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Saturday, March 30, 2013

Solace Album Sample Trailer 2012 (Jordan)

Q. Should I be time-aligning my drum tracks?

After recording my drum tracks, I spend some time aligning all the waveforms. As a matter of course, I align the overheads to the close top snare mic, then I align all the mics to the overheads. I finally flip the phase buttons so I get positive attacks on all the tracks (meaning I usually have to flip everything apart from the in/out kick mics and bottom snare mic).

If you prefer not to time-align the tracks in a multi-mic drum recording, achieving adequate transient attack can sometimes be a problem. Fortunately, there are now numerous dedicated transient processors on the market that can help. Here you can see a small selection: Voxengo TransGainer, Stillwell Audio Transient Monster and SPL Transient Designer.
However, I was recently working with a very experienced recording artist who recoiled in horror at what I was doing, declaring that he knew no engineer but me who did this. With due deference, I returned the track timings to their original positions and reset the phase settings (though I left the snare phase flips), but to my ears it now sounds awful; the phase cancellations are killing the mix. So am I really alone in doing this kind of time alignment?
Via SOS web site
SOS contributor Mike Senior replies: 
In a word, no. By the same token, however, every engineer seems to have their own preferences in this regard, and I can understand that an artist might never have encountered such practices if he happened to have worked with engineers who didn’t bother with time-alignment, or else dealt with it only surreptitiously under cover of an unattended mixdown!
The reason for the lack of common practice is that a time-aligned kit sound has different characteristics to one that has not been time-aligned, so both tend to serve different goals. If you time-align, what you get is a more phase-coherent attack to your most important drum sounds, and this can make it easier for them to cut through other elements of the mix. However, the danger is that the weight and body of your drum hits may then appear lacking by comparison, and you can find that this leaves more dynamics-processing work to do at the mix.
The other problem is that time-aligning the close-miked drums can cause the overheads and room mics to pull those drums backwards into the mix. If you fade the room and overhead mics down to reduce this effect, the drum sound, as a whole, can then begin to sound rather anaemic and disconnected because of its reliance on the, typically, less natural-sounding close mics. Once again, extra compression and/or reverb might then be needed to draw out a satisfying sound.
By contrast, if you leave time-alignment out of your drum mixing process, you won’t usually get the same transient smack that time-alignment can provide, but the close mics may sound clearer and more up-front as a result of the time gap between their transients and the corresponding peaks in the overheads and room mics. In fact, producer Steve Albini has said that he occasionally delays his room mics artificially to increase this temporal separation.
The ‘smearing’ of the drum transients, which can result from a lack of time-alignment, may also help make the drums feel more solid for a given subjective mix level, simply because each drum’s combined drum-mix peak is broader in the time domain and is also not as far above the level of that drum hit’s sustain tail (compared with a time-aligned drum mix). The down side is that you may struggle to achieve enough attack for some aggressive styles, unless you get busy with specialist transient processors on the close-mic tracks. (There are lots of these to choose from, though, if you need them: SPL’s Transient Designer, Waves’ TransX Wide, Voxengo’s Transgainer, Stillwell Audio’s Transient Monster, Flux’s Bittersweet... the list goes on and on.)
So, whether you time-align or not, the result will always be something of a compromise; there are pros and cons of both approaches. What really matters, of course, is the sound in the context of the final mix, so the choice of methodology depends on which set of advantages you value most highly, and which set of disadvantages you can remedy most successfully.
One further thing to add, though: even without time-alignment, phase issues still needn’t make mincemeat of a drum sound. However, you can’t expect to use some preset configuration of your channel polarity buttons, because the phase relationships between the mics are more complex than that. Much better to introduce each mic or mic pair into the mix sequentially (I typically start with the overheads and room mics), flipping the relevant polarity switch for the most solid-sounding combination. Just make sure that you keep an ear on the entire line-up of drum kit instruments while doing this, as adding a tom-tom close mic, for instance, can easily affect the tonality of the snare or kick drum.
If the polarity buttons alone fail to satisfy your ear, try getting a phase rotator involved, to provide you with finer phase adjustments for the most critical close mics. There’s the freeware Betabugs Phasebug, and there’s also the better-specified IBP Workstation from Little Labs, which has just been released for the Universal Audio UAD2 processing cards.  

Friday, March 29, 2013

Korg All Access: Brett Tuggle and Kronos

Q. Do I really need to replace my windows to reduce noise?

Having just purchased my first house, I’ve found myself living on a busier road than I would have liked. My studio is at the front of the house and there is a reasonable amount of sound coming in through the old double-glazing. I expect the windows are at least 15 years old and they have trickle vents at the top, which obviously mean a small portion of the frame is always ‘open’. I’ve replaced all the hinges to get them to shut up tight, but it’s still a bit too noisy for me.

Rather than fitting special ‘acoustic’ glass, it will be cheaper — and probably more effective — to add secondary glazing to your existing windows.
I’ve had a pretty expensive quote to get new glass in the front of the house and, from what I’ve read, having different depths of glass on either side of the sealed unit can help. This is ‘acoustic’ glass and is 10mm on one side and 6mm on the other, which seems quite high spec, as other companies offer 6mm x 4mm. The overall unit depth is 28mm. Their claim is that this will give, on average, 39dB of sound reduction.
Should I try to purchase a cheap dB-measuring device. or can I cobble something together with a decent mic and a laptop? I want to see how much reduction I have at the moment, to try and figure out if the investment is worthwhile. Also, do you have any experience with these ‘acoustic glass’ products? From what I know about glazing, the logic behind the design holds up. However, these windows are approximately two and a half times as expensive as a run-of-the-mill window.

Once you’ve matched the sensitivity of two microphones, you can use them to measure how much attenuation a window is providing.
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: 
The trickle vents (if left open, or if they have poor seals) will always be the downfall, regardless of how well specified, designed and installed the rest of the windows are. Unfortunately, planning regulations may require you to retain the trickle vents, depending on the age and design of the building, so it’s worth asking that question of your window installer.
As for measuring the current level of attenuation, the easiest way would be with a simple sound-pressure level meter, the kind that costs around £16 from the likes of Amazon. Set it to slow response, A-weighted, and obtain readings from about a metre in front of the window outside, and again from inside. The difference will give a reasonable idea of the attenuation provided by the window. This kind of simple meter is also excellent for setting up monitoring systems.
Alternatively, if you have two similar mics, a couple of very long cables, two preamp channels and a DAW of some sort, you could do the same thing with those. I’d start by placing one mic outside the window, facing the road and adjusting the preamp gain to get a sensible recording level you can then use to calibrate the sensitivity of the second mic.
Once you’ve established a reasonable recording level, bring that mic back indoors, stick it in front of a speaker that is producing a constant level tone of some kind and place the second mic alongside it. Adjust the gain of the second preamp, to match the signal level of the second microphone to that of the first. A really quick and easy way of doing this is to sum the two mics to mono and put a polarity inversion in the second mic. When the mic sensitivities are matched, the two signals will almost perfectly cancel each other out, so simply adjust the second preamp’s gain for the deepest ‘null’. Next, remove the polarity inversion and mono sum, take the first mic back outdoors and place it in front of the window looking at the road again. Set the second mic up inside the room looking at the window and record a few minutes of traffic noise with both mics.
All you need to do now is compare the average levels of the two recorded tracks to find out what level of attenuation the window is currently providing. The DAW meters will provide the information you need if you leave the peak hold indicators on. It might also be educational to close off and seal the trickle vents with gaffer tape to see what difference they make to the figures.
Assuming that the existing windows are in good condition, I suspect that replacing them with new ones — even the higher attenuation ones — won’t make that much difference. Secondary internal glazing, adding a third layer to the window sandwich, is likely to be far more effective, but perhaps not as attractive and maybe not as convenient. You’d need something like a 10dB improvement just to make the ambient noise sound half as loud, and that’s extremely difficult to achieve with normal domestic window designs.  

