Q & A

I'm too young to have experienced the first wave of FM synthesis, but I find those sounds very interesting. I was considering buying Native Instruments' FM7 soft synth, but I've noticed that many virtual analogue soft synths claim to offer FM capabilities, and provide virtual analogue besides. Is there any advantage to the extra expense of something like the FM7?

Richard Ledbetter

Native Instruments' FM7 soft synth is equipped with six operators, each with a 31-stage envelope which can be edited from this screen. Native Instruments' FM7 soft synth is equipped with six operators, each with a 31-stage envelope which can be edited from this screen.

SOS Contributor Craig Anderton replies: If your interest in FM synthesis is at all serious, then yes, there are advantages to dedicated FM soft synths like Native Instruments' FM7. But to understand why, it might first be helpful to explore some FM synthesis basics.

The basic 'building block' of FM synthesis is called an operator. This consists of an oscillator (traditionally a sine wave, but other waveforms are sometimes used), a DCA to control its output, and an envelope to control the DCA. But the most important aspect is that the oscillator has a control input to vary its frequency.

Feeding one operator's output into another operator's control input creates complex modulations that generate 'side-band' frequencies. The operator being controlled is called the carrier; the one doing the modulation is called the modulator. The sideband frequencies are mathematically related to the modulator and carrier frequencies. (Note that the modulator signal needs to be in the audio range. If it's sub-audio, then varying the carrier's frequency simply creates vibrato.)

Operators can be arranged in many ways, each of which is called an algorithm. Basic Yamaha FM synths of the mid-80s had four operators. A simple algorithm suitable for creating classic organ sounds is to feed each operator's output to a summed audio output, in which case each operator is a carrier. A more complex algorithm might feed the output from two operators to the audio output, with each being modulated by one of the other operators. Or, Operator 1 might modulate Operator 2, which modulates Operator 3, which modulates Operator 4, which then feeds the audio output. The latter can produce extremely complex modulations. Sometimes feedback among operators is also possible, which makes things even more interesting.

So more operators means more algorithm options. The Yamaha DX7 had six operators and 32 fixed algorithms; FM7 actually offers considerably more power because it lets you create your own algorithms.

A virtual analogue synth that offers FM typically allows one oscillator to modulate another one, essentially serving as a primitive two-operator FM synth with one algorithm. While this is enough to let you experiment a bit with simple FM effects, the range of sounds is extremely limited compared to the real thing. You'll be able to get some nice clangy effects and maybe even some cool bass and brass sounds, but don't expect much more than that unless there are multiple operators and algorithms.

I'm currently mixing a full funk band recording and have borrowed a pair of Behringer Truth B2031 monitors to do it. Everything sounds great to me, until I switch over to my Technics hi-fi to check it and it loses all bass definition and sounds muddy. When I check it again using a pair of Beyer DT100 headphones, things sound disjointed, with some things sounding miles away and others really close. Is this my lack of experience in mixing, or is there something I'm missing? How can I find a compromise?

SOS Forum Post

When using small monitors like Behringer's Truth 2031s for mixing, it's worth comparing the results on domestic hi-fi speakers and headphones. When using small monitors like Behringer's Truth 2031s for mixing, it's worth comparing the results on domestic hi-fi speakers and headphones.

SOS Reviews Editor Mike Senior replies: I've used Behringer Truths before, and I've found that the port sounds quite resonant (this is done to achieve a flatter frequency response, but at the cost of time-domain problems), which makes it difficult to judge the bass and lower mid-range with any great accuracy. For example, I've found that the difference between the mixes I've done on the Behringers and those I've done with a Blue Sky Pro Desk system (which has no ports at all) has been fairly dramatic. Admittedly, there's a price differential here, so it's not quite a fair comparison. However, bear in mind that the famous Yamaha NS10s and Auratones were both unported. But as with all monitors, it's essential to listen to lots of commercially produced CDs that you are familiar with played through them so that you can begin to get used to the particular way they emphasise and de-emphasise certain frequencies.

