Saturday, December 28, 2013
Tips & Tricks
Technique : Effects / Processing
The way you use effects and processors can make or break a mix. Paul White offers 20 useful tips to help you get it right first time.
Recording can be fun, but for me, the most rewarding part of any project is doing the final mix. It's at this stage of the proceedings that effects and signal processors can be used to turn a simple recording into a major production -- but it's also easy to overdo things and spoil the end results. This month I've put together 20 easy-to-remember tips that will allow you to control your effects units rather than vice versa. And so, without further ado and in no particular order of importance:
1. Reverb creates the illusion of space, but in doing so it also 'smears' the stereo localisation of the original sound source, just as it does in real life. If you want to maintain a specific stereo placement for one or more sounds in a mix, consider using a mono reverb effect and panning the reverb to the same position as the original dry sound.
2. Reverb is very useful for making vocals sound more musical and for making them sit with the rest of the mix, but adding too much will have the effect of pushing the vocals back, rather than allowing them to take front position. Experiment with pre-delay values of 60-100mS to help counter this, and also try using a reverb patch that has a lot of early reflections, as these help reinforce the dry sound. You can learn a lot from listening carefully to records you like to see how much and what type of reverb is used. Often it's rather less than you think.
3. Bright reverbs can flatter vocals, but may exaggerate sibilance. As an alternative to de-essing the vocals, try instead de-essing the feed to the reverb unit, so that sibilance is removed before the reverb is applied.
4. Reverb is probably the most important effect in the studio, so don't compromise by using a low-quality software reverb plug-in just because you're short of processing power. Use a good external hardware reverb unit if you have one, otherwise choose a more powerful software plug-in to treat the vocal track in non-real time. This may involve off-line processing or doing a real-time 'bounce to disk' of the vocal track in isolation, via the plug-in.
5. Vocals almost always require compression, but rather than doing all the compressing at the recording stage, apply a little less compression than you think you might ultimately need, then add further compression when you come to mix. This dual-stage process ensures you don't record an overcompressed sound, whilst still allowing you to even out the level of the recorded signal.
6. Compressors bring up low-level noise just as effectively as they do low-level signals, so try to gate the signal prior to compression when you're mixing. Also, use no more compression than you need, or the signal-to-noise ratio may be compromised unnecessarily. However, it's usually unwise to gate the compressor input during recording for the reasons explained in the next tip.
7. Avoid gating during recording if at all possible, as a badly set gate can completely ruin an otherwise good take by chopping out low-level sections of the wanted audio. Instead, gate during mixing, when you have the chance to reset the parameters and try again if it doesn't work out first time. A further benefit of this approach is that any noise, crosstalk or spill accumulated during recording will also be gated out.
8. Always gate signals prior to adding reverb if you can -- gates can easily chop off the tail end of a long reverb. Furthermore, if you add reverb or echo after gating, any minor gating artifacts may be completely
|"It's at the mix stage of proceedings that effects and signal processors can be used to turn a simple recording into a major production -- but it's also easy to overdo things and spoil the results."|
9. Don't always set your gate to fully attenuate the signal when the gate is closed. In some situations, it may sound more natural if a low level of background sound is still audible between wanted sounds, and when working with drums, you'll find the gate opens faster if the range control is set to around 12dB rather than to maximum.
10. Single-ended noise-reduction units (the type that work by applying level-dependent top-cut) can be very useful in reducing the perceived level of hiss during material where there are no silences that would allow a gate or expander to operate. However, make constant A/B comparisons to ensure that there's no obvious top-end loss when the unit is switched in. If there is, lower the threshold slightly until you get an acceptable compromise between high-end loss during low-level passages, and audible hiss. As with gates, applying reverb after dynamic filtering may help disguise any side-effects as well as safeguarding the reverb tails from being truncated.
11. Don't add long reverb to bass sounds unless you have an artistic reason to do so, as this tends to muddy the low end of the mix. If you need to add space to a kick drum, try a short ambience program or a gated reverb as an alternative. If you are in a position where you need to apply reverb to an entire drum mix, roll off the low end feeding the reverb for a cleaner sound.
12. Chorus is a useful effect for creating the illusion of space and movement, but it also tends to push sounds back in the mix, rather as reverb does. If you need a sound treated with chorus to stand out in a mix, try either panning a dry version of the sound to one side and a chorused version to the other, or ensure that the song's arrangement leaves plenty of room for the chorused sound.
14. Equalisation is often used as an alternative to getting a sound right at source, but the result is seldom as satisfactory as doing things properly. Nevertheless, on occasions where equalisation is necessary, applying cut to the over-emphasised frequencies rather than boost to weaker ones generally results in a more natural sound, especially with vocals and acoustic instruments. This is especially true of in-desk equalisers or budget parametrics, as they often sound nasal or phasey when used to boost mid-range sounds.
15. Sounds can often be made to sit better in a mix by 'bracketing' them with high- and low-pass filters so as to restrict their spectral content. Many console EQs don't have the sharp filters necessary to do this, but the side-chain filters fitted to many gates are often ideal for the job. Simply set the gate to its side-chain listen mode, then use the filters to shave away unwanted high and low frequencies. Acoustic guitars often work better in a mix if the low end is rolled off in this way, though the high end can usually be left alone.
16. When setting up a mix, try to get the mix sounding close to right before you add any effects or signal processing. Once you've got this right, add further vocal compression if needed and also apply just enough reverb to make the vocals sit comfortably with the backing track. When you're happy with the overall timbre and balance, adding effects for 'effect' should be easier. Remember that, in most cases, effects are there just to add the final gloss -- they won't compensate for a poor balance or bad basic sounds.
17. Still on the subject of effects in the mix, don't be tempted to hide poor playing by heaping on more effects, it never works -- take it from someone who's tried everything at one time or another! However, thanks to the wonders of modern technology, slightly imperfect vocal pitching can be tightened up almost magically using pitch-correction processors, such as Antares' Autotune software or ATR1 hardware.
18. Go easy when using enhancers to treat complex signals such as a whole mix as it's very tempting to go too far. Make frequent use of the bypass button to remind yourself just how radically the sound has changed, and if you're adding more than a little high-end enhancement, check the bottom end to see if that needs bringing up to keep the overall mix in balance.
19. Often it's better to enhance just some elements of a mix so as to make them stand out from the rest. The best way to do this is to connect the enhancer to a pair of group insert points, then send all the sounds that need enhancing to that group. Listen carefully to enhanced vocals as the process can often exaggerate sibilance problems.