The Korg MS-20 Mini: Envelopes

Thursday, March 28, 2013

The MS-20 Mini Part 7- VCA2

Q. Is there an easy way to match the gain of different channels?

I’ve committed myself to recording a school orchestra in a couple of weeks. Obviously, this will involve using stereo pairs of mics. However, none of my preamps have stepped gain controls and, in fact, most of them have very tiny knobs, so matching the gain on different channels by eye is unlikely to work well. Is there a better way to match the gain across different channels? Would it be better to take a small tone generator and hold it against the front of the mic, or something?
Ceri Jones, via e-mail
SOS Technical Editor Hugh Robjohns replies: 
A tone generator is one solution, if you can guarantee to get it the same distance from both capsules, but it’s fiddly and not that reliable, in my experience.
There are several good alternatives, though, depending on what kind of mic arrays you’re using and how easy they are to get to. It’s also made easier if you have a Lissajous meter display (goniometer) like the DK-Technologies MSD series, and a monitoring system that allows easy access to the side (stereo difference) channel.

A goniometer can help you easily match gain across channels if your preamps don’t have stepped gain controls, but there are also cheaper methods.
The easiest approach is to roughly set the mic gains by ear during rehearsal. Then at the break, when the room is empty and quiet, get someone to stand in the front-middle of the stage and clap their hands repeatedly (or, if they’re not shy, sing a constant note).
On a goniometer you’ll see very clearly the stereo axis of the sound source, so, you can then turn down the louder channel (the side the goniometer trace leans toward) to bring the display back to the centre line. Turning the loud side down maximises headroom, of course, and is a safer way to go than bringing up the quieter side!
If you don’t have a goniometer, a reasonably practical solution is to configure the monitoring to listen to the ‘side’ or stereo difference signal (polarity-reverse one channel and mono-sum them).
If the two sides are equally matched, there should be a deep cancellation null, so by looking at the meters to figure out which channel is louder, wind that down until you pass through the null, and then bring it back up to provide the deepest possible null. Then restore the monitoring to normal stereo. This process works well for continuously variable gain controls that aren’t closely matched, such as those you’re describing.
Frustratingly, though, few monitor controllers have facilities to switch to hear the side signal, and few people appreciate the true value of goniometer metering displays, both making the situation you describe trivially simple to check and resolve. 

SSL Nucleus - AES 2010

Wednesday, March 27, 2013

Q. Is working with digital recordings harder than working with analogue ones?

In the past few years, it seems that I have to work much harder to get things to sit properly in a mix — to get the vocal or horns to just blend with the rest of the track, rather than feeling ‘stuck on top’, for example. What has crossed my mind is that I rarely (if ever) seemed to find this an issue when I was working purely in the analogue realm. Was I being helped by the losses in the analogue system to blend the sounds? Is it harder to blend multitrack recordings in the digital world? I’m a musician, really, but I think I’ve improved as an engineer over time, so I should say that I’m not a total klutz at this. I do usually manage to get things to blend, but it does take effort. Do you have any tips for improving the situation?
Via SOS web site
SOS contributor Mike Senior replies: 
There are a lot of good reasons why recordings made entirely in the analogue domain often seem easier to glue together at mixdown. The compression side-effects of the tape recording medium often help to tame over-spiky transients (especially on drums), which can be difficult to tuck into the mix otherwise. The progressive high-frequency loss that tape-recorded signals suffer after multiple playbacks helps push sounds further away from the listener too; the brighter a sound, the more it tends to pop out of the mix.
Background noise is an inevitable side-effect of working in the analogue domain — not just on account of the tape itself, but also because of contributions from all the other processing equipment — and this combined noise floor usually makes it easier to blend a mix. To quote producer Steve Churchyard (in Howard Massey’s book Behind The Glass), “Tape hiss doesn’t bother me at all, never did. It’s like the glue that holds the record together”. A little added distortion is also unavoidable in analogue setups, and this can be turned to advantage by experienced recording engineers to make sounds fuller and more present. Such sounds don’t need to be faded up as high in the mix and are, thus, easier to balance.
One other factor a lot of people forget regarding analogue productions is that compression is more often done while recording, to make the best use of the tape’s dynamic range and the available gear resources, and then many of those parts may be further compressed at the final mix. This kind of serial compression is typically better at levelling out performance levels than a single, more heavy-handed, processing stage, so that can also affect blend and the overall sense of naturalness.
There are other factors that contribute to the analogue sound, but that’s enough to be going on with at the moment! Let’s start looking at how you can try to get similar effects in the digital domain. The bottom line is that you can’t expect to use all the same techniques you used for your analogue mixes when working on an all-digital production. So, for example, I normally find that I do a lot more work with tape emulation, saturation, clipping and specialist transient processors when mixing digital recordings, in order to bring the typically less-rounded transients under control. Tape emulations are, of course, an option here also.

Adding background noise artificially can also help achieve more analogue-style blend, and if you don’t fancy sourcing your own noise recordings, there are a lot of places you can find suitable samples. Most media sound effects libraries have a selection of what are usually called ‘room tone’ or ‘room ambience’ files, which are the sound of nothing happening in various common environments; not the most interesting sounds, but they really help to make tracks feel as if they’re all occurring in the same place.

Vinyl noise is another good option, and I’ve found good examples in many sample libraries. Spectrasonics’ Retrofunk (www.spectrasonics.com) and Tekniks’ The Mixtape Toolkit (www.tekniks.co.uk) spring to mind immediately, but there are lots of others. The Swedish developers Retro Sampling (www.retrosampling.se) have made background noise something of a speciality, and you can get whole CDs full of different vinyl noises from them, plus they also do freeware Audio Impurities and Vinyl Dreams VST plug-ins, which give a small taster of what their product range has to offer.

There are other plug-ins worth a look too, such as Izotope’s Vinyl (www.izotope.com) and Cubase’s built-in Grungelizer, but be aware that some of these don’t output everything in stereo, and mono noise won’t help the blend nearly as much in this application. One other freeware plug-in that you might try is Tweakbench’s Field (www.tweakbench.com), which provides a selection of mixable room tones./BodyI>

Retro Sampling’s Audio Impurities Vintage Edition and Tweakbench’s Field are two freeware plug-ins that can feed subtle background noise to your mix bus, thereby subtly improving your apparent blend.
Finally, it’s pretty easy to create serial compression digitally, given the practically limitless plug-in slots most sequencers are endowed with. My basic advice here is to use slower and gentler compression settings for the first compressor in the line, just to even up the levels, and then use faster and heavier compression only further along in the processing chain. If you do it the other way around, the fast compressor will usually cause too many audible processing artifacts, while the slow compressor won’t have much dynamic range left to work with. 

The Korg MS-20 Mini- Oscillators Part 2

Q. How do the different amp classes work?

I’m trying to learn a little more about amp design. One thing that really baffles me is the different classes available. What does an amp’s class mean, and how does this affect the way it is used?
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: 
In a Class-A circuit, the active device (whether valve or solid-state) passes current regardless of the polarity of the input signal; in other words, in an audio application, it is ‘biased’ so as to pass both the positive cycle and the negative cycles of an audio signal. The side effect of the biasing is that the active device has to pass current all the time, making it relatively inefficient.
In a Class-B circuit, the active device only passes current for one polarity of input signal — which polarity depends on the circuit design — and this makes it a much more efficient way of working. So, in this case, where it is required to pass a symmetrical audio signal using a Class-B circuit, the circuit will need two active devices, one to handle each polarity. This is an arrangement often also known as ‘push-pull’.
Class C is a format that only conducts on signal peaks and is rarely (but occasionally) used for audio in situations where power efficiency is more important than distortion. Class D — which is now becoming very popular in audio applications — works by generating a stream of high-voltage pulses at a very high frequency. These pulses are modulated in such a way that the average energy they convey follows the wanted audio waveform.
Returning to the Class-B design, this exhibits a problem called crossover distortion for audio applications, because both of the active devices in the push-pull pair turn off as the signal nears the zero line. The solution is to bias the devices so that they don’t turn off. They actually continue to pass signal as it crosses over into the opposite polarity. In other words, it works a little more like a Class-A device (but without the same levels of power inefficiency).

