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Old 20-06-2003, 05:13 PM   #1
dr.Ru's Avatar
Join Date: Jan 2003
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Compression is a subject that has been covered on numerous occasions in past issues of Sound On Sound, but it is worth revisiting, both because of the importance of compression in contemporary music production and because many people are unsure as to the best way to use their compressors. However, in order to avoid retreading old ground, I'll only be giving a very quick overview of the principles of compression before moving on to some of the more advanced concepts ? if you'd like a more in-depth discussion of basic compression, then refer back to my article on compression in SOS April 1997.

In A Nutshell

Most of you probably know that a compressor is a device for automatically controlling the level of an audio signal. More specifically, a compressor 'turns down' the audio when the level exceeds a threshold set by the user. The amount by which the gain is turned down depends on the ratio of the compressor ? for example, if a ratio of 5:1 is set, an input signal exceeding the threshold by 5dB will be output with a level of only 1dB over the threshold. Once the signal falls back below the threshold level, the gain returns to normal. It's exactly the same as manually turning the level down with a fader whenever it gets too loud, but it's much faster to respond than any human and it's totally automatic.

To make the effect of compression smooth and natural-sounding, compressors often allow attack and release time parameters to be set by the user, but just occasionally these are fully automated. The attack time determines how long the compressor takes to reduce the gain once the input signal has passed the threshold, while the release time determines how long the gain takes to return to normal after the input signal has fallen back below the threshold. If the attack and release are too fast, rapid changes in gain cause an effect known as 'pumping'. All that pumping means is that the compressor action is clearly audible rather than subtle. Because compressors work by reducing level, most models have an output control called 'gain make up' or something similar. This control is simply used to restore the peak level of the compressed output signal to that of the uncompressed input signal. In effect, this means that compression makes low-level signals louder if the peak level is returned to its former value.

The last concept to explore before moving on to the more advanced stuff is that of the 'knee'. A basic compressor does nothing to the input signal until it reaches the threshold, then the full amount of gain reduction is applied as fast as the attack time will let it. This is good for assertive level control, but can be a little too obvious when a lot of compression is being applied to critical sounds within a mix ? or to complete mixes for that matter. A gentler-sounding compression can be achieved by using a so-called soft-knee compressor, where the compression ratio increases gradually as the signal approaches the threshold. Once the signal passes the threshold, the full ratio as set by the user is applied, but, because some compression is applied to signals approaching the threshold, the transition from no gain reduction to full gain reduction is far smoother. Figure 1 shows graphs of input level versus output level for both hard-knee and soft-knee compressors.

So, if soft-knee compressors are so smooth and cuddly, why don't we use them all the time? Firstly, it's sometimes nice to use compression as an effect, in which case a fairly hard compression tends to work best. A little deliberate gain pumping can give the impression of loudness and hard-knee compressors pump more readily than soft-knee types. The second reason is that, at higher ratio settings, the hard-knee compressor provides firmer gain control, so if a signal is varying in level to an excessive degree, a soft-knee compressor might not produce the required degree of levelling. The choice of which to use has to be made by ear, especially as every soft knee compressor behaves differently. Some have a relatively small knee, where the ratio increases over an input range of just a few dBs, whereas some start compressing at very low signal levels and then gradually increase the ratio over a range of 20 or 30dB. In fact some of these compressors are not so much soft knee as soft leg!

Bending The Law

As if hard-knee and soft-knee compressors didn't confuse the picture enough, there are other 'control law' effects to consider. In a theoretically perfect compressor, once gain reduction is applied (in other words, once the input is above the threshold), the response is reasonably linear, so no matter by how much the input exceeds the threshold, the output level increase will always be the fraction of that amount determined by the ratio control. Both hard-knee and soft-knee compressors settle down into this type of linear response above the threshold. However, there are some compressor types that don't exhibit a linear response above the threshold, and it's not uncommon for the amount of gain reduction actually to reduce at very high signal levels. In effect, this means that at very high signal levels the compression ratio tends to fall to a lower value, as shown in the graph of Figure 2.