20. Treatments designed to increase the stereo width of a mix (other than the simple mixing-antiphase-signals-into-the-opposite-channel trick) can have detrimental effects on mono compatibility. Use your console's mono button to check that your mix doesn't lose too much when it's played in mono, as this is important when material is played over mono radios or TVs. Listen to see if the subjective balance or timbre changes by an unacceptable degree. If it does, either use less overall width expansion or leave the main mix elements untreated and only process secondary sounds, such as incidental percussion, sound effects, effects returns and so on.
Friday, December 27, 2013
Tips & Tricks
Technique : Effects / Processing
Antares Auto-Tune is a powerful pitch-correction tool which is already an industry standard for tightening up vocal performances. As Paul White explains, however, it has the potential to do much more...
Antares Auto-Tune is, in my view, one of the few truly innovative musical developments of the past couple of years. Other devices may employ more novel technology, but Auto-Tune provides a practical answer to the very real problem of pitching imperfections, via an easy-to-use box or software package. We've reviewed both hardware and software versions of Auto-Tune in SOS already, though there's a more recent addition to the series in the shape of Auto-Tune LC, a low-cost VST plug-in that offers all the functionality of the hardware version (with the exception of external MIDI control) for under £100. A DirectX version of Auto-Tune is also available for those working on the PC platform, so just about anybody who needs pitch correction can now access it affordably. What's more, at least two other companies are planning hardware or software products that do the same kind of job (indeed, TC Electronic's Intonator is reviewed in this issue of SOS, starting on page 198), so I foresee this being a growth area.
The main function of Auto-Tune is to tighten up vocal pitching, something it does very well, but I've since experimented with the system and found that it can also be used to produce creative effects and to improve monophonic instrument performances. This article is not a re-review of Auto-Tune, but rather a look at some of its less obvious applications.
Before setting out to use Auto-Tune, it's useful to have a general idea of how it works. In order to correct pitch, a system such as Auto-Tune needs to be able to read the pitch of the incoming signal. This is only really possible with monophonic sources at present - the algorithm was apparently a spin-off from work in analysing seismological data, and it seems well adapted to tracking the pitch of the human voice.
Once the pitch has been detected, Auto-Tune then uses pitch-shifting techniques to change the pitch of the original sound to match the nearest note in a user-definable scale. It is possible to leave Auto-Tune set to a chromatic scale (ie. one containing all notes), but then it's likely the pitch of a badly sung note could be corrected to an 'illegal' (out-of-key) note because that's what it happened to be nearest to. Far better to input only those notes the piece of music is using - then the worst that can happen is that a horrendously badly pitched note will be corrected to the wrong note in the right key!
Forcing incoming notes to the right pitch is only part of the story, because if that was done too efficiently, you'd end up with a very flat, pitch-quantised vocal performance. Fortunately, there's a slider to adjust the rate at which correction takes place: by adjusting this carefully, natural bends and vibrato are allowed through unaltered, but as soon as the input settles on a specific note for any length of time, it's pulled into perfect pitch. This was demonstrated most impressively at the Frankfurt MusikMesse by demonstrator Gerry Basserman, who used a theremin as the input source. With the tracking set to fastest, the theremin sound was changed so that it would only produce discrete, stepped pitches, just like a keyboard. However, setting a longer correction rate allowed Gerry to play the instrument apparently normally - except that when he settled on a note, it always slid into perfect pitch. As he himself confesses, this made him sound a much better theremin player than he is - though to be fair, he's actually pretty good anyway!
Because Auto-Tune doesn't use any kind of formant correction, any sound pulled far from its original pitch starts to sound unnatural in the same way any conventionally pitch-shifted signal does. In normal use, this isn't a problem, because the usual amount of pitch correction is less than half a semitone, but the effect can be abused in creative ways by deliberately setting large intervals. For example, you could set up a target scale containing only octaves and fifths, then sing a load of made-up 'pseudo-ethnic' nonsense to see what comes out the other end.
In this instance, the voice timbre is transformed proportional to the pitch difference between what you're singing and what the nearest scale note is set to. For example, you can sing a steadily rising tone and there will be no pitch change until you got within range of the next note in the target scale. However, the timbre of the fixed note will change as the pitch shifter works harder and harder to correct it. In this example, the pitch-shifter will attempt to push the pitch of the note down to compensate for you singing higher, so the timbre will take on a darker, bigger quality. Then, when you get within range of the next note in the scale, the timbre will flip as the pitch-shifter tries to push the pitch upwards. Gerry worked a few examples of this type into his demo and I have to say the resulting 'pseudo-world music' was most convincing.
I was recently asked to do an album of music themed around the solar eclipse, and this provided the perfect excuse to use Auto-Tune for something other than vocal correction. For one of the tracks, I wanted to use a wooden North American Indian flute I'd picked up on my travels, but there were two problems - I couldn't really play the flute very well, and the notes it produced were just far enough off concert pitch to be irritating. The first problem I got around by a mixture of practice, bluffing and hard-disk editing, while the second required the software version of Auto-Tune programmed to the same pentatonic scale as the flute. I set a slowish tracking time so all the note bends and trills would get through unaffected and let nature take its course. The result was a perfectly natural-sounding flute part with all the sustained notes nicely in tune - it even surprised me! This was the first time I'd tried Auto-Tune on a source other than a voice, but the tracking algorithms seemed to have no difficulty at all dealing with it.
The next test was to try Auto-Tune on a slowish lead guitar part that featured a lot of 'long' string bends and tremolo arm tricks. Though I don't play guitar nearly as much as I should, my pitching while bending is normally pretty good, but a slowish pitch-correction setting on Auto-Tune really added polish and precision to the performance. On the occasions that I picked double notes, Auto-Tune just ignored me - it only seemed to take effect when I was sustaining a single note, and of course the slow pitch-correction setting allows any vibrato to come through normally. Considering the complexity of a distorted guitar waveform, Auto-Tune tracked it very well with no glitches or warbles. I also conscripted Paul Farrer's help to test Auto-Tune on some cello parts he was recording, and he reported the experiment a great success.
"The more you think about it, the more applications there are for this amazing tool - using it just to polish up vocal tracks really is under-exploiting its true potential."
When Cher featured that distinctive vocoder effect on her 'Believe' single, a lot of us thought that it had been done on Auto-Tune, though as the February '99 SOS article revealed, that proved not to be the case. Nevertheless, you can get a very good approximation of that same effect from Auto-Tune simply by setting the pitch-correction rate to its fastest setting and entering the appropriate scale for the song. I tried it as a bit of a joke and found it uncannily like the real thing - it's a shame that after just one outing, the effect is already a cliché to avoid. A side-effect of using very fast pitch correction is that shallow vibrato is ironed out altogether, while deeper vibrato turns into a trill. Normally you wouldn't want this to happen, but when you're being creative, little accidents like these can produce very usable results. Auto-Tune also has the ability to add delayed vibrato to whatever sound is being processed, so in theory, you can strip out the vibrato from the original performance and replace it with something far more mechanical and precise.