In a push-pull amp design, each active device handles one polarity of the input signal.
Hence the compromise name Class AB; it is a Class-B design biased to operate in a similar way to Class A around the crossover region. However, it should also be remembered that push-pull designs can also be operated fully as Class A if required, and some high-power amps do work in that way. This is also a handy technique for cancelling out even-harmonic distortion products in tube-amp designs.

Tuesday, March 26, 2013

The Korg MS-20 Mini- Oscillators Part 1

Q. How can I improve acoustics in a long, thin room?

The diagram to the right shows my room, which serves as my studio. The dimensions seem to be bad for low frequencies and there are sound-pressure failures at 55Hz and between 110 and 140 Hz. I have an Auralex foam bass trap, but I don’t known if absorption is the answer. What should I do to improve this situation?

If you have any choice of rooms for your studio, try to avoid those whose dimensions are multiples of each other.
Via SOS web site
SOS columnist Martin Walker replies: 
I agree: that’s a bad choice for a room, dimensionally, as far as acoustics are concerned. The 2.6-metre width and 2.5-metre height are nearly identical, while the 5.8-metre length is close to double these, giving you a shape that’s almost two cubes joined together. The room is also relatively small, which will mean it’ll have relatively few modes below a few hundred hertz and, as the dimensions are closely related to each other, these modes will pile up at some frequencies (resulting in a huge peak), with large gaps between them (creating big dips in the frequency response).
Room-mode frequencies are fairly easy to calculate, but it’s even easier to plug your three dimensions into a utility, such as the on-line MCSquared Room Mode Calculator (www.mcsquared.com/metricmodes.htm) or the Hunecke Room Eigenmodes Calculator (www.hunecke.de/en/calculators/room-eigenmodes.html). However, if you’ve got a PC, the ModeCalc utility from Realtraps (www.realtraps.com/modecalc.htm) is one of the easiest to use, displaying the first 16 axial modes for each room dimension up to 500Hz in an easy-to-interpret graphics plot. It would show that the biggest gaps in your room mode plot occur between 30 and 60 Hz (which explains your hole at 55Hz), between 70 and 90 Hz, and again between 90 and 130 Hz (the other area you’ve already pinpointed).
Without acoustic treatment, your listening position will be very critical, since you can end up sitting in a bass trough at one frequency and a huge peak at another. However, your loudspeakers and listening position do look to be near their optimum locations for the flattest compromise response. The oft-quoted ideal is to place your listening position (ears!) close to 38 percent into the room from the front wall.
Acoustic-foam bass traps, like the one you already have, can certainly be effective, and acoustic foam is also excellent for dealing with mid-/high-frequency early reflections from your side walls and ceiling. However, acoustic foam is invariably a lot less dense than the 60kg/m3 Rockwool that is generally recommended for DIY bass traps, and you will, therefore, require a much greater volume of it to achieve a similar amount of absorption at lower frequencies. In a small room, you’ll simply run out of space before you can cram in enough acoustic foam traps to adequately deal with the problems.
Diffusion can be a good way to ‘break up’ your reflections so they become less troublesome, but you ideally need to be at least a couple of metres away from them to avoid hearing a set of discrete reflections, rather than a more diffuse soundfield, so they are not often used in small rooms like yours. Tuned traps also have their place in the grand scheme of things but, in my experience, tend to be more difficult to tune and place optimally compared with broadband trapping that you simply fit where the bass levels are loudest, so they absorb the sound more efficiently.
Overall, I think broadband absorption is your best bet; as much of it as you can reasonably fit into your room. Start by placing traps that straddle the front vertical corners of the room, then the rear vertical corners, followed by any other corners you can manage, such as the ceiling/wall corners, and even the floor/wall corners where feasible. Also, don’t forget some side panels and ceiling ‘cloud’ at the ‘mirror points’ to deal with early reflections.

The Korg MS-20 Mini- Introduction

Monday, March 25, 2013

Q Is it possible to achieve a ‘silent’ band setup?

To enable very quiet band rehearsals, I am thinking of feeding the whole band setup into my Behringer RX1602 Eurorack Pro mixer. This would be a guitar and bass through DI boxes, and I might be able to get my hands on an electronic drum kit. I’d then wire the mixer up to a headphone amp. If everyone gets a pair of headphones it should be pretty quiet from the outside. Could you tell me if I’m on the right track for what I want to achieve?
Via SOS web site
SOS Editor In Chief Paul White replies: 
While you can certainly use a mixer and its built-in monitor capabilities to do this, you’d still need to add a multi-channel headphone amp. These need not be expensive, however. Most mixers are also designed for mic or line-level inputs, not instruments, so you would need to use electronic keyboards, guitar DI preamps and electronic drums as sound sources.
A possibly neater solution is to look at the JamHub system, which has been specifically designed for the application you’re trying to achieve, though it would obviously require more of a financial commitment. It’s available in three models with varying numbers of inputs and outputs. So, depending on the model you choose, it could work with a maximum of seven musicians, each of whom can have a stereo instrument input and a mic input. This includes separate headphone amps for each performer, with control sections where everyone can set up their own monitor mix. We’ll be reviewing this in a forthcoming issue of Sound On Sound, but you can check out the details at www.jamhub.com.

The JamHub is designed to give you as close to a ‘silent’ studio as possible.
You may not be able to achieve a totally silent rehearsal space this way, but, presuming you are using electronic instruments, this kind of setup should definitely offer a serious reduction in sound levels.  

Trident Series 82 - AES 2010

Q How common is re-amping?

I’m curious to know how common the use of re-amping is amongst producers, engineers and composers. There are so many applications for this that I find it strange that re-amp circuits are not built into mixers, DI boxes, soundcards and audio interfaces.
There are also relatively few re-amp boxes available, and they’re all absurdly overpriced! What is so complex about a re-amp box to warrant a price tag of over £200?
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: 
Re-amping is the process whereby the direct signal from a guitar, bass or keyboard is recorded — usually on a separate track alongside the signal captured simultaneously with a microphone from an amp — and later routed to an amp in a studio to be miked up and overdubbed.
This approach allows the choice of amp or amp settings, or mic and mic position, to be changed after the initial recorded performance, but without the compromises and limitations inherent in trying to process an already recorded amp sound. It is a popular and widely used technique, although it is more common in the production of some musical genres than others. Re-amping can be both a time saver and a time waster, depending on how and why it is employed! As a way of modifying a guitar part to better suit the track as the mix progresses, it is an invaluable technique, saving the time and effort of having to record a new performance. However, if used to avoid committing to a sound during tracking, it can be an enormous time waster.
Often, traditional re-amping is replaced by virtual re-amping using guitar-amp plug-ins, many of which offer remarkably good quality and enormous versatility. The process is exactly the same, but without having to physically route the signal out of the DAW and into a real amp in a real studio, miked up with real mics.
There are various products available with integrated facilities for re-amping, as well as dedicated re-amping units, although the latter approach seems the more popular. There is nothing complicated about a re-amp box, which, in most cases, is essentially a passive DI box used in reverse.
A re-amping box accepts a balanced line-level signal (nominally +4dBu) and converts it to an unbalanced instrument-level signal (nominally -18dBu), usually via a transformer. A variable level control is often provided to optimise the level fed to the amp, along with a ground-lift facility to separate the balanced source and unbalanced output grounds, thus avoiding ground-loop hum problems.

