Paul White
Hello and welcome to this Sound On Sound podcast. I'm Paul White and with me is Hugh Robjohns.
Hugh Robjohns
Hello there.
PW
And we're going to talk about what we think are the four main studio effects that a newcomer needs to get their head around before they get into the deeper and more fancy toys. What would your main one be, Hugh?
HR
For me, I'd probably go with reverb.
PW
Yeah, me too and the reason for that is…
HR
I like reverb.
PW
Also yeah, music does sound nice with reverb.
HR
I'm a child of the 80s, the more reverb the better.
PW
When you record something in a small room, especially if you've put up your duvets and deadened all the unpleasant room acoustics, you end up with something that's very, very dry and that doesn't sound particularly nice on a record, especially with vocals. Excessive reverb can be equally sinful, but a little bit will actually give you a sense of being in a space. As you say, it just sounds musically nice. It glues the different notes together.
HR
Yes it does. It can also glue everything together into a complete stodge, as you say, if you overdo it. I mean really, there are two elements to reverb aren't there, there's the initial early reflections which really tell you the character of the room that this instrument or voice is in and then there's the reverb tail itself, which is what kind of glues everything together. So it's balancing those two which is a, you know, it's a skill that you can acquire. Presets have different balances of the two that you can pick quickly and see which one works best with your particular source in your particular mix. It's always a good idea never to judge reverb alone, you know, on a solo instrument, always judge it within the context of a full mix.
PW
That's true, because what sounds like a lot in the context of a solo instrument might seem to disappear in a mix.
HR
Yeah, it's possible.
PW
The other thing that's worth knowing is that the old fashioned plate reverb, or emulations of a plate reverb these days, are quite useful because unlike most algorithmic reverb switches, which as Hugh says, start off with early reflections, which build up into a reverb tail, the plate has a very rapid diffusion and so it doesn't actually suggest any particular kind of space which can be very, very useful.
HR
Yeah, absolutely. And it is a very popular choice to use on vocals.
PW
So in later years, we've had these things called reverbs that work with impulse responses, convolution. Do you want to explain a little bit about what they do?
HR
Convolution reverbs are based on real reverb that's actually measured and obtained from a real environment, a real room, real hall, real church, whatever and the way you obtain that information is typically, you could fire off something that was a click, a transient, a shotgun, a starter pistol, that kind of thing, something that gives you a burst of noise that then excites the reverb in the room and capture it that way. More commonly, it's actually done in a mathematical way by using a frequency chirp and you do it multiple times so that you can average out the noise and that generates again, you know, excites the room and generates the reverb tail. Once you've got that data, you can then put that into a convolution reverb processor and it uses that data to convolve with the audio that you're putting through it and effectively, in a very simplistic way, what you're doing is multiplying each sample of your recorded sound with a sample of the reverberation that came from a particular room and you end up with that room's reverberation on top of your original sound. I'm probably making it sound much more complicated than it is, but actually there's maths and it is complicated.
PW
It is complicated and in the early days it was very processor intensive until somebody found a few shortcuts, but now you can run several of them on even a modest computer without the computer getting upset about it.
HR
Yeah I mean the first one I came across was made by Sony and it was a, I think it was a 2U rack box and it got very hot and it was just a stereo reverb. But it sounded amazing because it sounds real in a way that algorithmic reverbs don't. But then again, there are algorithmic reverbs that sound unnatural in a very pleasing way. I mean Lexicons, there aren't very many rooms that sound like a Lexicon sounds, but it's a lovely sound to put on your recordings.
PW
Yeah, that's true of a lot of the early algorithmic reverbs. It was the imperfections that made them sound so musically interesting and there are lots of plug-ins now which emulate these old things.
HR
There are indeed, yeah. So I mean, it's horses for courses, there's, there's no right or wrong with any of this. If you want to use convolution reverbs with IRs obtained from particular spaces around the world, then go for it. It's fine. If you want to use simple algorithmic reverbs and it does what you need it to do, then that's fine too. There are no rules with any of this, it’s all about artistic intent and whether you like the way it comes out.
PW
Okay so let's talk about some of the key controls because some of these things have got millions of parameters, but the most important one is probably the reverb decay time, which is the time it takes to decay to the point where you can't really hear it and that's usually adjustable from less than a second up to maybe greater than 10 seconds with all the musically useful ones between probably one and three seconds, I'd say.