Things to try
Auto-Tune is so effective that I'm always adding to the mental list of things to try out with it. Monophonic slide guitar should be easy enough to fine-tune, but then there's the option of speeding up the pitch correction and turning it back into a chromatic instrument. Set a musical scale and you have the makings of an electric slide dulcimer!
By the same token, those who can play just a little fretless bass, cello or violin might find that their efforts become a lot more musically useful after a trip through Auto-Tune. Certainly Paul Farrer's experiment showed that a reasonably well-played cello part could be made to sound much more precisely pitched without the result appearing to be processed.
One function of the ATR1 (the hardware version of Auto-Tune) I haven't mentioned yet is its ability to shift pitch according to a MIDI input. In theory, this means you can input a MIDI melody and whatever audio input you have fill be forced to fit that melody. With careful setting up, you can get very natural results, but how about striving for the unnatural by making the human voice sing impossible arpeggios or taking a single-note instrument like the digeridoo and forcing it to play a melodic bass line? The more you think about it, the more applications there are for this amazing tool - using it just to polish up vocal tracks really is under-exploiting its true potential.
Tips & Tricks
Technique : Effects / Processing
In the final part of his short series on pushing back the boundaries of effects processing, Paul White explores many different applications of audio filters, as well as exploring the possibilities of granular synthesis. This is the last article in a two-part series. Read Part 1.
One effect that has migrated from the synthesizer world is filtering. The simplest synth filter has a low-pass response with variable resonance at the cutoff point, however, there are a number of other filter types which can be useful, and details of all the different filters can be found in parts four and five our Synth Secrets series, in SOS August and September 1999 respectively. Just running audio through a synth-type filter can be fun, particularly when you experiment with higher resonance settings, but things get more interesting when you manipulate a filter in real time. An envelope is often used to do this, and the trigger that this envelope requires can be derived from the audio signal itself.
The least complicated approach is to use the envelope generator to control the filter's cutoff frequency, and to cause the envelope to trigger whenever the incoming audio signal exceeds a certain threshold. This works well when sounds to be processed are separate and have clearly defined starting points, but is less effective for sounds that overlap, as triggers can become less reliable.
Remember that the envelope above (or a different one triggered in the same way) can also be routed to control a filter's resonance, for extra variety. What's more, other modulation sources can be used in place of an envelope generator. LFOs can be fun in this role, either used free-running or with their rates sync'ed to tempo, and envelope followers can also be used. The latter could control the filter cutoff frequency according to the level of the input signal — in other words, the higher the input signal level, the higher the filter frequency (or vice versa if that's how you want to set it). The sound this creates can often be musically useful, though where the initial sound is modulated in level, the resulting undulating envelope can lead to the filter opening and closing in a unmusical and seemingly erratic way.
Where the incoming sounds are from a sampler or a MIDI synth, then MIDI triggering and control of any audio-processing filter can be employed, and clearly this will work consistently whatever sound is being processed. However, it is important to note that, in a typical synthesizer, each voice will have its own filter while, in the case of an external filter module, all the voices will be processed via the same filter — this means some decision has to be made as to how the single processing filter will behave when a new MIDI trigger is received. Should the filter envelope start again as each new note is played or should it only reset after all keys are released? Some units make this choice for you while others provide switchable functions. Other points to note are that stereo filters are necessary for stereo signals and that filters should be inserted into the signal path, rather than being used in a send/return effects loop configuration.
Stand-alone filters can be used in a number of different contexts to add interest and movement to a sound. An obvious application is to take an otherwise filterless synthesizer (such as a Kawai K1, Korg Wavestation or Alesis Quadrasynth) and treat its output with a filter. It's also very common to use filters to process sample loops and, in this application, the automation offered by many software plug-ins and MIDI-controllable hardware units is very attractive. Complex, tempo-related effects can be created within your sequencer and then copied to as many bars as are required.
More Fun With Filters
We've had resonant high-pass, low-pass and band-pass filters ever since the first analogue synths, and though they're still widely used today, there's a lot more that can be done using more sophisticated filter types. We've already seen how you can use the complex filtering of a vocoder to create interesting new sounds, but that is only the start. There is a great deal of mileage to be had from processing different frequency bands separately (see 'Divide & Conquer' box on page 124)
One particularly complex multi-band filtering effect is available within Emagic's Logic Audio — the Spectral Gate plug-in. Although the documentation doesn't make it clear exactly how the filters are configured, a few minutes playing with the controls demonstrates the range of effects that can be created: everything from the more obvious filtering characteristics to sounds that appear almost resynthesized. It's very easy to create metallic, electronic-sounding textures from quite conventional input material, though, as with all filtering, the more harmonically rich the input, the more interesting the output.
While on the subject, it's also worth trying out the 'Convolve' process in Bias Peak or any other software that offers the facility. This sounds to me like a filter-based effect, and it allows the characteristics of two sounds to be merged in order to produce a new sound sharing characteristics of both. This process is often used in sound design to combine sounds in unusual ways.
breveR & yaleD!
Conventional delay and echo effects are a mainstay of music production, though last month I suggested a few ways to make these more interesting. A further way of adding interest to effects is afforded to us by the recording process, which enables us to make use of negative time. By this, I mean effect sounds which are audible before the sound that they process even starts playing — something that the physical laws of real life don't allow, unless in close proximity to a concentrated source of tachyons! Once something is recorded, temporal rules can be broken. Sounds can be reversed, they can be treated with reverb or delay that starts before the sound itself, or reverse reverb can be added to a 'right-way-round' sound. This latter trick used to be popular during the '70s for music production and is still used extensively in film work to create demonic voices.
When analogue tape was the standard recording medium, reverse reverb was accomplished by playing a tape backwards, feeding the desired track through a reverb unit, then recording the reverb to a spare track. Once the tape was replaced on the machine the correct way around, the reverb track would start playing before the track it was derived from, with the reverb sound's envelope building up slowly in a suitably eerie manner. The same trick can be achieved in a tapeless environment (such as within a MIDI + Audio sequencer) by reversing a section of audio, adding reverb or delay, bouncing the processed result to a new track, then reversing both tracks again. I covered this procedure in some depth in SOS December 1998, for those who'd like to try it. For a more ambitious project, try one of the chopping techniques described last month, using reversed sections of sound or effect for some of the segments.