The Radial ProRMP allows fairly hassle-free reamping, giving you the ability to change amp settings and effects, for example, after the initial recording has taken place.
A passive DI box can often be used reasonably well in this role, although it is normally necessary to attenuate the line-level input significantly, to avoid saturating the transformer and generating an excessive unbalanced output level. Alternatively, the kind of line-level balanced/unbalanced interface intended for connecting domestic equipment to professional systems can be used, and the original ART CleanBox is often recommended in this role. However, for only a slightly greater outlay, a dedicated re-amp box, such as the Radial ProRMP, is rather more convenient to use.  

Saturday, March 23, 2013

Allen & Heath GS-R24 - NAMM 2011

Q What’s the best way to create sub-bass synth sounds?

Do you have any pointers on making a great sub-bass synth sound to underpin bass lines? It seems to be one of those things that’s hard to get wrong but, at the same time, difficult to get bang on the money. Should I use sine, square or triangle waves? How steep a filter should I use, and at what frequency should I set it? I’ve used sub-bass synths with varying degrees of success, so I’d be interested in any handy hints.
Via SOS web site
SOS contributor Mike Senior replies: 
Sub-bass synth parts can operate in very different ways, so it’s difficult to generalise. The simplest application is where you double your tune’s bass line at the octave below using a simple sine-wave synth patch. In this case, there’s nothing much to do other than set the synth’s level and have a listen to it in solo, just to check that the synth’s envelope settings aren’t so fast that they create unwanted clicks. Because a sine wave is effectively only a single frequency, and that frequency doesn’t overlap the main bass part’s range in this scenario, there’s no need to filter the sub-bass synth at all.
Things get more complicated if you’re using a sub-bass synth to try to beef up the existing fundamental frequency of your bass part, because the sub-bass synth, therefore, overlaps the main bass part’s frequency range. The problem is that if the peaks and troughs of the ‘sub’ synth’s waveform don’t track those of the existing bassline’s own fundamental frequency component, the combination can actually end up sounding less bassy than before! And because the relationship between these two sets of waveform peaks will usually change from note to note, you may end up with a very uneven low end that’s all but impossible to balance with the rest of your mix. If the existing bass sound’s fundamental frequency is weak enough, by comparison, with the added sine wave, this effect may not be significant enough to be a problem, but if you do get into difficulties you need to try to get rid of the original sound’s fundamental frequency, in order to clear the field for the sub-bass synth at the low end. A steep high-pass filter on the main bass part is one solution, but at times you may need to use the more surgical approach of notching out individual fundamental frequencies with narrow-peaking EQ cuts. A high-resolution spectrum analyser may help, or you could, alternatively, plug the bassline note names into the note-to-frequency calculator at www.muzique.com/schem/freq.htm, in order to find approximate notch frequencies. Again, though, filtering the sub-bass synth won’t help at all in this case.

In some cases it makes sense to eliminate, very precisely, the fundamental frequencies of your main bass part, to avoid troublesome conflicts with an added sub-bass synth part. If you find yourself in this position, a note-to-frequency calculator, such as this one at www.muzique.com, can save a lot of time by identifying the EQ frequencies you need.
The point at which filtering becomes an issue is when you’re wanting to round out the overall low-end tone of the bass sound, rather than just adding a sub-octave or emphasising the existing fundamental. A sine-wave sub-synth won’t help you here, because you want a waveform that has some harmonics in addition to its fundamental frequency. I like using a triangle wave instead of a sawtooth or square most of the time, because it seems to be better at blending with (rather than overwhelming) the sound it’s layered with. The triangle wave doesn’t have such dense harmonic spacing as a sawtooth, and is duller-sounding and less characterful than a square wave.
Whatever waveform you use, though, you still need to take exactly the same precautions with the sub-synth’s fundamental frequency as you do when using a sine wave. I’d also steer clear of detuned multi-oscillator patches, because the ‘beating’ between the two detuned layers may cause the sub-synth’s fundamental frequency to fluctuate unacceptably in level. Stick with mono patches too, because low-end stereo width can reduce the power and consistency of the bass sound in mono, and will also interfere with vinyl pressing if you’re planning to take that route. These restrictions mean that you only really need a very simple instrument to generate sub-synth parts. For the ‘Mix Rescues’ I do in Cockos Reaper, even that sequencer’s no-frills little ReaSynth instrument is over-specified, and I’ve used that as a sub-bass synth on numerous occasions.
The decision as to whether to filter the sub-synth is purely a question of what kind of low-end tonal enhancement you’re looking for. With a triangle wave, in particular, you might not feel any need to filter it at all, although I do personally find myself employing some kind of low-pass filter to restrict its input to the lower octaves in most cases. The slope of the filter is typically quite critical, though, so if you can find yourself something with a variable roll-off slope, that does give you a useful amount of extra control. However, I wouldn’t use a resonant filter in this kind of application unless that filter is set to track the synth oscillator’s pitch, otherwise the filter’s resonant peak ends up boosting different harmonics as the note pitches change, and this makes the sub-bass synth less likely to blend consistently with the main bass part.
One final point to make is that sub-bass synth parts usually need to be controlled quite tightly in terms of dynamic range, or else they can really eat into your track’s overall headroom. It’s also usually sensible to avoid having a sub-heavy kick sound when there’s a prominent sub-bass synth underpinning the bass line, for similar reasons. There’s only so much space down there, so if you want massive, subby bass, you either have to sacrifice some of the kick’s weight or turn down the overall level of your track to accommodate the low-frequency build-up.

Friday, March 22, 2013

Midas VeniceF - NAMM 2011

Q. Can I use a Mac Mini for music?

I always hear people saying that the Mac Pro is the Mac of choice for musicians but, as a hobbyist, I simply can’t justify the expense. I’m tempted by a Mac Mini, as I already have a decent screen, but am concerned that it won’t be able to cope with the requirements of audio recording. What are the pros and cons?
Petra Smith via email
SOS contributor Mark Wherry replies: 
While it used to be the case that a high-end computer like the Mac Pro was essential for running music and audio applications, these days it’s really hard to purchase a system that will be incapable of such tasks. It’s all a matter of how many audio tracks, instruments and effects you need the computer to handle. Among the most important factors to consider in determining such handling are the type and speed of the processor, the amount of memory and the speed of the hard disk.

The updated Mac Mini comes with a 2.4GHz processor, 2GB RAM and a 320GB hard drive as standard, making it perfectly capable of running decent numbers of tracks.
Photos courtesy of Apple
Since the first Power PC-based model was introduced (see the full review at www.soundonsound.com/sos/may05/articles/applemacmini.htm), the Mac Mini has established itself as a basic-yet-capable studio computer. The current range features Intel Core 2 Duo processors, and the 2007 MacBook Pro (which, with a 2.4GHz processor, had similar performance capabilities) gives us a rough guide of the performance you can expect: using Logic Pro 7, this was capable of running 150 PlatinumVerb instances, 54 Space Designers and 512 EXS24 voices (with the filter enabled). Today’s baseline Mac Mini also has a 2.4GHz processor, so those figures should be roughly comparable.
When it comes to memory, the 2GB supplied in the entry-level Mac Mini should be just enough to get you started. But you’ll find life rather more comfortable with 4GB, especially if you want to work with sample-based instruments. It’s worth bearing in mind that 8GB is the maximum amount of memory supported by the Mac Mini.
In terms of storage, the basic Mac Mini comes with a 360GB drive. But, perhaps more crucially, this internal drive runs at 5400rpm — slower than those used in most other Macs — which will limit the number of audio tracks you can play back simultaneously. As a guide, you should expect to be able to handle approximately 50 to 60 mono 16-bit tracks at 44.1kHz. However, it is possible to connect a faster drive for audio, thanks to the Mac Mini’s built-in FireWire 800 port — assuming you’re not already planning to use this port for an audio interface, of course, since daisy-chaining devices isn’t always possible.
Another important factor when considering the Mac Mini, and one that might initially sound a little bizarre, is price. Although the Mac Mini is the cheapest Mac that Apple sell, its starting price can be deceptive in terms of value, even though, on paper, it’s several hundred dollars cheaper than the cheapest iMac. If you already have a suitable monitor, keyboard and mouse, that’s fine. But if you factor in the cost of these required devices to even the cheapest Mac Mini, the price difference between that and the low-end iMac starts to narrow considerably.
In a nutshell, the Mac Mini remains a basic, yet capable machine that provides a good starting point. However, in many ways, the entry-level iMac represents better value for those on a budget, especially if you see yourself quickly outgrowing the Mini’s capabilities.  