HR
Yep, I'd agree with that, yeah.
PW
Then you have a thing called pre-delay, which delays the onset of reverb after the dry sound, so that it creates that illusion that the sound has had to travel to some wall and bounce back again before the reverb starts, so it enhances a sense of space.
HR
Yeah and distance.
PW
The tonality of the reverb tile in the real world high frequencies tend to get absorbed more quickly than low frequencies so we have this thing called damping which allows that to happen in the algorithmic reverb as well.
HR
Yes and the same for low frequencies, not every reverb gives you that option but it can be there and again depending on the material of the room some rooms will, with stone for instance, it would tend to reflect a lot of high frequencies whereas if it was wooden, it would not tend to reflect a lot of high frequencies. So adjusting that damping not only has an effect on the impression of the size of the room, it's also the nature of what the room is built from.
PW
And of course if the room is really huge, like a cathedral, even though it's got hard walls, there's absorption by the air itself over those distances, so you still lose some top end.
HR
Yes.
PW
Of course, with plug-ins being so easily accessible within DAW software, it's very tempting to stick one on every channel that needs reverb, but you don't have to do this. If you go back to the way things were done back in the old days, you'd probably have a single reverb unit that was fed from sends on the individual tracks.
HR
It's this sort of send return idea rather than an insert idea. So what happens is, you use an Aux send from your track, you send it out to a bus, that bus then is fed to your reverb device and the output from the reverb device is brought back in in your mixer, whether it's a real mixer or a virtual one. So you've got two paths, you've got the direct path from the tracks through your mixer to the output and you've got this send path that goes out to the reverb and comes back in and then gets added in and turns up at the output. So that way you can adjust the send level into the reverb by adjusting the, the Aux Send controls and you've got the level coming back from the reverb, which you can adjust on the faders for the channel it's coming back in on.
PW
So the idea behind that is that you can use the same reverb on several different tracks using the Aux Send control to control how much reverb that particular track gets. So they're all sharing the same reverb.
HR
Exactly that and that's kind of, that resembles what happens in real life. If you take a, you know, a five piece string section and put them in a room, they're all contributing to the same reverberation in that room. So using one reverb for a lot of different sources is a way of replicating what happens in real life and it makes everybody sound like they're in the same space, but equally you could put different reverbs on different sources to give you some audio texture perhaps, or to be able to adjust the pre-delays differently on different instruments to make some sound like they're closer or further away. There are options. In the early days of DAWs they just didn't have the processing power to put a reverb on every single channel. These days you can and it's not really a problem. But it's worth remembering this old idea of using send returns, aux sends and bringing them back in on return, because it's a useful technique.
PW
And a way to get your head around this to start with, I think, is to set up one longish reverb on one send and a short one on another send and then you can blend the two for different tracks and get lots of different results out of just two different reverbs.
HR
Yeah, that works well. Yeah, I agree with that.
PW
When it comes to choosing the next effect in the list, that's quite a difficult one because we've still got EQ to look at, which we think is important, compression we know is important, but also delay, which is used to create echoes. Which would you pick out of those three Hugh?
HR
Personally the effect I use most often after reverb is probably EQ. But if I was doing live sound, it would probably be delay.
PW
Okay, so for the benefit of the viewers at home, EQ is a kind of really serious tone control.
HR
Yes.
PW
But it's not just bass and treble.
HR
EQ, equalisation. Equalisation is an old engineering term and it stems out of the use of telephone lines to pass audio between studios and transmitters and you would equalise the line to maximise the, or other to normalise the frequency response. But yes, it's, it's basically it's tone controls, but with more flexibility and variety than a standard hi-fi type tone control.
PW
So the different filters that a typical EQ might have would be low and high pass, low and high shelving and bell shaped cut and boost in the mid-range. Can you explain the difference between low cut and low shelving?