The best advice I can give you would be to use more than one pair of speakers when mixing. I've been getting results which translate pretty well on most systems by alternating constantly between the Blue Skys and a pair of £30 multimedia speakers (as well as checking occasionally on a good pair of Sony headphones). Although you'll never get a completely accurate representation of the sound from any single system (not at this price level, anyway), the constant switching between systems really helps give you a good overall idea of the way the mix is working. If I had the space to set up even more sets of speakers I would, as moving to a dual-system monitoring setup has made a dramatic improvement to my results. I've heard it said that Steve Power (of Robbie Williams fame) uses as many as half-a-dozen monitoring systems while mixing.

Another thing to think about when mixing is the acoustic environment that you and your monitors are in; this is the most common issue we encounter on home studio visits for our Studio SOS features. Just a few panels of acoustic foam or a couple of bass traps can make a world of difference. I can't actually fix any acoustic treatment permanently into my home setup (my wife wouldn't let me!), but I still insist on putting up foam-covered boards while I'm mixing, because it makes such a difference to the usability of the system.

As to the 'disjointed' sound you hear from your headphones, the problem probably has to do with your use of effects. If things seem too far away, then you're likely to be using too much reverb or delay, or just the wrong patch. If they seem too close, then add in a bit more of these effects. It could also be that you're compressing too little or too much. Only a bit of experimentation can really help out here, but check out some of SOS's past articles on reverb and delay for some tips. It sounds like you might also be going overboard with panning to make up for mix problems. I've always considered it a bit dangerous to rely on panning to separate your sounds in the mix, because of the fact that so few real-world listening systems actually exhibit stereo performance — even most apparently stereo systems often don't deliver stereo to the listener because of the positioning of the speakers or the listener. If it's taking panning to sort out a mix problem, I usually try to reassess the arrangement instead. Something just has to go, or be filtered or something.

When approaching a mix, it's often a good idea to start by mixing the most important musical element of your track first. Often, it's the lead vocal, but it could equally be the drums in dance styles, for example. If, for example, your lead vocal and electric piano are the most important instruments in a track, then it makes sense to process and mix everything else to fit with them, rather than compromising the lead instruments' sound to fit them around all the secondary stuff. The best mixes I've done have closely adhered to this principle, and my biggest mixing mistakes tend to occur when I stray from it. I also try to start mixing with the climax of the song first (the final choruses in a pop song, for example), so that I can pace the rest of the song and keep it from getting too big too early.

I'm trying to rescue a fairly limp kick drum sound, which is the result of poor mic placement when it was recorded. What's the best way to turn a dull thud into a tight, clicky thrash/black metal kick sound? I'd rather avoid using the original recording to trigger a sample, and, so far, I've been using Sound Forge to multi-band-compress the sound, then EQ'ing it. Can you suggest any other ideas?

SOS Forum post

The SPL Transient Designer is a useful tool for restoring definition and punch to lifeless drum sounds. The SPL Transient Designer is a useful tool for restoring definition and punch to lifeless drum sounds.

Reviews Editor Mike Senior replies: My secret weapon for all such cases is the SPL Transient Designer. This unit can increase the attack and release phases of drum sounds pretty much independently of the level of each hit. Feed the kick signal to two (or more) desk channels and Transient Design the hell out of the additional channels to get the attack you need — in your case you might need to dial the Attack control up to maximum. Next, EQ (and even distort) the hell out of the multiple processed signals to concentrate the attack elements in the frequency areas you need, and then mix them back in with the main sound. This technique gives any drum sound attitude.

If you haven't got a Transient Designer, then try Emagic Logic's Enveloper plug-in (see this month's Logic Notes for more on how), the Wave Designer patch in Behringer's Virtualiser, or the Envelope mode in the TC Electronic Triple*C as other potential options, although matching the phase between the original sound and the mults when you mix them back together may be a headache in these cases.