Q. How can I prevent feedback?

When setting up for a gig we always suffer really bad feedback from the singer’s mic. We’ve tried positioning things differently, but it doesn’t seem to help. We’re pretty new to this; how can we counteract feedback?
Jo Ellison, via e-mail
SOS Editor In Chief Paul White replies: 
Acoustic feedback is caused when sound from the speakers gets back into the microphones at a high enough level to cause the signal to keep increasing. This produces acoustic feedback as the signal cycles round and round the system. Positioning the main speakers well in front of the vocal mics and aimed so as to minimise the amount of sound bouncing back into the microphones will help, but there are other issues to consider. For example, if the wall behind the band is hard, it will reflect more sound back into the live side of the microphones. Imagine the room is made of mirrors and it’ll be easier to establish where the problematic reflections are likely to come from. If you can hang up a thick fabric backdrop, it will help, as will positioning the main speakers so that most of the sound goes into the audience, and as little as possible points toward the walls and ceiling.
Feedback always starts at the point where the gain is highest and where the phase of the audio picked up by the mic reinforces what is coming from the speakers. If you apply EQ boost, there’s more likelihood that feedback will occur at the boosted frequency, as that’s where the gain is highest, but the same applies to microphones and PA speakers that have significant peaks in their frequency response curves. Choosing good-quality mics and speakers might help to minimise the risk of feedback. A mic with a gentle presence peak should be OK, but some cheaper mics have very pronounced peaks that can cause problems. You also need less gain if the singer has a naturally loud voice, so those with quieter voices need to work close to the mic. Quiet singers who stand back from the mic have no chance in smaller venues, where mics are invariably closer to the speakers than is ideal.
Stage monitors can be particularly problematic when it comes to feedback, so it pays to spend a little more on monitors that have a reasonably flat response. You also need to ensure monitors are aimed toward the least sensitive part of the vocal microphone, which, for a cardioid pattern mic, is directly from the rear. You may need to angle the back of the mic downwards to achieve this, but it will help. Hypercardioid mics, on the other hand, tend to be least sensitive around 45 degrees off the rear axis, so aim the monitor there.

The area directly behind a cardioid mic is the least sensitive, so positioning stage monitors there will reduce the risk of feedback. However, if you’re using a hypercardioid mic, this is true of the area at a 45-degree angle to the rear axis.
A third-octave graphic EQ can help pull down troublesome peaks, but the type you find built into mixers, with only five or six bands, isn’t very useful for dealing with feedback, as they change too much of the wanted sound. They can help balance the overall room sound, but that’s about it. A better solution may be to connect an automatic ‘feedback eliminator’ hardware device to the mixer output. These are set up during the soundcheck by turning up the mic gain until feedback occurs, at which point the device measures the frequency and sets up a narrow filter to pull down the gain at that frequency. Most have several filters that can lock onto the main feedback frequencies, and they can help you gain a few more dBs of level before feedback becomes a problem. As the filter bands are so narrow, they have little effect on the overall sound. Most also include roaming filters that can lock onto feedback that occurs during performance, as it might if the singer moves the mic around.

Small venues of the type that so many up-and-coming bands play definitely make the fight against feedback harder, as they provide fewer opportunities for optimum positioning of PA speakers.
Finally, when setting up levels, establish a maximum safe vocal level, leaving a few dBs of fader travel in hand, rather than working right on the edge of feedback where the sound is ringing all the time. Then set up the level of the back line to match the vocals. It’s no good setting up the backline first and then expecting the vocals to match it, because in most small venue situations the vocal level is the limiting factor. You’ll also find that some venues are inherently worse than others for feedback and you just have to live with it.  

Spectrasonics Omni TR - NAMM 2011

Thursday, March 21, 2013

Q Are there any studio monitors available to fit my budget?

I want to get a pair of active monitors to do a bit of home recording and try my hand at producing. I’m not after professional quality, just entry-level monitors, as my maximum budget is around £180. Do you have any recommendations or advice?
Via SOS web site
SOS Editor In Chief Paul White replies: 
The answer depends on your room size and, to be honest, your budget is a tight one, but there are a few viable options. If your room is small, aim for a speaker with a bass driver no larger than five or six inches, but no smaller than four. Unless you already have a suitable amp, you’re better off going for active speakers where the amplifiers are built in. As a very general rule, larger home-studio rooms can take monitors with up to eight-inch drivers, but these tend to be more expensive. 
The current Behringer B2030A Truth active speakers offer a good performance/value ratio (though they are slightly over your budget, at around £200 per pair in the UK), as do the Fostex PM04s and the lower-cost M-Audio models, such as the Studiophile AV40s and Studiophile BX5A Deluxes (these can all be found in the UK for well below your budget, at as little as £100 for a pair of the Studiophile AV40s).

The M-Audio AV40 Studiophiles offer decent value if you’re on a tight budget, going for as little as £100 a pair in the UK. They’re also convenient for a bedroom studio, as they are only six inches wide and have a front-panel volume control.
These are all fairly small speakers, so don’t expect very deep bass. However, they should all be loud enough for close-up monitoring. You also need to be aware that the room acoustics and the way you mount your speakers will affect the sound, so you might like to take a look at some of our ‘Studio SOS’ articles at www.soundonsound.com to help fill the gaps in your knowledge. Bear in mind also that speakers with volume controls on the front may be more convenient if you don’t have a monitor level controller.

Alesis Multimix 6 Cue - Musikmesse 2011

Q What buffer size should I use?

I’ve worked with tape and ADAT in the past, but have been out of recording for a few years. I’m just getting back into it and have got my first computer recording setup, with a PC and a Focusrite Saffire Pro 40 audio interface, but I’m confused by the buffer settings: what buffer size should I use in my projects?
Dom Gately, via email
SOS Reviews Editor Matt Houghton replies: 
When it comes to buffer settings, there’s a trade-off between achieving low latency and reducing the strain on your computer’s CPU. The smaller the buffer size, the greater the burden placed on your CPU, but you’ll get lower latencies (for less audible delay), which is what you want when monitoring recordings through your sequencer and any processing. Similarly, the greater the buffer size, the greater the latency, but with less strain being placed on the CPU. If the latency is too low, you’ll hear pops, clicks and glitches as your computer struggles to keep up. You’re not doing any damage, so if you need low latency, try setting it down as low as you can until you hear those glitches and then raise it up a little.