HR
I can try. A filter is, as you say, typically low cut, sometimes called high pass. They're the same things if you think about it, we're going to allow high frequencies through, we're going to get rid of low frequency. So we're cutting the lows and we're passing the highs. And the response of that filter typically is 6db per octave or 12dbs per octave or 18dbs per octave. Very occasionally you might come across 24. That tends to be more used on things like synthesisers than mixing desks. So a 6db per octave is relatively gentle and what it means is if you set the filter to, let's say, 100 hertz, an octave down from that, 50 hertz, the level will be attenuated by 6db. An octave down again, it'll be attenuated by 12db and so on and so on. And theoretically, the filter goes on forever. So very low frequencies are attenuated by a huge amount. So it's a way of getting rid of subsonic rubbish.
PW
Yes, quite useful on vocal tracks to get rid of breath noises that have made it past the pop screen.
HR
Yeah, or foot tapping on the floor that's coming up through the microphone stand, or air conditioning rumble or traffic rumble outside, things like that. It's just a way of getting rid of stuff that is not musical content and will actually get in the way of your mix.
PW
And likewise, you could use a high cut to get rid of something like hiss and noise on an electric guitar track because the electric guitar doesn't have a hugely wide frequency response. So what exists in that top one or two octaves is probably unwanted.
HR
Yeah, absolutely. The idea of using low cut filters and high cut filters or high pass and low pass filters is called bracketing and what you're doing, it's a little bit like you're pulling in the low frequency limit and you're pulling in the high frequency limit to just contain the musical stuff that you want and excluding everything else. And the more you do that, this bracketing idea, often the easier it is to mix because you're getting rid of information that is no longer relevant to your mix. Listen to it in isolation, you might think, you know, if it's a guitar or a vocal, you might think actually it sounds a bit thin, I've gone a bit too far. But in the context of a mix you'll often find it helps because you're concentrating the information that you need and getting rid of surplus stuff that is not helpful.
PW
Yeah, certainly depends what else is in the mix, doesn't it? I mean, if you've got a bass guitar and an acoustic guitar strumming chords, it's often very useful to take out the low end of the acoustic guitar to stop it conflicting with the bass. But if it's a solo acoustic guitar, you probably want more low end in it to make it sound more natural.
HR
Yeah, absolutely. Again there are no rules, there's no right and wrong EQ. It's making things sound right in the context of the mix.
PW
Okay, now the shelving filter. Again you have high and low ones, so how do they differ from your high and low cut filters?
HR
Well, the high and low cut filters just cut, they just get rid of stuff. Whereas the shelving filter, it can reduce levels or it can boost levels. It goes both ways and if you look at the control, you'll see it goes to the positive side or the negative side, so you can boost or you can attenuate and if you look at the frequency response, it looks a little bit like, you know, a flat line is the stuff that you're passing normally and then you'll see it ramp up and become flat again at a higher level, or ramp down and become flat again at a lower level. So you've effectively built this kind of shelf of boost or cut at a range of frequencies that might be the low end, or it could be the high end, depending which control you're using and the idea is that you're just adding some emphasis or you're reducing the emphasis of that particular sound. And typically again the slope, the rate it changes from not doing anything to full effect is about 6dbs per octave, so it's relatively gentle. It acts over quite a wide bandwidth because of that but it's just useful for giving a little bit of emphasis to some low frequencies or just tailoring them away a little bit more gently. It's more creative than the filter, which is more kind of corrective, if I can put it like that.
PW
Yeah, I mean, I suppose the key point is that it cuts or boosts whatever is above its cutoff frequency by the same amount. It doesn't kind of keep rising or sinking forever, as a standard filter does.
HR
That's right. Once it's got to its maximum level that's where it stays. So if you dial in 6dB of bass boost below 100Hz, then at 50Hz it's 6dB higher, at 40Hz it’a 6dB higher, at 20Hz it’s 6dB higher, they're all the same. So you've just lifted a chunk of your response and just brought it up or pushed it down a little bit.
PW
Yeah, it's quite a handy tool for adding a bit of air to vocals, isn't it, if you just lift up that very high end.
HR
Yep, using the high shelf, absolutely, yes. Yeah, it just compensates for that lack of HF you can get sometimes in a mix.
PW
Then we've got all this stuff in the middle, which people call parametric equalisers, because they've got three controls, haven't they? They've got cut or boost, they've got where is the centre frequency and they've got what's the width of the cut or boost, which is often called the Q.
HR
Or the bandwidth. Yeah. Not all of them have that. Some of them do, some of them don't and sometimes you might have just one mid-bell frequency, sometimes you might have two or three or even more and the idea of those is that you can tune them to a specific frequency region of your track. And again you can either boost or you can cut a particular frequency and you can control the spread of that boost or cut to determine what range of frequencies you affect.