Compressors that use lamps and photocells are notoriously non-linear but, rather than this being deemed a fault, it is acknowledged as one of the factors that gives them their distinctive sound. Compressors using valves within the gain-control circuitry may also be non-linear. It's not important to know a lot about the technicalities of such non-linearities ? just be aware that this factor contributes to audible differences between models of compressor that might otherwise appear to have the same broad technical specification. As is so often the case in audio, 'theoretically perfect' doesn't always equate to the most musical sound.

Inside Vintage Compressors
In the days before dedicated VCAs (Voltage Controlled Amplifiers), the most common gain elements used in compressors were valves, FETs (field effect transistors) and photo-resistive devices. While these gain-control elements are not nearly as accurate as VCAs, each gives a particular sonic character to the gain reduction which has been deemed musically useful for many types of music. Though some manufacturers, for example Aphex, now concentrate on the sonic purity of highly sophisticated VCA chips with ultra-low distortion and linear control responses, ultimately the choice of a compressor that flatters or one that controls dynamics as transparently as possible is an artistic one.
Valve gain elements tend to add a certain amount of distortion ? I find that this distortion sounds subjectively similar to that produced through soft limiting. Transients that are not caught by the compressor still tend to be softened by the non-linear characteristics of the valve circuitry, which is one of the reasons why valve designs are often felt to be more musical than their more accurate VCA counterparts. However, their control law is also somewhat non-linear, so some modern hybrid compressors use VCAs for the gain-control elements while using valves for the amplification stages. This can provide better control of the compression while emphasising the valves' soft-limiting sound, and such units can have a warm, musical sound when well-designed.

The Field Effect Transistor (FET) is often used as a gain-control element within more cost-effective solid-state designs. FETs have similar non-linear transfer functions to valves, and so tend to distort the signal in a similar way. LA Audio have made a number of compressors which are known for their vintage FET sound.

Photo-resistive gain-control devices are particularly interesting because they actually add very little direct distortion to the signal. Being purely resistive, they can be hooked into a circuit much like any other potentiometer. However, while they don't add distortion to the signal being processed, the non-linear control law of a combined photo-resistive device and light source gave them a unique sonic signature. The very first such compressors were designed before the invention of the LED and so used regular filament light bulbs. Compared to the dynamics of audio, light bulbs have a pretty slow turn-on/turn-off rate, so big attack overshoots were typical. Modern photo-electric compressors use LEDs along with compensation circuitry to speed up the gain change response, but there are still non-linearities in some designs that produce a musically interesting result. Modern opto-compressors are made by companies such as Joemeek and Focusrite (their Platinum range). You can approximate the sound of a vintage opto compressor using a regular VCA or FET compressor by setting a fairly long attack time (around 100mS) and a faster release than normal.

Digital compressors and plug-ins can be designed to emulate all types of hardware compressor, though how good they sound depends on the designer's understanding of the distortion and control law mechanisms of the original. Even today, experts argue over which aspects of a valve's electrical characteristics have the greatest influence on 'musicality', so do try out the real thing if you get the opportunity, rather than relying entirely on plug-ins just by default ? you might be surprised how different real analogue circuitry sounds!

Dynamic Damage

Basic compressors are little more than automated faders, but sometimes their action is at odds with the way audio behaves. It's well-known that you need a lot more energy to make a loud bass sound than a loud high-pitched sound, so it comes as no surprise in pop music to discover that most of the sound energy in a mix comes from the kick drum and the bass guitar or bass synth. When you compress a mix, it stands to reason that the compressor will respond mainly to the levels of these instruments so that, whenever a loud kick drum comes along, the level of the whole mix will be reduced for a few moments. Unless the amount of compression is quite modest, this can lead to an audible pumping of the high frequencies in a mix as they are reduced in level needlessly. Setting an attack time long enough to allow high-frequency transients to pass before gain reduction occurs can help in some cases, but this isn't always successful. Furthermore, there are occasions on which a fast attack time is necessary to achieve the right overall effect.