Of course, the other thing that's very easy to do in a tapeless environment is set up a conventional reverb, record this to a spare track, then slide it a beat or two ahead of the track being processed. This produces reverb that's the right way around, but which still comes before the sound that supposedly created it — you could almost think of it as negative pre-delay!
One of the current buzz words in sound creation is 'granular' synthesis, which simply means taking very small slices of sound, often from very different sources (or from different times within the same source), and then joining them together to create a new sound. The difference between granular synthesis and wavetable synthesis seems to be mainly the length of the individual segments used. Granular segments may only be a few tens of milliseconds in length. Most of the results I've heard from granular synthesis remind me of a day out at the dentist, but for those on the cutting edge of techno, it might be exactly what is needed. I don't know of any granular effects boxes, but it should be possible to fake your own using a sequencer if you have the patience and a suitably sadistic mind. There are also software packages that can help, such as CDP's GrainMill.
At its simplest, you could set up two audio tracks with the level automation programmed to gate the sound on and off very rapidly. This is easiest to achieve using your graphical editor to draw nice neat square waves linked to MIDI continuous controller seven. By arranging it so that one channel is off while the other is on, and vice versa, you should end up with something very crunchy indeed. You should probably be aiming for something like 64 transitions per bar or more, so make use of the copy and paste facilities of your sequencer when creating the granular 'chopping' templates. When you get a good one, save it in your default song so that you can use it with other sounds. If you have a plug-in filter that offers a sample and hold facility, you can get some pretty granular-sounding effects out of it if you can persuade the sample-and-hold rate to go up high enough. If you can get it fast enough so that the individual steps blur into a continuous noise like somebody cleaning their oven with an angle grinder, you've probably got it about right!
Another way to approximate a granular type of effect is to use the manual digital scrubbing facility on a piece of recording hardware or software, and then record the results. Often you'll find there are two scrubbing modes, one that works rather like tape varispeed and another that constantly plays back a tiny loop of audio as you move through the audio file. It's the second one of these you need for real granular-style emulation — just move through a sound slowly and linger on any sections that sound particularly interesting. Again, sample anything that sounds useful.
There are some spectacular granular treatments in Native Instruments' Reactor and Dynamo that work on short samples of sound, so you can load in your own audio clips and mangle them mercilessly before recording the results as a separate audio file. It's more of an instrument than an effects processor, but if you get the results you want, does it really matter? What's more, because VST 2 virtual synths can be automated, you can set up your granular effect within a song and know that it's always going to come back sounding the same each time.
While there may not be as many new effects as we'd like, it's often possible to create some extreme-sounding treatments by either combining existing effects or by making existing effects dynamic in some way. Those reclusive people who design the sounds for sample CDs often set up huge loops of sound processors and delays that feed back on one another, then they record the output and select the best bits. If you have a few pieces of outboard, give this a try using lots of delay feedback so that everything is just below the point of breaking into self oscillation. Sometimes you don't even need an input signal to set the whole thing off!
By creating repetitive changes that happen too fast for the human ear to perceive, it's easy to emulate granular synthesis or complex modulation where the result is often dissonant and mechanical sounding. What counts is that the result is musically useful, and even treatments that result in atonal mayhem can sound good in context, especially when used as part of a rhythm track. Don't rule out cheap effects pedals either, as some of these produce surprisingly musical sounds, even if they don't have the best technical spec. My best advice is to break a few rules. If a box says 'guitar processor', try it out on vocals or drums to see what it does. You have nothing to lose but your sanity!
Thursday, December 26, 2013
Tips & Tricks
Technique : Effects / Processing
Most people are familiar with basic reverb, delay and modulation effects, but what lies beyond? In the first part of a new series, Paul White explores the twilight zone of effects processing. This is the first article in a two-part series. Read Part 2.
It's often said that there's little to be had in the way of novel effects nowadays. Most effects are either standard reverb, delay, modulation or pitch-shift, but that doesn't mean that there aren't other effects to be found lurking in the dark corners of your multi-effects box or software plug-ins folder. Some of the more bizarre effects have been around for years — for example, vocoders, ring modulators and chordal resonators — but the freedom provided to designers by the newer plug-in formats means that more off-the-wall stuff is appearing all the time. The aim of this article is not to concentrate too much on specific products, but rather to explain some of these less common effects types and to make a few suggestions about the ways in which they can be used.
Ring Modulators are intriguing devices designed to process two input signals in such a way that the sums and differences of the input frequency components are generated while the input signals themselves are suppressed. For example, if you were to put in two sine tones at 500Hz and 600Hz, the output would comprise tones at 1100Hz and 100Hz. Conversely, feeding the same 500Hz tone into both inputs would produce components at 0Hz (a silent DC offset) and at 1000Hz (an octave up from the pitch at the inputs). However, the results are only as simple as this when you input pure tones — when harmonically rich sounds are used, all those harmonics contribute to the sum-and-difference process, resulting in a harmonically very complex output. Note that an output will only be produced from a ring modulator when signals are present at both inputs, so if level fluctuations are a problem then it may be worth compressing one or both input signals.
Processing percussion via a ring modulator can be good — use a pitched synth sound for the other input and you'll end up with a metallic, pitched drum part that could form the basis of an experimental electronic song or dance track. Ring modulating different cymbal sounds together is also an interesting experiment, which creates new, electronic-sounding cymbals.
If you want to create new sounds and treatments based upon a basic ring-modulation sound, try combining it with other effects. For example, use a dry sound as one input to the ring modulator and its reverb or delay as the other. You can also further process the ring modulator output using conventional but dramatic effects such as flanging or heavy delay.
Spark's Magic Piano
At one time, vocoders were considered quite esoteric, but nowadays they come built into some multi-effects units — less costly stand-alone units are also fairly common. On top of that, there are some very effective vocoder plug-ins that can be used within sequencers.
Like the ring modulator, a vocoder requires two inputs to generate an output, and to make this process clear, a block diagram is shown in Figure 1. Essentially, the vocoder superimposes the frequency spectrum of one sound (called the modulator) on a second sound (known as the carrier). The way this is achieved is that the frequency spectrum of the modulator is continually monitored using a bank of frequency-spaced band-pass filters, and the information used to control the gains of a corresponding bank of band-pass filters in the carrier's signal path. Thus, as the spectrum of the modulator changes, the carrier's filter bank settings follow it. If a voice is used as the modulator and a harmonically rich musical sound as the carrier, this results in the classic vocoder sound — the voice seems to take on the pitch and timbre of the carrier sound, but the vocal articulation is still recognisable, because of the dynamic action of the filter bank following the continually changing spectrum of the voice. As you might imagine, the more filter bands the vocoder has, the more accurate and intelligible the speech-like element of the output signal.