PW
Yeah, you can make them very narrow. In fact, when I'm doing these podcasts I've got a sibilance problem, which occurs at about four kilohertz, and I can notch that out very effectively with a very, very narrow filter.
HR
They work better generally if you use them as a cut, to attenuate problems that you don't like, rather than as a boost to try and boost things that you do like. Our ears are not very good at spotting notches in the frequency response, whereas they can always pick up a boost. So you get a more subtle, less obvious effect if you cut rather than boost in general.
PW
That's a very good point. So one of the rules would be if you do have to boost somewhere, try and keep the boost fairly wide, whereas, uh, if you're cutting, then you can afford to make the cuts quite narrow.
HR
Yeah and the other issue of course, particularly with digital equipment these days, is that if you start boosting frequencies, you're going to risk running out of headroom and you could start clipping your output bus. So if you can, try and think about getting rid of the things you don't like, rather than amplifying the things you do. It's a bit of a mindset you need to get into, but you'll find it works better and more reliably if you do it that way.
PW
Okay Hugh, that's fair enough. So now we're down to delay or compression. Who gets your vote?
HR
I'm gonna go with delay, because I tend to use that more often.
PW
Okay, so delay is basically echo, as we would call it and back in the old days it was created using tape loops. You'd record onto one part of the tape and then a little further down the line would be one or more replay heads and you've got the echoes coming back, like the old Watkins copycat in the UK and later on Roland Space Echoes. Now of course we've got delays as plug-ins, which can emulate all the alt types, or they can be completely clean and very importantly, they can now be synced to the tempo of the song and that's something that was a real problem with a manually adjustable tape echo.
HR
Yes, it was. I remember the days of trying to adjust the speed on Roland Space Echos to try and get the slapbacks to work in the context of the music. It's a lot easier these days with tap tempo and that kind of thing.
PW
Yeah, now it comes down to where you would use echo. I mean, quite often it's used on guitar solos to some extent, to give you that big stadium sound. But you might be surprised to find it's also used quite a lot on vocals on contemporary pop records. So where you think you're just hearing reverb, what you're actually hearing is probably less reverb, although still there and some fairly subtle delays added in.
HR
Yeah, it is used a lot. I mean, I'm a keyboard player, I use echo all the time on keyboards for the same reason as you do on guitars. It gives you that sort of big stadium kind of sound. It makes the instrument sound a lot bigger and beefier. But as you say, you can also use it on vocals and it works very well because you don't tend to get the polyfilla effect that reverb has of clogging everything up. Echoes tend to be more distinct and it gives you a sense of space and it can emphasise the rhythm of the music without getting in the way of things.
PW
That's perfectly true. In fact, we first probably came aware of echo a long time ago with things like the early Shadows records and American bands like The Ventures and the surf bands who would use echo in quite a distinctive way.
HR
Yeah, absolutely.
PW
Yeah and after that, there was no going back, we just had to keep it. Okay, so that brings us on to compression which is a very, very useful process and at the same time, often very misunderstood.
HR
And misused, I think.
PW
And abused, yeah. So before I throw in my tips, I'm going to ask Hugh to explain what it is and why we need it.
HR
Compression is essentially a means of restricting the dynamic range of a single track or a complete mix or any combination of the two. The idea is that you're going to make the quietest things sound a little bit louder so they don't disappear and you're going to control the loudest things so they're not quite so loud so that they don't overload the equipment in whatever way. So it has that technical aspect in that you're trying to get quiet signals above the noise and keep them audible, you're trying to stop loud signals from overloading things. But you're also increasing the density of the sound, so the mix sounds richer and more interesting and fuller and…
PW
More even.
HR
More even, indeed, yeah. The extreme of compression is limiting where you don't really affect anything below the threshold but anything that goes above the threshold is pulled back very aggressively. But at the other extreme, you can have really, really low ratio compression, which squashes everything to a very small amount, but across a very wide area, so that tends to be much more subtle.
PW
I think ratio is quite often misunderstood again, so let's talk a little bit more about that. Essentially, ratio has it so that if you set a ratio of, for example, 5 to 1, that means that for every 5dBs that your signal exceeds the threshold that you've set, you only get a 1dB increase in signal level.
