsoever.



I actually use a Bryant Unlimited SDU (http://sosm.ag/bryant-sdu) to power up most of my racked equipment with inaccessible power switches (cheaper switching units are also available, often from DJ or disco-gear suppliers).



Reviews Editor Matt Houghton adds: The number of rackmount devices that appear with power switches only on the rear is a bugbear of mine! Like Hugh, I switch things on in groups, both for the reason he describes and to ensure I don’t end up with unexpected loud noises blowing the drivers in my monitors — in particular, I make sure that the power amp for my studio monitor speakers is the last thing that gets switched on, and the first to be switched off when closing down.  

Published in SOS May 2015

Korg Monotribe - Summer NAMM 2011

Korg microSTATION Video Manual: Set Up & Navigation: 2

Product Review - Akai APC40 MkII

Article Preview :: MIDI Controller For Ableton Live

Reviews : MIDI Controller

 A reboot of the first dedicated Ableton Live controller brings with it many welcome improvements.



Simon Sherbourne   Hardware companions to music software are now commonplace, but when the original APC40 launched in the pre–iPad days of 2009, it was one of the first of its kind. In my original review I was enthusiastic, bar a couple of layout gripes and feature requests. I’m happy to say that these have almost all been addressed in this new version without losing any of the essence of what made the MkI so usable. The question now is whether Akai’s take on what makes an Ableton Performance Controller stand up against the alternatives that have come since...


Ableton Control Relaunched



The new APC is significantly more compact that the original. The bulky bevelled surround is gone, and the launch pads are now slim rectangles instead of squares, trimming the size in all dimensions. The unit is also much lighter, which is a good thing for the travelling performer, right? Well yes, but it is a by–product of cheaper, more plastic, materials. The MkI was built like a piece of traditional pro-DJ equipment, with a tough metal enclosure and replaceable crossfader module. The MkII has more in common with modern music tech commodities that spend most of their time in the studio. That being said, it seems well built and sturdy enough to survive being carted around in a padded gig bag.



The core of the APC40 remains the 9x5 clip-launch grid, laid out above faders and track controls to mimic Live’s Session View. This maps to eight tracks at a time in Live, plus the Master and Scene Launch column. As well as the shape change, the most noticeable change is that the MkII’s pads are now fully multi–coloured, upgraded from the MkI’s green/yellow/red  

Published in SOS April 2015

Stealth Tuning with the Korg PC-1 Pitchclip Clip-On Tuner! (with Aliens!)