It is apparent from this description of the vocoding process that you need signals arriving at both inputs simultaneously before you can obtain an output signal. It can also be helpful to compress both inputs to the vocoder, in order to keep the output levels stable. On the other hand, if the modulation input is a voice, you might find that the vocoder is triggered undesirably by breath noises, in which case a gate inserted between the microphone and the vocoder's input will also be an improvement.
The talking synth effect has been used on countless records (for example, 'Blue Monday' by New Order, 'Mr Blue Sky' by ELO and 'Rocket' by Herbie Hancock), but this isn't the only way to use a vocoder. By substituting the vocal input with a recording of background noise in the local pub, and by vocoding this with a rich synth pad, you can create a very organic pad sound with a lot of movement. Similarly, two different synth sounds or samples can be vocoded together to create a totally new sound. If the modulator signal includes dramatic changes, such as a filter sweep, these will be imposed on the carrier. The real key is to experiment, but a point to keep in mind is that, because the end result is created subtractively, the carrier signal needs to be harmonically rich in order to give the filters something to work on. If the carrier is a synth sound, an open filter setting combined with a sawtooth or pulse wave works well.
When vocoders were first developed, it was realised that, while the pitched elements of vocal sounds provided a good modulation source, unpitched vocal components such as 'S', 'F' and 'T' sounds tended to get lost, and so vocal clarity was lost. Different strategies were devised to help with the intelligibility of vocoded speech. One of these was to replace some of the consonant sounds with bursts of noise, but a far simpler method was to add a high-pass filtered version of the vocal input into the output. The latter method works well, because the higher-frequency region of the vocal spectrum contains most of the energy of many vocal consonants, yet without many pitched components. By filtering out everything below 5kHz or so, then adding the remaining high frequencies to the vocoded signal, vocal intelligibility can be improved enormously. If this facility isn't already included in your vocoder, it can be patched up using hardware, or within a plug-in environment that allows series and parallel routing (such as in TC's Spark).
Further sophistication is offered by some advanced vocoders where it is possible to swap around the filter bands, such that the level of the modulator input at one frequency can be mapped to control a filter band at a different frequency. Though few dedicated vocoders have this function and there's no simple way to fake it, implementations are possible using a software modular synth and it can really open up the gates of weirdness.
Almost everyone has tried setting a DDL to a very short delay time and then increasing the feedback. The result is a 'ringing' sound at a pitch determined by the delay time, particularly when it is excited by percussive sounds. For example, a 1mS delay will produce a resonant pitch with a fundamental of 1kHz, 10mS will produce a resonant pitch with a fundamental at 100Hz, and so on. With drums, it produces an effect not unlike playing them in a resonant brick tunnel — check your train timetable, though, before trying this at home!
A similar effect can be created by setting up a band of equalisation with high boost and resonance values — this will ring at whatever turnover frequency you select. If you have a number of bands available then you can set up a number of resonant peaks, which already holds a lot of scope for experimentation. However, if you can automate the frequencies of the resonant peaks, then you can get really creative.
Lexicon used the resonator principle in the PCM80 to create their resonant chord effect, and something similar was used by Alesis in their original Quadraverb. The concept was that MIDI note information from a keyboard or sequencer could be used to tune one or more resonators to specific musical notes so that any input signal passed through them would cause the filters to ring or resonate at musically relevant pitches.
Percussive sounds seem to work best with resonator programs and, because of the effect of the resonators and their musical pitches, the drum sound becomes more abstract and harmonic. Considering how dramatic this effect can be, it's surprising that it doesn't feature on more records, though I know that a number of sound designers use it for creating less conventional rhythm loops. If you have a multi-effects unit with a MIDI controllable resonator, either monophonic or polyphonic, I'd recommend you try it out at least once so that you get to know the extent of its capabilities. It can help to make the effect more obvious if you increase the resonance or feedback, but otherwise I don't have many tips for experimenting with this effect other than 'suck it and see'!
Chopping & Changing
I'll finish off for this month by looking at some of the triggered gating effects that can be set up. It's fairly well known that a gate can be triggered via its side chain to chop up audio in a rhythmic way, but it's sometimes possible to take this concept a little further. A MIDI gate or plug-in is easiest to use for this purpose, as you can feed in a rhythmic sequence of note-messages to trigger it. However, any regular gate with an external key input can be triggered using a fast-attack, fast-release synth tone.
Now let's look at some ways of making rhythmic chopping more interesting. One option I've heard used to great effect on vocals is to set up an even tempo-related chop at around eight chops to the bar, then to use a DDL to delay a copy of the chopped signal so that the repeats fall exactly into the gaps created by the gating of the original part. Figure 2 should make it clear how this is done. What you hear is a kind of chattering effect as the repeated sections are joined up. The intelligibility is, of course, pretty poor, as half the signal has been discarded and the other half doubled up, but it makes for an interesting interlude. Pan the original gated signal and its delay to opposite sides for an even more dramatic effect.
If you're feeling adventurous, you could set up two or more gates triggered in such a way that each gate is only on for certain beats of a bar. Arrange your trigger material so that only one gate is open at a time, but with one of the gates open on each beat. This will require the generation of two or more MIDI note control tracks or, if the gate is being triggered from a synth, you'll need two or more different outputs to feed the gate key inputs. Finally, feed the same signal into all three gates, but then apply a different effect to each gate output. For example, use heavy flanging on the first output, distortion on another and perhaps an envelope-following filter on the third. When the outputs are heard together, you'll hear all the differently effected beats spliced together. Figure 3 shows this technique using three gates. If the gates click during the transitions, lengthen the attack and release times slightly, but otherwise use the fastest settings for the cleanest chopping.
Taking the chopping up idea even further leads us into the murky terrain of granular synthesis and processing, where audio is sliced into extremely short pieces that are then joined back up in different orders to create new sounds. This requires special software, or ingenious use of a sampler —it's not something you can knock up using a multi-effects box, certainly, and in many cases the process isn't even real-time. I must admit that I've yet to hear anything musically worthwhile from processing of this kind, but if I discover anything, you'll be the first to hear about it!
That's probably enough weirdness for this month, but don't relax yet, as there's more to come next time when I'll be concentrating on filtering and 'time travel'.