I have recorded an acoustic guitar into Cubase SX, both with a mic and DI'd. As I had expected, there are some phase issues. Having read a relevant SOS article, I zoomed in as much as possible on the two waveforms and tried to align them so that they are in phase. The problem is that the two waveforms are almost completely different! There is no way that I can align them! What can you do in such a case?

SOS Forum Post

Technical Editor Hugh Robjohns replies: The two waveforms will be very different, so trying to find some point with which to align them in the middle of a 'strum' will be difficult. Try lining up the beginning of a strum instead, or alternatively, take a look at the review of the Little Labs IBP on page 144 for another way to tackle this problem.

However, the very different tonal characteristics of these two sources, and the inherent timing variations between the mic pickup and DI as the guitar swings back and forth slightly during the playing will mean that phasing is likely to persist as a problem unless the signals are a) panned apart (although listening in mono will bring the phasing problems back again) or b) one of the sources is kept very low in the mix.

Perhaps you should really be asking yourself why you want to mix these two sources together in the first place. What does one source lack that you think the other brings to the party, and could you achieve that desired sound by using a different mic, positioning it differently, or processing the DI signal in a different way?

I find it always pays to follow the KISS principle ('Keep It Simple, Stupid!'), and mixing two signals from the same source is always going to make life harder than it needs to be!

I've just been looking at buying a cheap condenser mic and I was wondering how useful it is to have the low-frequency cut and a 10dB pad options that are on some mics. I assume the low-frequency cut is there to cut out bumps and rumble from knocking the mic stand. Is this still a problem when using a shockmount? How essential is it to have these facilities on a cheap mic?

Nick Bramwell

Low-cut and 10dB pad switches, as found on the Groove Tubes GT67 and GT55, are a useful feature, though not essential. Low-cut and 10dB pad switches, as found on the Groove Tubes GT67 and GT55, are a useful feature, though not essential.

Technical Editor Hugh Robjohns replies: The 10dB (or sometimes 20dB) pad on capacitor mics is intended to prevent the head amplifier (the part of the mic which amplifies the signal picked up by the diaphragm) from overloading when the mic is placed in front of very loud sound sources. Obviously, if the head amp overloads there is nothing you can subsequently do at the desk or preamp to rectify the problem. If recording loud sources like close-miked drums or brass instruments is something you intend to do, then a pad switch will be very useful.

The low-frequency cut is a more complex matter. Different manufacturers intend this 'low-cut' filter to serve different functions. Some do indeed provide a high-pass filter intended specifically to reduce low-frequency vibration and rumbles. Again, in severe situations, low-frequency energy picked up mechanically by the mic can cause overloads in the head amplifier, and hence building the filter into the mic amp circuit can be very beneficial. In more typical situations, though, the HPF on the desk, preamp or recording channel is usually as effective.

However, it is worth being aware that some manufacturers provide a 'low-cut' filter which is intended instead to help compensate for the rise in bass energy when a source is close miked — the proximity effect. In this situation, the 'low-cut' filter will not be so effective in removing mechanical rumbles and the like. So, it is worth finding out what kind of 'bass-cut' filtering is provided — it should be listed in the manufacturer's specs.

If you're using a decent elastic suspension shockmount, you may well find that you don't need to use the HPF. There's a good case for not using it unless you have to — unlike head amp overloads, which you're stuck with, the decision to filter out low frequencies can be made later on, should it be necessary.

How does a converter actually make an analogue signal into a digitally represented signal? What are a converter's components that enable such A-D conversion?

SOS Forum Post

Though gearheads rarely lust after them in the way they do with other types of equipment, A-D converters, like these Apogee Rosettas, are arguably the most important tool you possess when working in the digital domain. Though gearheads rarely lust after them in the way they do with other types of equipment, A-D converters, like these Apogee Rosettas, are arguably the most important tool you possess when working in the digital domain.

Technical Editor Hugh Robjohns replies: There are three steps: filtering, sampling and quantising.

First, the audio bandwidth of the analogue signal has to be defined using a low-pass filter called an anti-alias filter. This is to make sure that the signal contains no frequencies higher than slightly less than half the sampling rate. So, in the case of a 44.1kHz sampled system like CDs, the audio bandwidth would be restricted to about 21kHz.