HR
Yeah, that's right. You set a threshold which is user adjustable and then anything that goes above the threshold gets squashed and the amount it gets squashed is set by the ratio. So a 2 to 1 ratio means you've got to go 2dBs above the threshold and the output will only go 1 dB above. If you go 10 dB above the threshold, the output will go 5dB above, because it's a 2 to 1 ratio.
PW
And if you set a huge ratio, you know, 20 to 1 or something like that, then the signal barely lifts at all above the threshold and that's when you get very close to being limiting.
HR
Yes, that's right. I mean, anything above 10 to 1 is effectively limiting. True limiters might be 50 to 1 or more, but you know, anything above 10 to 1 is pretty heavy and you can hear it working, you know, it goes from not doing anything at all to really clamping down on the sound very aggressively and you can hear that and you may like that effect, or you might not, depending on what you're trying to do with it.
PW
If you want something a bit more transparent, there is a thing called a soft knee compressor, which again can be a little bit hard to get your head around because it doesn't actually have a defined threshold point, does it?
HR
No, that's right. It moves gracefully into the ratio, so the ratio builds as you approach it, it's kind of soft focus effect, almost as you approach the threshold it starts to introduce the compression at a low ratio and as you progress beyond the threshold, that ratio gets steeper up to the level you've set, so that it becomes more assertive. So it gives you a more gentle transition from doing nothing to doing something. DBX were the classic people that introduced the soft knee compressor, but a lot of other compressors do it now and it's a very common effect.
PW
And I think one of the tips that we would probably both endorse is that if you go to DAW software, you find the compressor, you find there are lots of presets in there and the temptation is you find a preset that says rock vocal, so you stick it on your vocal and think the job's done. But of course the job isn't done because that compressor plug-in has got no idea how loud you've actually recorded your track. You might be one of these sensible people who records peak levels at about -15 or thereabouts. Or you might be one of these brave souls who has everything going to within a dB of maximum and so this poor old compressor doesn't know what it's doing until you go in there and adjust the threshold control while watching the gain reduction meter to get the thing starting to work.
HR
Yeah, absolutely. Presets can be useful places to start from. With EQ presets, obviously you can plug them in and often they work reasonably well from the off, but compressors and other forms of dynamic control, as you say, are very dependent on the threshold and the threshold is very dependent on what you've actually recorded. So the first thing you have to do is go in and adjust the threshold to get it to work the way you want it to work and as you say, the gain reduction meter is the key to knowing what you're doing there. The gain reduction meter is showing you how much it's reducing the gain, in other words, how much it's turning down the loud things and you may want that just to be one or two dB. 4dB is probably as much as you usually want to go to, or if you're trying to get a, you know, a powerful effect on a kick drum or something, you might want to go higher than that, you know, 10dB of gain reduction or more. That meter is the key to knowing what your compressor is doing and whether you've adjusted it into a sensible part of its range.
PW
Other than threshold, the other key controls that you will find are ratio, which we've already talked about and attack and release times, which I suppose really should be called sort of response times in a way, because when you think of attack, you tend to think of creating an attack, whereas with a compressor, you're kind of often removing the attack of a sound.
HR
Yeah, that's right. The attack control determines how quickly the compressor reacts when something exceeds the threshold. I think in one of your books you described it as how drunk the operator is operating the fader, because if he's had a good skinfull of beer, he's going to be a bit lethargic in pulling the fader down when he notices it's got a bit loud.
PW
So a long attack time.
HR
So a long attack time is like that. It takes a while to react, which means that it's not controlling the dynamics very well, it's actually letting quite a lot of those initial transients through, which may be the effect you're after. Alternatively, if you set it to a really fast attack, you know, it's a guy who's been on the speed pills and he's really jumpy and as soon as he hears anything that's even slightly close to being over the top, he's pulling the fader down and really snatching in those transients. And if it's too fast, that can actually create distortion. I recorded a solo flute and I was trying to control its dynamics and I couldn't figure out where the distortion was coming from and then I realised I’d inadvertently set the attack control to a stupidly fast time and it was actually creating distortion because it is true transient distortion I was doing.
PW
If its reaction time is faster than the cycle time of one waveform of the audio it can distort that waveform.