Monday, December 23, 2013
Technique : Effects / Processing
Modern digital effects units always include emulations of analogue effects such as tape delay and flanging -- but none of them ever seem quite like the real thing. Paul White explains how these vintage effects worked, and offers insight into how our modern attempts could be made more accurate.
We live in a digital age, but you can't go into a studio without hearing stories about how good things use to sound in the old days. But were the old electromechanical effects really that good, or have we fallen foul of the 'nostalgia isn't what it used to be' syndrome? Certainly some old effects were noisy and unreliable, but in most cases, there was an element to the sound that the human ear found especially pleasing in a musical context. Sometimes the reasons are fairly obvious, but other times the magic element remains elusive, nowhere more so than in the case of tape flanging or phasing.
Tape Phasing & Flanging
Tape phasing, commonly known as tape flanging, is a unique effect, and though some digital flangers have managed to approximate it, I've yet to hear a truly convincing emulation. If you've never experimented with tape flanging, the effect is created by running two identical copies of the same recording on open-reel analogue recorders (usually in mono) and then summing the two outputs together via a mixer at exactly the same levels. The two recordings are started together -- a hit-and-miss business at the best of times -- then the speed of one of the machines is slowed slightly by using hand pressure on the tape reel. The idea is not to get so far behind that you can hear a tangible ADT-style delay, but simply to produce a comb filtering effect. Comb filtering occurs by virtue of the addition and subtraction of frequencies that end up being in-phase or out-of-phase as determined by the delay time. Whichever machine is leading is then slowed down so that the delay decreases until the point where the other machine takes the lead. As the relative delay between the two tapes changes and finally passes through zero, the familiar whooshing effect is created as the comb filter sweeps through different frequencies in the source material. And so it continues with the leading machine being slowed manually so that the two recordings drift in and out of phase with each other. Because tape flanging is literally hands on, the effect is different every time. Some of the more sophisticated studios used electronic speed control instead of hand braking to create the effect.
Modern flangers seek to emulate this effect by digitally delaying one signal relative to another (by just a few milliseconds), then modulating the delay time using a low-frequency oscillator, or LFO. To make the effect stronger, some of the output is fed back to the input, which adds resonance to the comb-filtering effect. Note that with tape flanging, no feedback is used. Artificial flanging of this kind sounds different to tape flanging for a number of reasons -- the LFO-controlled modulation is regular, feedback is used to add depth to the effect and the delay between the two signals never passes through zero, as it does when two tape machines are used.
A refinement of this method is to delay one signal by a very small amount (say 5mS), then modulate the delay of the other signal path so that it slowly changes from less than 5mS to more than 5mS. This provides the 'through zero' element of the effect but does nothing to break the regularity of the modulation unless the delay time is adjusted by hand. Furthermore, if feedback is applied, it doesn't create the desired effect, as the feedback-induced resonance will be a function of the whole DDL delay time, whereas the comb-filtering effect itself is related to the difference between the two delay times. A simple setup for through-zero flanging is shown in Figure 1 alongside the original tape-based arrangement.
In theory, it should be possible to emulate tape flanging much more closely by using techniques such as physical modelling. For example, one reason the effect sounds the way it does with analogue tape machines is that analogue machines don't have the precise phase response of a digital system. For example, put a 1kHz square wave into a digital recorder or effects processor and what comes out will be recognisable as a square wave. Not so with analogue tape -- the necessary frequencies are all there, but because of phase shifts in the electronic and magnetic components of the system, their time relationship is disturbed, which is why the waveform looks very different to the original. The simulations might be significantly closer if we were able to emulate this smearing before delaying the signals, as well as introducing more randomisation into the modulation.
During the '60s and '70s, most artificial reverb used on recordings was generated using a reverb plate, sometimes called (inaccurately) an echo plate.The reverb plate is an ingeniously simple device, but it takes a lot of tweaking at the design stage to get it sounding right. Plates work by suspending a thin sheet of metal under tension within a rigid frame via springs or clamps attached to the corners. A transducer similar to the voice-coil of a cone loudspeaker is used to inject audio energy into the plate and two or more contact mics fixed to the surface of the plate then pick up the vibrations inside it and feed them to preamps connected to the console effect returns. By feeding the different contact mics to left and right channels, a pseudo stereo reverb output is created.
Because metal plates have a tendency to 'ring', getting the plate thickness, size, material and tension right is quite an art, and some pre- and/or post-reverb EQ is invariably needed to fine-tune the sound. Furthermore, because the plate is very sensitive to external sounds and vibrations, it has to be mounted in a soundproof box, ideally on shockmounts.
Unlike digital reverbs, which have innumerable adjustable parameters, the plate reverb relies purely on EQ for tonality and physical damping for decay-time control (usually via a motorised felt pad). Pre-delay was often added by using an open-reel tape machine on the input, and replaying the input signal via the replay head to exploit the time gap between the record and replay head. You just had to hope the tape reel didn't run out during a mix...
Because a typical plate may only be between one and two square meters, and because sound travels much faster in metal than in air, the reflection density within the plate builds up very quickly following an impulse. The sound from the input transducer spreads rapidly across the surface of the plate in all directions until it encounters the edges of the plate, whereupon it is reflected and re-reflected back into the plate. To get the most random reflection build-up, it's best to have the transducer mounted a little way off-centre, and to get a wide stereo image, the two pickups are sited at slightly different distances from the plate edge, as shown in Figure 3 (right).
The characteristic plate sound is bright and extremely dense, with little or no impression of individual early reflections. The reverb builds very quickly and decays smoothly with a maximum undamped decay time of several seconds. Digital reverbs can provide a reasonable emulation of the coloration and envelope of a plate reverb unit, but only the more processor-intensive models produce the speed of reflection density build-up required to be truly convincing. Plate reverb was very popular for vocal and drum treatments, and though digital reverb has largely replaced it, many purists still prefer the 'real' plate sound for certain applications.
Those who couldn't afford plate reverbs used spring reverb -- a system still used today in guitar combos. The physical principles are similar to those of the plate reverb, except that sound is injected into one end of a loosely coiled metal spring rather than a metal plate, usually via a small magnetic transducer. A pickup transducer at the other end picks up the sound as it reflects back and forth along the spring.
Springs invariably impart a metallic coloration to the sound and they also tend to have a cyclic characteristic as percussive sounds cause vibrations to bounce back and forth along the spring in a fairly regular manner. Excessive input levels cause the springs to 'twang', so some systems incorporated an input limiter. To help even out the coloration and cyclic modulation, it's possible to use two or more springs operating side by side, each with slightly different mechanical characteristics and serviced by individual transducers. Using two similar sets of springs to treat left and right channels produces a convincing stereo effect due to the non-correlated nature of the two spring outputs.