Next, the signal is sampled at the appropriate sampling rate. In other words, the instantaneous voltage of the analogue signal is measured at regular time intervals. This stage is essentially turning the continuously varying audio waveform into a series of discrete snapshots, in much the same way that a film camera stores continuous live action as a series of still frames.

These individual audio snapshot voltages are then 'quantised.' This means measuring the fixed analogue voltage which represents the amplitude of the waveform at the moment the sample was taken, and describing its value with a binary number which can be handled by the rest of the digital system. The output from the quantiser is the digital signal that is then recorded, transmitted or used in whatever way is intended.

A two-bit system can count four discrete levels or voltages, which is obviously pretty crude. A 16-bit system can count 65,536 different levels and a 24-bit system can count over 16 million levels. Obviously, if the maximum size of the original analogue waveform remains the same, then the greater the number of bits used in the quantiser, the smaller the voltage change that can be represented. This translates to a lower noise floor (or greater dynamic range). So a two-bit system has a dynamic range of about 12dB, while a 16-bit system has a dynamic range of about 96dB and a 24-bit system theoretically reaches about 140dB.

I'm going to be adding some woodwind parts to a track, recording flute, clarinet, alto and tenor saxophones one at a time. Can you offer any advice on how to mike them up?

Martin Brady

Technical Editor Hugh Robjohns replies: Unlike brass instruments, which emit sound solely from the bell and are therefore highly directional, the woodwind family emit sound from the entire length of the instrument, as the player covers or uncovers the finger holes, effectively lengthening or shortening the vibrating column of air within the instrument and changing the pitch of the note being played. As a general rule, therefore, you need to place the mic in such a way that it can 'see' the whole thing. This will usually mean a position about 18 inches away and about half the way down the length of the instrument. As always though, experiment and listen to find the optimal microphone position and angle for that combination of instrument, player and room. A condenser mic should give you the best results.

So for the clarinet, start about 18 inches away and at a height which puts the microphone about half to two-thirds of the way down the body of the clarinet. Try to stop the musician from swaying about, and adjust the distance, height and angle of the mic to optimise the tone and body of clarinet and minimise the key action and wind noises.

The flute presents a slightly different case, as it is held horizontally rather than vertically. Start by positioning the mic level with the flute's horizontal plane, halfway along its length and 18 inches away. Classical flute playing is often recorded with the mic placed slightly above the horizontal and at a distance of three or more feet away, while for more modern styles, the mic is sometimes placed less than a foot away for a more breathy, jazzy feel. Again, you should experiment to find the sound you're looking for. If you're recording in a small room with poor acoustics, putting the mic more than two feet away may mean you pick up more of the room sound than you want.

For the saxes, try a large-diaphragm condenser placed about 18 inches away and 'looking' roughly just below the halfway point of the body of the sax. You could also use a large dynamic mic for a slightly fuller, softer sound. Angling the mic downwards towards the upper rim of the bell can help to reduce how much noise from the keys is picked up — some consider it to be an intrinsic part of the instrument's sound, while others find it unsatisfactory.

I'm looking for a decent stand-alone Leslie-style rotary speaker effect, mainly for use with keyboards, but also for guitars. Can you offer any suggestions?

Michael Reed

The Little Lanelei is a miniature, single-cone rotary speaker designed for recording. The Little Lanelei is a miniature, single-cone rotary speaker designed for recording.