The smaller the buffer size, the greater the strain on your computer, though you’ll experience less latency. If you need low latency, set the buffer size as small as your computer can manage without producing clicks and pops.
When mixing, you’re likely to need more processing power as you start to add more and more plug-ins. So if starting a project from scratch, I’d usually set buffer size as low as possible while recording or playing parts via a MIDI keyboard, but increase it later, when the recording was finished and I was ready to begin mixing in earnest. It’s also worth mentioning that, while tracking, it should be fine to use a ‘lite’ version of a reverb plug-in for artist monitoring duties, if this helps take the strain off your CPU, and replace it later on when you want to sculpt the sound for your mix.
SOS Features Editor Sam Inglis adds: 
It’s not clear from the question what sort of recordings you’re making. However, unless you’re using soft synths or samplers, it might be better to use the Saffire Pro 40’s mixer utility to set up a low-latency monitor mix. That way the question of buffer size becomes largely irrelevant.  0

Wednesday, March 20, 2013

Spectrasonics Omnisphere 1.5 Part.2 - NAMM 2011

Q. How should I record an upright piano?

I have a pretty basic recording setup and, up until now, have just been making vocal and guitar recordings using an Audio-Technica AT2035 and an Edirol FA66 audio interface with Reaper. However, I’ve been playing the piano a lot lately and would like to incorporate that. I have access to an old upright that’s in the corner of my mum’s living room. How can I achieve the best recording of the piano? Will I need different equipment?
Fiona McKay, via email
SOS Editor In Chief Paul White replies: 
There are many different ways to mic the upright piano, but in a domestic room a pair of cardioid capacitor mics would probably be the best option, as they would exclude much of the room reflection that might otherwise adversely colour the sound. Aim each mic at an imaginary point about a quarter-piano’s width in from the ends of the piano, as that helps keep the string balance even. 
If the piano sounds good to the player, you can use a spaced pair of mics either side of the player’s head, but it is also common practice to open the lid and, often, to remove the upper front cover above the keyboard as well. With the strings exposed in this way, you have more options to position the spaced pair either in front of or above the instrument, and I’d go for a 600 to 800 mm spacing between the mics, adjusting the mic distances as necessary to get an even level balance between the bass and treble strings.

If a piano sounds good to the player, it’s worth trying the recording from just either side of their position, placing the microphones 600 to 800 mm apart. However, it’s also common practice to open the lid of the piano and place the mics above the exposed strings at that same distance apart.
If you’re lucky enough to have a great-sounding room, you can increase the mic distance to let in more room sound or switch to omnis. But in a typical domestic room I’d be inclined to start with the mics around that 600 to 800 mm distance apart. Also listen out for excessive pedal noise on your recording and, if necessary, wrap some cloth around the pedals to damp the sound.
SOS contributor Mike Senior explored this subject in some detail back in April of 2009. It’s probably worth going to www.soundonsound.com/sos/apr09/articles/uprightpianos.htm and giving it a read.

Tuesday, March 19, 2013

Spectrasonics Omnisphere 1.5 Part.1 - NAMM 2011

Q. What pocket recorder should I buy to record my music?

I’m interested in buying a small digital stereo recorder that I can use to record my band in a variety of situations, including rehearsals and at gigs in small venues. It would also be handy to be able to record acoustic guitars and so on for possible use on a demo or in a track. There seem to be loads of products on the market, so what would be the best one to go for?
David Hamilton, via email
SOS contributor Tom Flint replies: 
There’s a great number of pocket-sized digital recording devices that incorporate low-cost condenser mics and exploit the latest generation of SD and Compact Flash cards as a means of storing audio and transferring data. Just about every one of them now has USB connectivity, a speaker for quickly auditioning what has been recorded, data storage capabilities and some basic record and playback processing options. Even handy extras like remote controls, guitar tuners, overdubbing and four-track recording facilities, effect processors and metronomes are becoming standard as the manufacturers battle to outdo each other.
Generally speaking, though, however comprehensive the spec sheet may look, you get what you pay for on some level. Up to about the £200 mark in the UK, the record quality will be more ‘demo’ than professional, even though the latest generation of budget recorders are capable of recording at 24-bit, 96kHz. This is due to the lower-quality preamp circuitry and microphones producing a relatively high noise floor and compromising the audio in other, subtle ways. That said, even the cheaper ones are still capable of making surprisingly well-balanced recordings, and a standard feature is an external mic input supplying some level of phantom power, so there is the option of hooking up better microphones, albeit at the expense of the pocket recorder concept!

Though, in many areas, you get what you pay for in hand-held recorders, the features of some less expensive models sometimes outdo their expensive counterparts. The flexibility of the Tascam DR08’s adjustable capsules offers more recording options than the higher-quality DR2d, for example.
What is a little curious is that many budget products outdo their high-end counterparts in some areas. Tascam’s new DR-08, for example, has a pair of highly adjustable, independently articulating capsules on the front, offering a range of recording possibilities, whereas the manufacturer’s more expensive and better-sounding DR2d has fixed mics and is only configured for omnidirectional recording. Similarly, Yamaha’s W24 has to be connected to a computer using a USB cable, whereas the cheaper C24 has a more convenient, memory stick-style retractable USB connector. Furthermore, some professional products don’t bother with MP3 or 96kHz recording. In other words, paying more does not necessarily mean extra options or convenience.
Paying more does tend to translate into quality, however, and products priced from about £200 up to £400 in the UK offer much better shielding from handling noise, superior build quality, improved metering and, of course, better mics and preamps. If you can afford it, and the recordings to be made are intended for commercial use, these are certainly the ones to go for.
The problem at this level is deciding which microphone configuration best suits the sort of recording jobs the product is going be used for most often. To record guitars, for example, something with an X-Y (coincident pair) mic configuration is arguably more desirable than other designs, as the setup tends to result in focused recordings with good stereo imaging, so long as the capsules are well matched. Yamaha’s W24 is a good example of a product of this kind, as is Zoom’s H4M, although the latter can also be adjusted for wide-angle recording.
Omnidirectional setups might be a better bet for band rehearsals, though, as the recorder could be mounted on a mic stand in the middle of the room (a metal screw thread is usually embedded in the underside for stand mounting), capturing the sound from all around. Tascam’s DR2d and Sony’s PCM M10 are both designed with omnidirectional characteristics, the latter using electret condenser capsules.
It becomes necessary to pay a little more for products that are capable of both omnidirectional and coincident-pair recording. Tascam’s DR100 and Sony’s PCM D50 are serious professional products that fall into this category, and can be bought for a little under £500 in the UK. Naturally, these also come with a host of other professional features, although, on the down side, they are relatively heavy and large and, therefore, not so pocket-friendly.
For live gigs and use in darkened rehearsals or atmospherically lit recording sessions, a large bright screen, displaying clear metering, is vital. If record levels are set wrongly, the mistake could compromise or ruin a take, so accurate visual feedback is important. It tends to be the mid-priced recorders that supplement the metering with warning LEDs, indicating when clipping is occurring, and some, like the PCM D10, also have green LEDs that illuminate when a level of -12dB is reached. In most cases, these provide extremely useful feedback, particularly for the self-recording musician.
A remote transport control is another very useful thing to have when working in rehearsal spaces and small venues, as it enables someone on stage to discreetly trigger recording from afar. Several recorders ship with remotes as standard; the Yamaha Pocketrak and Tascam DR2d both have remotes with a range of seven metres. Others, like the Sony PCM D10, use cables, which is clearly less convenient.
Looking at the market as a whole, there isn’t one product that is best for every recording situation, so the choice as to which one to buy will have to depend on what it is going to be used for most frequently.  