Like plate reverbs, their main advantage is that they don't produce the gritty 'shattering' early-reflection effects of digital reverberators. Figure 4 shows a stereo spring reverb.
Tape Loop Echo
Before digital electronics and charge-coupled delay lines (analogue echo), delay effects were invariably created using tape-loop echo units such as the Echoplex, Roland's famous RE201 Space Echo, or, at a somewhat lower cost, the Watkins Copicat. They all worked on the same principle -- a loop of tape passes around a series of heads starting with an erase head, followed by a record head fed from the signal to be treated. Playback heads are positioned after the record head to provide the echoes, and some of the delayed signal is fed back into the record circuitry to create decaying echoes. Delay time is varied by switching heads or varying the tape speed, and models with multiple heads usually have switching systems for setting up different delay patterns (see Figure 5).
Basic digital delays only approximate tape-loop devices, even if multiple delay taps can be set to different delay times. There are various reasons why tape-based systems sound so distinctive; one of the main ones is the restricted frequency response of a loop of tape that's been dragged over a set of tape heads thousands of times. The tube circuitry of the original models also had a limited bandwidth and introduced a significant amount of harmonic distortion. This, combined with tape's tendency to saturate meant that when feedback was used, successive echoes became less bright and more distorted, creating a sense of the sound receding into the distance. On top of that, there was instability in the tape path caused by worn rubber pinch rollers that translated into low-level pitch modulation. As with distortion, this type of modulation becomes cumulative when feedback is used. Another feature that most of us would rather forget was tape noise, but there's no denying that a good tape-echo unit has a much more 'organic' sound than even the best digital emulations.
In conclusion, it is odd that vintage equipment is now valued because of its sonic 'imperfections', yet none of these were designed in deliberately. Modelling is still in its infancy, however, and I don't expect it to be long before vintage effects can be replicated much more accurately. Perhaps then we will be able to move on to inventing the future rather than striving to recreate the past.
Tips & Tricks
Technique : Effects / Processing
Part 2: Gates are far more than just problem solvers for reducing spill and noise. They can be used to add punch to drum sounds, put rhythmic interest into sustained parts or even as mixing automation, as Paul White explains. Additional material by Mike Senior. This is the last article in a two-part series. Read Part 1.
Last month I explained that there are good reasons why fully featured gates have more sockets and knobs than you'd expect — though simple audio I/O and a threshold control might be enough to deal with the simplest of gating tasks, there are many situations where such facilities would prove inadequate. Now I'm going to consider how best to apply such features to your studio tasks, and how the host of gating controls can transform a useful studio problem-solver into a versatile and creative mixing tool.
To Gate Or Not To Gate?
As a starting point, you have to decide exactly when in the recording process to gate your audio signals. A gate which is badly set up can completely ruin a signal, so if at all possible it's best to gate when mixing rather than when recording. If you must gate when recording, then double-check that the settings you have chosen at least don't cause any wanted audio to be muted in the part you're working on.
When mixing a multitrack recording, it is common practice to employ several gates, even if individual tracks don't seem too noisy on their own. This is because noise is cumulative, with every playback track of a multitrack recording contributing to the general level of noise arriving at the mix buss, so tracks are always best muted when not in use. If you are working with a digital system you can edit out any regions containing only noise, or you can simply mute tracks whenever they're unused with mixer automation, if you have it. However, the fact that gates can be set up to perform this function automatically often means that gating proves a more elegant solution.
If you are using any equalisation or compression, you'll normally want to use the gate first in the signal chain. The reason for this is that successful gating usually requires adjustment of the gate's response to the exact levels and timbres of the wanted and unwanted parts of your audio signal. Tweaking a pre-gate equaliser could easily mean that you have to also re-tweak your gate's threshold and filtering controls. Compressing a signal before gating it can make reliable triggering even more difficult to achieve — the compressor will make the unwanted signal a 'moving target' by modulating its level.
Generally it gives you more flexibility at mixdown if you record sounds without any effects such as delay and reverb, as the levels of such effects often need to be assessed in the context of the complete track. However, if you have recorded any sound with such effects, you should be careful when gating that track, so as not to shorten or modify the decay characteristics unnaturally. Unless, that is, you want that effect...
Traditional gating is all about the reduction of unwanted noise or spill, and the first thing to tackle with such problems is getting the gate to trigger exactly when you want it to. The first thing to reach for will be the Threshold control, which you should, in general, set as low as possible while avoiding false triggering. It can help here if you use fast times for the gating envelope's attack and release, as these will allow you to see exactly how the gate is triggering — though many gates also have useful LEDs which indicate the current action of the gate.
There are, though, occasions where increasing the gate threshold beyond the optimal noise-removal level can be desirable. With a fast attack time, a higher threshold causes the gate to open abruptly only when the signal has already reached a high level, adding a useful degree of punch to soggy kick drum sounds. Do make sure none of the quieter hits are being missed, though.
You will often be able to get satisfactory triggering with no more work than this, particularly if the unwanted signal is noise or mains hum. However, if background noise is particularly obtrusive and localised in the frequency range, then you'll probably need to dial in some side-chain filtering to get the gate triggering cleanly. For example, the fundamental frequency of mains hum and its first couple of harmonics could be removed by filtering the side-chain below about 200Hz, whereas many electronic buzzes can be dealt with by removing a little top end.
Side-chain filtering will almost certainly be necessary when dealing with spill from other instruments, particularly when gating a multitrack recording of a drummer (see my article on drum mixing in SOS February 2001 for more on this). However, it's worth bearing in mind that excessive high-frequency filtering might have a fairly noticeable effect on how quickly the gate responds to attack transients, even when dealing with basses and kick drums.
If triggering is still proving a problem, even when you've experimented with the gating threshold and the side-chain filters, then it could be worth bringing hysteresis and hold-time controls to bear on the problem as well, as they can often help the gate respond more reliably. However, if nothing can get your gate triggering absolutely how you want then you should try at least to make sure that no parts of the wanted sound are lost. It's always better to accept some noise or spill, rather than taking the risk of losing part of your recording.
If you're using only a simple gate, some false triggering is often unavoidable, and gating out spill on individual drum tracks can be particularly demanding, even when you have relatively sophisticated gates at your disposal. Usually this won't cause you any serious problems, as any spill that occasionally gets past the gate will normally be obscured by other instruments or ambience. However, in some cases the sound of spill being gated on and off can be more distracting than more continuous background noise. In cases like these, try setting the gate's range control to simply drop the noise by a few decibels rather than muting it altogether. Also, remember that adding effects such as reverb and delay at mixdown will often help disguise any small gating irregularities.