Staff Writer David Greeves replies: The Leslie rotary speaker uses relatively simple mechanical principles to produce a sound that is very hard to reproduce accurately by other means. Sound from the treble driver is dispersed through a pair of horns, pointing out in opposite directions from the central pivot point around which they rotate (although one of the horns is actually a dummy, included to balance the rotating pair). The bass driver, meanwhile, points straight down at a rotating drum which is scooped out on one side to allow the sound to escape. As these two sound sources spin around, they produce not only a tremolo effect (amplitude modulation) but also vibrato (frequency modulation) and phasing thanks to the Doppler effect, the same principle which explains why the pitch of an ambulance siren or car horn seems to rise and then fall in pitch as it speeds towards you and then away. Add to this the fact that the bass rotor is uni-directional and the treble horns are bi-directional, the differing acceleration and deceleration speeds of the two rotors, the build of the Leslie cabinet itself and the particular grungey overdrive that a vintage Hammond/Leslie combination is capable of, and you start to see why 'that sound' is so difficult to reproduce. That's not to say that there aren't some excellent hardware rotary speaker effects out there, and any of the following would be worth a try to see if they meet the needs of your setup.

Dunlop (www.jimdunlop.com) produce a couple of rotary effects. The Rotovibe pedal is mainly intended for guitarists who are after a Jimi Hendrix-style Leslie effect. It looks like a wah-wah pedal, with a single knob on the side to alter modulation depth while the expression pedal controls mod speed, a useful feature for live performance. Dunlop's UV1 Uni-Vibe offers a broader range of sounds, featuring tremolo and chorus modes with adjustable speed, volume and intensity. The optional UV1FC foot controller allows remote control of mod speed and bypass switching.

Dunlop's Uni-Vibe takes its name from the original Uni-Vibe produced by Roger Mayer, an effect used extensively by Hendrix, Robin Trower and many others. The original model is no longer in production, but Roger Mayer (www.roger-mayer.co.uk) produce an updated version, called the Voodoo-Vibe. This Class-A, all-analogue guitar effect features chorus, vibrato and tremolo modes and controls for modulation speed and range (there are three sine-wave and three triangle-wave speed ranges), intensity, symmetry and bias. It also supports an external speed control pedal. Though its earliest ancestor was originally developed as a rotary effect for the organ, the Voodoo-Vibe can produce a range of experimental and psychedelic effects that stretch far beyond Leslie emulation. You might even be able to get away with using it on vocals as well as guitars and keys.

A rear view of the Leslie 122XB cabinet, with panels removed to reveal the treble horns (top) and foam bass rotor (bottom) responsible for the distinctive Leslie sound. A rear view of the Leslie 122XB cabinet, with panels removed to reveal the treble horns (top) and foam bass rotor (bottom) responsible for the distinctive Leslie sound.

Hughes & Kettner produce a well-specified rotary effect pedal called the Rotosphere (see www.hughes-and-kettner.com), which features stereo inputs and outputs and high-voltage valve circuitry for tube drive. Authenticity is high on the agenda, with two-speed operation (the 'slow' and 'fast' speeds can be set), independent acceleration and deceleration settings for the (simulated) treble and bass rotors and an emulation of the breaker circuit, a popular modification which allows the player to disengage the motor, allowing the rotors to slowly come to a stop. When you take your foot off the breaker switch, the rotors slowly spin back up to speed. The Rotosphere is designed for both guitar and organ.

Another such effect is the Spin II from Italian organ manufacturers Voce (www.voceinc.com). This stompbox, coloured an eye-catching orange, is a stand-alone implementation of the rotary simulator from Voce's keyboards. The input can be switched to suit guitar or line-level sources and the stereo output can be turned down to feed a guitar amp. There are bypass and fast/slow footswitches and controls for the fast and slow rotor rates and rotor acceleration.

If none of these pedals will really give you the sound you need, and you've plenty of room to spare in your studio, perhaps you should consider buying a second-hand Leslie cab — I saw a solid-state Leslie 760 advertised for £150 recently, which is less than the list price of some of the effects I've mentioned above. Be aware, though, that Leslies of different vintages use completely different connection and power standards, so even if you have a Hammond organ, or a modern Hammond-modelling keyboard with a Leslie output (like Roland's VK8), it's not the case that any old Leslie will do. If you can get hold of a Leslie Combo Preamp (and I wish you luck), you'll be able to put just about anything through a Leslie cab. This handy chrome wedge-shaped pedal features bypass, fast/slow and breaker switches, provides power to the Leslie's motors and offers a quarter-inch jack input. Once again though, you'll need to match the right model of Leslie Combo Preamp with the right model of Leslie cabinet. Trek II Products in the USA (www.trekii.com) produce a modern equivalent. There's a wealth of information on Leslie maintenance, modification, and compatibility at web sites like www.theatreorgans.com and www.dairiki.com/hammondwiki where you'll find links to many other places of interest on the Net.