As we shall see later, the best solution to this problem is to use a multi-band compressor, but the designers of conventional compressors have also come up with some ingenious solutions to lessen this problem. For example, some designs use circuitry that allows a small amount of high-frequency signal to bypass the compression process so that, when a loud bass sound causes a drop in the overall level, the high end doesn't get killed. Once again, the technicalities aren't as important as the results, and what I'd like to get over to you is that, when trying out any compressor, you ought to listen to the way the high end changes when heavy compression is being triggered by low-frequency sounds. The variety is enormous ? some compressors sound quite dull and choked while others maintain the high end very effectively.

Peak Or RMS?

Going back to the 'compressor as a fader' analogy, the side-chain of the compressor is that part of the circuitry that listens to the incoming signal to see if it needs turning down or not. Most often, compressor side-chains are designed to respond pretty much like the human ear, which means that short duration sounds aren't perceived as being as loud as longer sounds of exactly the same level. This is called an RMS response (an abbreviation for 'Root Mean Square'), a mathematical means of determining average signal levels. The implications of using a compressor with an RMS control law are that the compression will sound natural, but short duration, high amplitude sounds may pass through at a higher level than you expect. One solution when feeding digital systems that can't tolerate overload is to use a fast acting peak limiter after the compressor.

Some compressors offer switchable RMS/Peak operation, and in Peak mode, the gain control responds more accurately to brief signal peaks than in the RMS 'averaging' mode. This ensures peaks are more accurately controlled, but at the same time introduces a greater risk that the broadband audio will be squashed unacceptably whenever a loud, short transient sound occurs. For this reason, it may be most effective to use Peak compression when treating individual drum and percussion sounds prior to mixing.

The Future Of Audio

Just like our hypothetical engineer controlling levels with a fader, a compressor can't take action until it 'hears' something that's too loud. To put it another way, a basic compressor's level corrections inevitably come slightly late. If a compressor is set to have a very fast attack time, the signal level can be brought under control before it overshoots, but even then the rise of an attacking sound will be distorted slightly by the compressor action ? though, fortunately, very short periods of distortion during transient sounds are not generally audible.

One way of getting around the 'too late' problem is to use a so-called 'look-ahead' compressor, where the side-chain is allowed to see the input signal a fraction of a second before it arrives at the gain-control stage. To do this in real time would require circuitry that could see into the future, so a more practical ploy is to delay the audio passing through the gain-control stage by just a few milliseconds while the audio feeding the side chain remains undelayed. In normal situations, a delay of three or four milliseconds is insignificant to a signal, but you should be aware that any hardware look-ahead compressor will introduce a tiny delay. Figure 3 shows the block diagram of a look-ahead compressor.

Software plug-ins used to process audio that's already been recorded fare somewhat better than hardware, because they can often get a chance to read an audio file slightly in advance of playback, therefore enabling them to work without introducing any delay. It's for this reason that look-ahead functionality is far more common in software compressors than in hardware. Many traditionalists don't like look-ahead compressors, because they don't give the same result as the analogue compressors they are derived from, but in situations where transients with extremely fast attack times are present, using a look-ahead compressor may be the only way of bringing peaks under control fast enough.

Within Limits

We are often told that a limiter is simply a compressor with an infinitely high ratio, so that once a signal reaches the threshold, it is prevented from exceeding it. This is pretty much true, but in this digital age where even very short periods of clipping may not be acceptable, a regular compressor is unlikely to be able to act fast enough to function as an effective limiter ? fast transients can pass through a system before your compressor is able to react, and this can result in clipping at your A-D conversion stage. In the days of analogue tape, this didn't matter so much, as short periods of analogue overload tended to be inaudible, but some digital systems can't cope with any clipping at all, however brief. In such situations, a dedicated, fast-acting limiter is the best bet.

In order to control signal peaks without affecting a sound's subjective level, some digital limiters may be programmed to allow a certain number of samples to clip before the level is reduced. In situations where the equipment next in line doesn't object to short periods of clipping, this can actually make the material seem much louder, though, as a rule of thumb, the period of clipping should be less than 1mS, which is equivalent to 44 consecutive samples at the sampling rate of CD-quality audio. However, if frequent clipping is expected, then the maximum length of clipped signal should be reduced to below 10 samples, as research indicates that repeated clipping within a short space of time is more audible than widely spaced instances of clipping. Some limiters emulate analogue soft clipping, where the top few dBs of any peaks are rounded off rather than clipped. Soft clipping can also help preserve the impression of loudness, though the effect can be audibly unpleasant if the signal is forced into limiting for more than very brief periods of time.