Spectrasonics Bob Moog Tribute library - Musikmesse 2011

Monday, March 18, 2013

Roland Jupiter-80 - Musikmesse 2011

Q. How should I mount a pair of AKG C414s?

I’ve been trying to use a pair of AKG C414s in a coincident X-Y mode, but am finding it physically difficult to mount the microphones. I’ve seen references to vertically aligned and horizontally aligned methods, but these terms imply different mounting arrangements to me. I’ve also heard reference to a Blumlein technique, but I thought that utilises figure-of-eight polar patterns, whereas I was planning on using cardioid in order to maintain focus. Can you clarify the correct technique for using X-Y with the 414s, please?
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: 
Blumlein is a specific sub-form of a coincident (often referred to as an X-Y) stereo microphone arrangement. Basically, X-Y is normally used to imply a stereo array with coincident capsules, whereas A-B normally means spaced microphones, although not everyone uses these terms in the same way. The physical angle between the two microphones in an X-Y array (the mutual angle) and their polar patterns is not defined in the umbrella X-Y term. More or less any mutual angle can be used, and any directional polar pattern, and it would still be an X-Y array. Blumlein is a very specific form of X-Y array. It uses coincident capsules with figure-of-eight polar patterns, and a 90-degree mutual angle (although this is sometimes eased out to 80 degrees to alter the stereo imaging).
As for the correct X-Y mounting technique, there is only one arrangement for end-fire small-diaphragm microphones, shown below in the first example.

Example 1: This is really the only suitable XY-mounting technique for end-fire small-diaphragm microphones.
However, as you have discovered, mounting side-address large-diaphragm microphones can often be a little more taxing and requires more versatile mounting hardware. Basically, the two microphone capsules have to be mounted such that they are coincident in the horizontal plane, and that means they have to be placed with one directly above the other. In this way, sound wave fronts from any source arrive at both capsules at the same time. Stereo imaging information is captured by the level differences imposed by the polar patterns and the fact that the mics are pointing in different directions; there are no timing differences between the left and right channels. So, ideally, the microphones should be mounted vertically with one above the other, as in the second example,

Example 2: Mics mounted vertically with one above the other.
or horizontally one above the other, as in the third.

Example 3: Mics mounted horizontally with one above the other.
Mounting the mics vertically one above the other generally requires either two stands or the creative use of some guitar clamps, although Microtech Gefell and AEA (among others) make suitable brackets for supporting mics vertically. Mounting the mics horizontally above one another can be achieved a little more easily with a wide stereo bar and some pillars or stacked thread adaptors to hold the mics clear of the bar.
An arrangement that’s often used and is far more convenient, albeit with slightly less imaging accuracy, because the capsules are spaced slightly in the horizontal plane (and will therefore capture some small time-of-arrival differences, as well as the wanted level differences due to the polar pattern), can be seen in the fourth example.

Example 4: This technique offers slightly less imaging accuracy, but is far more convenient.
This format can be achieved with a short stereo bar very easily and, in practice, works very well. With a wider stereo bar to allow greater spacing between the mics, you can easily turn this into an ORTF stereo array (capsule spacing of 17cm with a 110-degree mutual angle on cardioid patterns).
The one arrangement you should never use can be seen in the fifth example.

Example 5: This arrangement should never be used, as one microphone will acoustically ‘shadow’ the other. interfering with stereo imaging.
The problem with this configuration is that each microphone sits directly in the active area of the polar pattern of the other mic, forming an acoustic shadow for high-frequency sounds, which will mess up the imaging fairly comprehensively.

Roland R-26 - IBC 2011

Saturday, March 16, 2013

Q. Where can I get raw files to practise my mixing?

I was wondering where I might be able to find raw tracks that I could use to practise my mixing skills? I’ve searched on Google and the SOS forums and not yet got very far. Ideally, the type of music I’d like to practice on would be blues, rock, punk or metal.
Via SOS web site
SOS Reviews Editor Matt Houghton replies: 
Funnily enough, for the Mix Rescue article in this very issue (page 138), both the artist and Mike Senior have kindly agreed to let us make the entire Reaper project available for download. So not only will you be able to practice mixing on it (the full version of Reaper is free to download and evaluate for 30 days, and it’s cross-platform, which means that everyone can have a go, unless you’re one of the few who are stubbornly sticking to Atari or Linux!), you’ll also be able to take a look inside Mike’s mix and hopefully learn a thing or two in the process.
As for other sources of raw multitrack recordings, I’m surprised you haven’t had more luck with a Google search. Get the search terms right (“multitrack wavs” or “multitrack download”, for example) and quite a few sources seem to spring up, including some commercial artists, such as Nine Inch Nails, who have made material freely available (http://ninremixes.com/multitracks.php), and Peter Gabriel, who has held competitions where he’s made material available for would-be remixers. Good as Google is, trying a different search engine can also throw up some different results.

Some commercial artists have made their songs available to download as raw multitrack recordings, which are perfect for practising mixing. This one — ‘Hyperpower’ by Nine Inch Nails — was originally downloaded for Garageband, but is easily opened and worked on in Logic.
Finally, of course, there are always the potentially rewarding options of tracking some of your own material, working with someone else to track your own material, or getting out and seeing some gigs in the hope of finding a good local band and offering to record them for free!  

Roland R Mix - NAMM 2012

Friday, March 15, 2013

Q. What is that Jimi Hendrix effect?

I’m trying to do something psychedelic with guitars — a bit like the song ‘NY’ by Doves — and I think the same effect was previously used on ‘Voodoo Child (Slight Return)’ by Jimi Hendrix. I have tried messing around with the Leslie and delay effects that you get with Logic 9, but have not even come close. What is that effect?
Via SOS web site
SOS Editor In Chief Paul White replies: 
The sound on that record was almost certainly produced by flanging the whole track. You can get close using a flanger plug-in, though the original effect was created by running two tape recorders carrying copies of the same tape, then adjusting the speed of one of them so that one machine overtakes first of all, then falls behind the other. As the machines weren’t perfectly in sync, the small delays caused phase cancellation of specific frequencies, and these varied as the relative timing between the two delays varied. That’s what produces the familiar ‘whooshing’ sound.

The recognisable sound of Hendrix’s ‘Voodoo Child (Slight Return)’ was created by flanging the whole track. This was achieved with two tape machines carrying the same recording, with the speed of one or both being adjusted throughout.
The tape speed was adjusted either by using the varispeed control on one machine, or by slowing one, then the other machine slightly, by dragging the hand on the supply tape-spool flange. The most impressive effect occurred when one machine caught up with, then overtook the other. As you can imagine, the process was a bit hit-and-miss, as you had to line up both machines so that they’d start at the same time, but it certainly produced a trippy sound.
Flanger plug-ins can process both mono and stereo mixes, but most tend to operate from an LFO and so can sound rather too regular. But if you automate the speed and depth controls to create a pseudo-random effect, it can add an authentic feel. Most flanger plug-ins are also limited in the minimum delay time they can apply, so can’t quite recreate the ‘through zero’ effect of tape where one machine passes the other, though some of the more advanced plug-ins use an additional delay in one side of the signal path to fake this effect. 

Musikmesse 2012: Roland GR55 & Boss GT100

Q. What are the best freeware plug-ins?

There are loads of freeware plug-ins floating around out there now, so I find I’m getting swamped by choices. One site I checked out listed 670 of them! I’d rather not slow down my sessions looking for the perfect delay when just sticking with a good one and working with it would be much more productive. I’ve checked out a few of the ones mentioned in Mix Rescue and have been quite impressed, so I was wondering whether you could give me some further suggestions for a couple for each basic category of plug-in. In particular, I’d be interested in any ‘go to’ freeware choices. I’m on a PC, so VST would be best.
Eoghan Brady via email