A final option to consider is the Little Lanilei Rotary Wave speaker (www.songworks.com), which was reviewed in SOS January 2001 (www.soundonsound.com/sos/jan01/articles/sessentials.htm). This stylish rotary speaker, available in 65W/10-inch and 35W/6.5-inch versions, features a single rotor and a speed control. While you couldn't expect it to reproduce the sound of a Leslie at full blast, it's also much smaller and lighter, and you can plug an external amplifier into it via quarter-inch jack, making it an easy way to add a subtle rotary effect to guitars, keyboards, vocals and just about anything else.

I found an SOS article on stereo mixing from October 2000 on your web site which explains how you can increase perceived stereo width by using extra mixer channels out of phase. However, the diagram is missing. Can you explain how it's done?

SOS Forum post

Editor In Chief Paul White replies: Diagrams for older articles aren't published on the Web, but we've reprinted the one you're referring to below. This is a simple and effective technique for widening the stereo image, provided you have a couple of spare mixer channels. First, you need to set up your mixer so that both the left and right stereo signals feed two mixer channels each. You can either do this using a couple of 'Y' leads to split the signals, or by using channel insert sends to feed the secondary left and right channels (as shown in the diagram). Ideally, the mixer should have channel phase-reverse buttons, though if you have balanced line inputs and no phase buttons, you can instead reverse the phase by switching the hot and cold cores on your 'Y' cables so that one of the left inputs and one of the right inputs is reversed.

To set up the effect, pan the main left input hard left and the main right input hard right as normal. Next, pan the secondary left input hard right, but switch in phase invert on its mixer channel or swap the hot and cold cores. Similarly, take the other right input, pan it hard left and invert its phase. In the diagram, the secondary, out of phase left and right channels are placed on opposite sides to the main left and right channels, so that, from left to right, the channels are R2, L, R and L2. Start with the faders down on the two out-of-phase, 'wrong-side' panned signals and then gradually bring them up to add stereo width. If you go too far, you'll notice the centre signals becoming weaker and a hole appearing in the centre of the stereo field, so don't overdo it.

What is the difference between audio and data CD-Rs? Surely digital audio is just data — a string of ones and zeroes in an encoded file. So why do we need different CDs for audio?

SOS Forum Post

Technical Editor Hugh Robjohns: There are two fundamental types of CD-R blanks: the standard type and the Consumer Digital Audio type. The latter is intended for consumer stand-alone CD recorders which require a special code within the blank 'groove' of the CD-R disc before they will work properly. These discs are slightly more expensive as there is a 'copyright' levy raised from sales of these discs.

Going back to the standard types of CD-R blank, you may still find some types being marked as 'optimised for audio applications'. These are generally higher-quality discs, and usually designed for use in relatively slow burners (below 24x speed).

The point is that when recording audio on a CD-R disc with the intention of playing it back afterwards on a consumer CD player, the data has to conform with the 'Red Book' standard, which takes its name from the 1980 document issued by Sony and Philips detailing their new medium's physical and technical specifications, which, it's said, was bound in a red cover. The Red Book standard has relatively simple (by modern computer standards) error-correction facilities. Hence, a reliable recording really does need good quality blanks to ensure the inherent physical error rate of the disc itself is very low. The faster the recording speed, the more likely errors in the recording process become too. This is another reason why CD-Rs optimised for audio will not support write speeds above 24x.

In contrast, a computer file, such as an audio WAV file, is protected by much more sophisticated error-correction facilities. As a result, these kinds of data file will generally survive better on poorer-quality CD-R blanks.

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