HR
Exactly that, yeah.
PW
And it's going to be more of a problem with lower frequencies, because you've got longer wavelengths and therefore you need a longer attack time to allow those to get through safely.
HR
Exactly that. So again, there's no rules on this, there's no fixed, you know, optimum attack time or fixed attack time. You need to vary it with the material that you're trying to compress and it'll be different for different kinds of instruments and different kinds of mixes.
PW
Yeah, it is a good idea to explore some of the presets if you're not too experienced in this world of compression, because you can see what kind of attack and release times they've set for different types of application.
HR
Yeah, good idea. Yeah and then of course the opposite of the attack time is the release time. So you've got a loud sound come along, it's exceeded the threshold, the attack has happened and it's, the gain reduction has been applied, it's pulled the level down. What happens when the signal falls back below the threshold. Well, when it falls back below the threshold, because we've got gain reduction in place, it's going to end up quieter than it really should be. So at that point, the side chain processing in your compressor says, oh look, it's gone below the threshold, I need to get rid of this gain reduction and the recovery or the release time is how quickly it dumps that gain reduction and goes back to normal. So again, you can have it really fast and a very fast release or recovery time tends to make things sound louder, it tends to make it sound like they're louder than they would otherwise be. If you have a very slow release time, it's very gentle, you probably wouldn't notice it, it’s like somebody gradually creeping the fader back up in a kind of auto gain control kind of situation, it’s very transparent. The downside with a slow release is that if you have a sudden brief transient comes along, it effectively punches a hole in the track because the gain reduction gets applied and it takes a long time for it to go away again, so you kind of get this hole punched in and people call that a kind of noise modulation effect or pumping, it’s sometimes known as.
PW
Or in dance music, they call it a beneficial effect.
HR
Exactly that, yeah. I mean I've always brought up this as a bad thing, but if you're doing dance music it is what they require apparently. So yeah, again, it's a different effect, you can use it in different ways but just be aware of those are the kind of extremes of what happens.
PW
Yes, the fast release times work particularly well on things like drums where you might have several beats coming in quick succession.
HR
Yeah, ideally you want it to have got rid of the gain reduction before the next beat comes along.
PW
Yeah.
HR
So you know, you need to adjust it to achieve that.
PW
So although it's a more advanced technique, parallel compression has become a thing. Would you like to explain a little bit about how that works before we start going into applications of it.
HR
Before I do that, one control we haven't mentioned so far is makeup gain and most compressors have makeup gain. Sometimes it's done automatically so you don't actually see the control, but it's still happening in there. Usually you get the control and the idea is that, as I've mentioned earlier, all compressors work by pulling the loud stuff down to make it quieter. But sometimes you don't want that, sometimes you want the loud stuff to stay as loud as it is, you just want the quieter stuff to be more audible, so you're looking at pulling up the low level stuff and the way you achieve that with a standard compressor is you pull the loud stuff down and then you use the makeup gain to turn everything back up. So in that way, you're restoring the peak level to where it was when it started, but in the process you're pulling up the quiet stuff underneath. So you've still reduced the overall dynamic range, but you've done it by pulling the quiet stuff up and leaving the loud stuff where it is. Except that, you haven't really left the loud stuff where it is because you've squashed that top end of the dynamic range and you can hear what which parts have been squashed and which parts haven't sometimes, so an alternative way of pulling up the low end is this idea of parallel compression and the way that works is you you do the send return idea that we had with the reverb earlier, you take the direct signal and you feed it through to the output, you take a feed of that signal through an aux send or something and you route it into the compressor and then you bring the compressor's output back in and mix it with the direct sound. So you've got a send return path, you've got two paths, direct and processed. And then what you do with the compressor is you set it up so that when the signal is loud, it's compressing really, really hard. I mean, really hard. It's putting in 20dB of gain reduction or more, typically. So what we now have is when the signal's quiet, the compressor's not doing very much because the signal is way below its threshold. So all the signal comes through the direct path and you've got the same signal coming through the compressor path. Those two add together and that's going to give you, because they're the same signal, it's going to double the level. So it gives you 6dB of boost for low level signals, so it's made quiet signals 6dB louder. When the signal gets really loud, you've got a really loud signal coming through the direct path, but the path that's going through the compressor is being really heavily squashed by the compressor, so it's not really adding very much at all, in comparison. So the loud stuff stays loud, the quiet stuff is made louder, but the peaks aren't being distorted by the compressor's action of pulling loud bits down, because the compressor's contribution is now so small in comparison, it's not really there.