Tweaking The Envelope
Once you're sure that the gate is triggering reliably at the right time, then you'll need to consider how the gating sounds. The first control you're likely to have to reach for now is the release-time control. This will normally need to be adjusted so that the natural decay of the sound being gated is disturbed as little as possible. If set too short, the end of the sound will be unnaturally truncated, whereas if the release time is too long, you'll hear noise or spill dying away after the wanted sound has finished. The hold-time control can sometimes be tweaked to help solve any particularly difficult problems.
For normal applications, attack times should be as fast as possible, particularly for percussive sounds. The only thing to bear in mind is that extremely fast gate settings can cause clicks when the gate opens — while this can be a boon for drums, it can be problematic elsewhere. On the other hand, you might find that your gate doesn't seem able to react fast enough to your drum sounds, in which case you may have to reduce the gating range to allow it to open more quickly.
Pushing The Envelope
Normally the gate is a problem solver, removing elements of your audio tracks which you'd prefer not to hear. However, gates have many uses which are far more creative than this. Threshold and time controls in particular can reshape sounds in interesting ways. For example, fast-attack sounds can be made to sound almost 'bowed' if their gating envelope is given a long attack time. Drum envelopes can be tweaked for more or less attack, and you can make them seem almost synthetic if you gate with a very short release time. Strummed acoustics, rhythm guitars and rhythmic bass parts can also often be made more punchy by getting the gate to trigger on every strum or note (rolling off a lot of low end in the side-chain can help here) and then setting the range control for only a few decibels of gain reduction — any and all time controls available can be used to tweak the resulting envelope modulation.
The 'chattering' effect which can occur when signals linger around the gating threshold can also be used creatively as a new type distortion process. If you heavily compress your audio signal before it reaches the gate, and then set up the gate with its fastest time settings and with the minimum of hysteresis, you can often achieve quite consistent periods of extremely rapid chattering. The distortion that this causes can be extremely harsh, but can also be softened into something much more useable with the range or time controls. Bass sounds respond particularly well to this technique, because the waveform often moves slowly enough that the gate can actually modify the individual cycles of the waveform itself — great for adding a little edge to the sound.
Triggering From Your Sequencer
As nifty as the above processes can be, the real creative potential of gating becomes available when an external side-chain input (or 'key' input) is provided. This is because this input allows you to control the gating action of one signal from the level envelope of another, a technique which I introduced last month as a way to improve the rhythmic tightness of bass instruments and backing vocals.
However, a more up-to-date use for gates is for chopping up sections of audio in a rhythmic manner — an effect used by the Prodigy on some of their guitar parts. (Garbage even gate the entire track at several points on the opening track of their eponymous debut album!) This is a very simple effect to achieve: the signal being chopped is passed through the gate in the usual way but the gate is externally triggered from a rhythmic sound fed in via the key input. If a sustained sound (such as an organ patch) is used to trigger the gate, the note duration can be used to control the the duration of each segment of gated sound. However, if you are triggering with a drum sound, the gate's hold-time control can also be used. Some hardware and software gates can be controlled directly via MIDI, and these may be able to achieve this effect more easily — see the 'Creative Uses Of MIDI Gates' box.
But the usefulness of the above triggering example doesn't end here. If you lengthen the attack and release times and reduce the gating range, you can created a rhythmic tremolo, rather than hard gating. What's more, if you feed the same signal to a second gate, triggered with a delayed version of the first gate's key input signal, you can pan these two gates to opposite sides of the stereo image to implement auto-panning. (Note that this can also be achieved by using a gate and a ducker, with the advantage that you don't have to delay one of the trigger signals — see the 'Duckers' box.) Alternatively, you could feed the two gates to a couple of different effects processors to create interesting rhythmic modulation treatments. And there's no need to stop there. If you have enough gates and triggering sources, there need be no limit to the effects wierdness you can generate — check out my article on extreme effects in SOS November 2000 if you want to go further with this.
Mix & Match
While using sequenced MIDI sound sources to trigger gates can be fun, you don't have to use artificially generated trigger signals to get creative with gating. One of the joys of mixing in the analogue domain is that every track in a mix can be used as a trigger source for any number of gates operating on other tracks — a flexibility which is still surprisingly rare in digital systems!
One of the most famous (some might say infamous) effects that can be produced in this way is the '80s drum sound with gated reverb, as evidenced on the Phil Collins classic 'In The Air Tonight'. If an extremely ambient drum sound is gated with a high threshold, fast attack and release times, and a longer hold time, each drum hit becomes a concentrated burst of sound and can therefore seem more powerful. Figure 2 shows how to set this up. If you are wanting to create this effect with natural ambience, the effect would work best in a large, live room. A concrete stairwell will produce good results as long as the mics don't pick up the neighbours complaining! However, the gated reverb effect can also easily be achieved using an artificial reverb unit — simply trigger a gate on the reverb return with signals sent from the aux sends of individual drum channels. For more characteristic results, compress the reverb return.
Though the gated reverb effect on drums was done to death in the '80s, that doesn't stop you using it to thicken other sounds — both rock guitars and backing vocals can respond well to this. In fact, the lead vocal for David Bowie's 'Heroes' was apparently recorded with three different mics — one close, one a few feet away and the third at the other end of the room — the latter two of which were gated with different thresholds in order to introduce increasing amounts of room ambience as Bowie sang louder. And remember that delaying the gate's side-chain trigger signal by a beat or two can often provide extra rhythmic interest, when gating ambience signals.
Gating one signal from another can also help with balancing your music while mixing. For example, it can often really help vocal intelligibility in rhythmic music, even when the overall vocal level is quite low. If you gate the lead vocal, with the range control set close to minimum, and trigger the gating action from a drums submix, this will effectively mean that the vocal level rises momentarily with each loud drum beat, making it less likely that it will be masked out. It even helps to counteract the possibly detrimental effect on the vocal level of later mix compression. This trick can be applied in a host of other situations as well, wherever different sound sources are fighting for space in the mix.
Shut The Gate Behind You...
Gates are one of the most useful of the studio workhorses, particularly when recording live drums and rock bands. However, it is a shame to use them simply to reduce noise and spill when they are capable of so much more than this. Modifying the amplitude envelope of a sound subtly can ease the mixing process, while drastic measures can be taken for a number of special effects. And, once you start experimenting with external side-chain signals, the creative possibilities further multiply.