Some good freeware and donationware VST equalisers: Cockos ReaEQ, Bootsy Nasty CS, Antress Modern Black Dragon, and DDMF LP10.
SOS contributor Mike Senior replies: 
First of all, you could do worse than just download the ReaPlugs VST suite, which is a big chunk of the Reaper plug-in complement and includes everything you’re after, in one form or another. I’ve done whole mixes with just Reaper’s plug-ins, so I can vouch for their effectiveness. Other particularly worthwhile sets I’ve found are those from Antress Modern (http://antress.er-webs.com), Bootsy (http://varietyofsound.wordpress.com), GVST (www.gvst.co.uk), MDA (http://mda.smartelectronix.com) and Voxengo (www.voxengo.com), which cover a lot of bases between them.
But on to some specific things I like, all of which have proved their worth in the heat of Mix Rescue! For general-purpose EQ’ing, I do like Reaper’s ReaEQ a lot, but for extra colour, try Bootsy’s Nasty series and the Antress Modern emulations. DDMF (www.ddmf.eu) have a great donationware linear-phase EQ called LP10, too. For synth-style filtering, I usually just tend to automate ReaEQ, but Camel Audio’s Camel Crusher (www.camelaudio.com) and Ohm Force’s Frohmage (www.ohmforce.com) have more obvious attitude, if required. As far as dynamics are concerned, ReaComp and ReaXcomp in the ReaPlugs set are, again, good all-round workhorses, but things like Georg Yohng’s W1 (www.yohng.com), Buzzroom’s BuzMaxi 3 (www.x-buz.com), Bootsy’s Density, Jeroen Breebaart’s PC2 (www.jeroenbreebaart.com) and the Antress Modern vintage emulations all get regular use on my projects. ReaGate and ReaFIR are a solid bet for most expansion and noise-reduction tasks, so I’ve never really bothered looking elsewhere.
My freeware fallback for chorus, phaser, and flanger effects is Kjaerhus Audio’s Classic series, and although I could no longer find a web presence for them at the time of writing, it’s still possible to find the plug-ins hosted on other sites via Google. MDA’s Leslie and The Interruptor’s Wow & Flutter (www.interruptor.ch) are cool for general modulation grunginess and I use those a lot. For tremolo/chopper effects, try Tweakbench’s Cairo (www.tweakbench.com) or Oli Larkin’s Autopan and LFO Chopper (www.olilarkin.co.uk). When it comes to distortion/saturation, there’s lots of good stuff and I admit to being a bit of a collector in this respect. Some of my favourites are Bootsy’s Ferric, GVST’s GClip and GRecti, Jeroen Breebaart’s Ferox, MDA’s Combo and Bandisto, Mokafix Noamp (www.mokafix.com), Silverspike’s Rubytube (www.silverspike.com), and Voxengo’s Tubeamp: so much dirt, so little time! For more outrageous grainy and grungy effects, DBlue’s Glitch (http://illformed.org) is a good bet, as are Jack Dark’s outrageous Darkware series (www.gersic.com/plugins/hosted/darkware/darkware.html) and Tweakbench’s Pudding and Sideslip.
The Interruptor’s delay plug-ins are good, as are GSi’s WatKat (www.genuinesoundware.com), Tweakbench’s Maelcum and GVST’s GDuckDelay. That said, I tend to use ReaDelay for basic delay requirements most of the time. Smart Ambience is a great functional reverb demo, but Christian Knufinke’s SIR (www.knufinke.de/sir/sir1.html) with impulses from Echo Chamber (www.memi.com/echochamber/responses/index.html) takes the cake for me in the freeware reverb department. For stereo image adjustment and M/S processing, my clear favourites are Voxengo’s MSED and Flux’s Stereo Tool (www.fluxhome.com). The latter has one of the best stereo vectorscope displays I’ve encountered anywhere. Speaking of displays, Roger Nichols’ Inspector (www.rndigital.com) was my metering and spectrum-analysis plug-in of choice for a long time, although Voxengo’s SPAN is also good. I tend to use Schwa’s payware Schope instead for most things these days, however. And speaking of Schwa (www.stillwellaudio.com), they have a great freeware bitscope plug-in called Bitter that can be handy for digital troubleshooting. The TT Dynamic Range Meter is great if you’re interested in the mastering ‘loudness wars’; you can get it free on request via the Brainworx site (www.brainworx-music.de).
Finally, here’s a couple of odds and ends. Although I’ve yet to come across a decent, simple, freeware pitch-shifter, if you’re after freeware pitch correction, look no further than GVST’s GSnap, which is pretty effective and has seen use in a number of Mix Rescues before now. If you’re a fan of Aphex-style psychoacoustic enhancement, also be sure to fire up Stillwell Audio’s exciter, one of the plug-ins available within the ReaPlugs ReaJS host, which does the same kind of thing.  

Thursday, March 14, 2013

Q How much headroom should I leave with 24-bit recording?

I used to record in a very ‘old school’ way; as ‘hot’ as possible without clipping, and always watching the meters like a hawk. But what average levels should I use if I’m working with 24-bit digital audio?
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: 
The basic idea is to treat -18dBFS as the equivalent of the 0VU mark on an analogue system’s meter, and that’s where the average signal level should hover most of the time. Peaks can be way over that, of course, typically kicking up to around -10dBFS or so. Drums, being largely transient peaks, will be kicking up there regularly.
If the material you are recording is well controlled and predictable in terms of its peak levels — like hardware synths tend to be, for example — you could legitimately reduce the headroom safety margin if you really want to. But in practice there is little point.
The only advantage to recording with less headroom is to maximise the recording system’s signal-noise ratio, but there’s no point if the source’s signal-noise ratio is significantly worse than the recording system’s, and it will tend to be that way with most analogue synth signals, or any acoustic instrument recorded with a mic in a normal acoustic space. The analogue electronic noise floor or the acoustic ambience will completely swamp the digital recording system’s noise floor anyway.
Recording ‘hot’, therefore, won’t improve the actual noise performance at all, and will just make it harder to mix against other tracks recorded with a more reasonable amount of headroom. One issue that comes up a lot is the confusion between commercially released media (CD, MP3, for example), which have no headroom margin at all (they peak to 0dBFS), and the requirement for a headroom margin when tracking and mixing.

This diagram shows a comparison of traditional professional analogue console signal levels and SMPTE recommended digital equivalents.
Going back to traditional professional analogue audio systems, the practice evolved of recording signal levels that averaged around 0VU. OK, you could push things a few decibels hotter sometimes for effect with analogue tape, but a level of around 0VU was the norm, and that normally equated to a signal level of about +4dBu (VU meters are averaging meters and don’t show transient peaks at anything like their true level).
Analogue equipment is designed to clip at about +24dBu, so, in other words, the system was engineered to provide around 20dB of headroom above 0VU. It’s just that the metering systems we use with analogue don’t show that headroom margin, so we forget it’s there. Digital meters do show it, but so many people don’t understand what headroom is for, and so feel the need to peak everything to the top of the meter anyway. This makes it really hard to record live performances, makes mixing needlessly challenging and stresses the analogue monitoring chain that was never designed to cope with +20dBu signal levels all the time.
By recording in a digital system with a signal level averaging around -18 or -20 dBFS, you are simply replicating the same headroom margin as was always standard in analogue systems, and that headroom margin was arrived at through 100 years of development for very good practical reasons.
Furthermore, the noise floor of a typical analogue console might be around -90dBu (-100dBu was always the holy grail). That gives a total dynamic range of 90 + 24 = 114dB, which happens to be the same as a typical budget 24-bit digital interface. The very best interfaces and converters are currently providing dynamic ranges of around 124dB, which is the same as the holy grail of analogue gear.
So working with average levels of around -20dBFS or so is fine and proper, works in exactly the same way as analogue, and will generally make your life easier when it comes to mixing and processing.
The old practice of having to get the end result up to 0dBFS is a mastering issue, not a recording and mixing one. It is perfectly reasonable (after the mix is finished) to remove the (now redundant) headroom margin if that is what the release format demands.
A sensible headroom margin is essential when tracking, to avoid the risk of clipping and allow you to concentrate on capturing a great performance without panicking about the risk of ‘overs’. A similar margin is also required when mixing, to avoid overloading the mix bus and plug-ins (yes, I know floating-point maths is supposed to make that irrelevant, but there are compromises involved that can be easily avoided by maintaining some headroom!).
Once the mix is finished, the now redundant headroom can be removed, and that is a standard part of the mastering process for digital media like CD and MP3.