PW
That's right. You're not reshaping the attacks at all. All you're doing is filling in the valleys, if you like.
HR
Exactly that. Rather than sawing off the top of the mountain peaks, you're actually filling in the valleys. So it's a different way of achieving the same thing and consequently, it has a different character of sound. It's more natural, particularly with transient rich sources like drums, it gives you a much more natural compression effect than straightforward compressors.
PW
But it is an advanced technique, it does take a little while to get used to it, which is why a lot of the early books on recording would say compressor is an insert effect, you wouldn't use it normally on a send.
HR
Yeah.
PW
Which is true, but this is one way of abusing the compressor, if you like, that's fallen into common practice and is now very, very popular.
HR
Well and you find that a lot of compressors these days have a mix control, a dry wet mix control, which is providing that parallel path inside the box. So, obviously, the dry is the straight through, the wet is the compressed path and you can adjust the balance between the two to generate the amount of parallel compression that you want.
PW
And the amount of compressor that you have to add in to fill in those values, if you like, is usually quite small. You do it by ear of course, but it'll be less than you think. It's a really effective way of keeping things sounding a lot more even than they did sound and it doesn't mess up the shape of the transients.
HR
Yeah, absolutely. I tend to use it most in mastering work, because I find you can get, you can bring up the quiet stuff and reduce the dynamic range in a way that is not at all obvious, it sounds very natural.
PW
So there you have it, that's our big four of effects, which is in no particular order, reverb, delay, compression and EQ. But of course there are lots of other effects that you can get into after that and I suppose the next one to think about is modulation.
HR
Yeah, there's lots of modulation effects, but that’s, you know, a huge range of different things. Phasers, flangers, time modulation effects.
PW
Yeah, most of them reflect what we're already used to finding in guitar pedals. As you say, phasers, flanges, chorus, tremolo, vibrato, all of those things and so they're worth exploring. They're all creative effects rather than curative effects, if you like. By all means, knock yourself out with the creative stuff, once you've got the big four under your belt, that's the main thing.
HR
Back in the day when I started, when you started doing audio, effects were relatively rare, weren't they? I mean, you would have, even a big studio would have one reverb, or maybe even not an electronic reverb, it would have a room that was a reverb room that you'd send sound down to on a loud speaker and pick it up with microphones and bring it back again and you would have maybe one or two compressors. You would have one or two EQs that you'd have to patch into channels. So we tended to use these things much more sparingly in days of old and actually, I think that was a very good way of learning how to use them effectively and what they were used for and where they worked and where they didn’t. These days, because it is literally a preset, you can just click on a button and drop it in. You can have EQ, you can have 10 EQs on a channel, you can have different compressors on channels, you can have reverb on every channel. It's very easy to get carried away with them and thinking that they're curing all of your ailments in your recording. Actually, it's much better to try and use them sparingly. Use them where you need to use them, not just because you can.
PW
Yes, it's always best to try and get the sound as close at source as to the way you want it to sound in the final mix. I mean, it's probably time to re-relate that story of the studio SOS we did, where the guy went off to make the cup of tea and said, can you have a look at my mix? First thing I did is I bypassed all the plug-ins to see what it actually got, you know, what's the raw audio actually like? And he comes running back in and says, it's better already, what have you done?
HR
Yeah, that's very true and that happens a lot, to be fair. I remember you once said on one of the talks we gave at a college that really there ought to be a slot on your computer keyboard that takes a credit card and you have to run the card through every time you want to put a new plug-in into the track just to make you think seriously about do I actually need this, is it going to help me or am I just going to make it louder, brighter, shinier.
PW
Yeah, especially if all that card money went to me, I'd go for that system.
HR
Yeah, it'd be good, wouldn't it?
PW
Yeah. See you next time.
HR
Thanks for listening.
Thank-you for listening and be sure to check out the show notes page for this episode, where you'll find further information along with web links and details of all the other episodes. And just before you go, let me point you to the soundonsound.com/podcast website page, where you can explore what's playing on our other channels.