That's all for now, but in part two, next month, I'll be covering the many uses of the powerful dynamics-processing tool that is multi-band compression.

EQ Before Compression Or After?
Compressors are often used in conjunction with equalisers, especially in mastering applications. However, there's a big difference in the results achieved, depending on whether you put the EQ before or after the compressor, especially if the compressor is a full-band type. Let me give you an example: let's assume that a mix needs more low-end energy, so we add some bass boost at 80Hz. If we then feed the EQ'd signal through a compressor it will respond most to the loudest signal peaks, which in all probability will occur exactly where we applied the boost ? in other words the compressor will attempt to turn down the level of the sounds we've just tried to emphasise. Sometimes this will produce a musically useful effect, but where you want the EQ to be unaffected by the compressor, you're better patching the compressor first in line. If you have a hardware EQ and a compressor, or corresponding plug-ins, I'd recommend you try a few experiments to demonstrate just how great a difference can be made simply by moving the EQ before or after the compressor

In this second installment, I'll be covering compression in mastering and multi-band compression. But first I'd like to take a closer look at the main ways in which a full-band compressor can be used ? after all, if you are going to set the appropriate controls correctly, you have to know what you are trying to achieve. I like to simplify things by defining two main types of jobs you might want a compressor to do ? the effects of these two approaches can be seen on the waveforms in Figure 1 (right).

Double Vision

The first use of compression is for controlling signal peaks, so if you want to reduce peak levels without affecting the dynamic range of the rest of the recording, the usual approach is to set a threshold that's just above the average music level. This way only the peaks are subjected to gain reduction, and the more compression you wish to apply to those peaks, the higher the ratio you'll need to set. As a rule, ratios of between 2.5:1 and 8:1 are used for this kind of work.

It is sometimes easier to set up the threshold control using a high ratio along with fast attack and release settings, as the gain-reduction meters will kick in very obviously whenever a signal peak exceeds the threshold. Simply reduce the threshold until the gain-reduction meters start to show a significant amount of gain reduction between peaks, then bring it back up until only the peaks are affected. Once you've adjusted the threshold so that only peaks are being affected, you can return the attack and release settings to more suitable values and then work on the ratio control. A practical way to set the ratio control is to watch the gain-reduction meters as you vary the ratio and aim for a maximum gain reduction of between 8dB and 10dB. However, it's still vital that you listen carefully to the processed signal to see if it sounds the way you want it to ? meters can only tell you so much, and if the peaks start to sound squashed, you'll probably need to either reduce the ratio or increase the compressor attack time. As a rule, a hard-knee compressor will give the most positive results in situations where the signal peaks are in need of assertive control and, as explained last month, a compressor with a peak-sensing side-chain mode will track peaks more accurately.

Even though you are, in effect, compressing the signal peaks, it is important to keep in mind that, unless you are using a very fast compressor set to its fastest attack time, there may well be signal overshoots that the compressor can't catch. In a situation where overshoots can't be tolerated, it's safest to follow the compressor with a dedicated peak limiter. In a CD mastering situation, following compression with limiting is standard practice ? it's unreasonable to expect a compressor to prevent digital overloads on its own.

The second basic way in which you can use a compressor is for compressing the dynamic range of an entire signal, not just the peaks. In this case, it's usual to set a very low ratio of between 1.1:1 and 1.4:1 and to set the threshold at around 30dB below the peak level. Soft-knee compressors work well in this role and gentle overall compression is commonly used in mastering or for processing submixes. Conventional RMS, rather than peak, sensing would be the norm for this type of job, though don't let that put you off experimenting, as different makes of compressor can behave very differently.

Mastering The Art

One question I frequently hear is, 'Why should we need to compress at all during the mastering stage if individual tracks have already been compressed during recording and mixing?' The answer is that not all material will need compressing, but the application of a little overall compression can help the sounds within the mix to gel more effectively, even in cases where every track was compressed flat at the time of mixing. Just because individual tracks have been compressed doesn't mean the mix is always going to be at the same level throughout ? vocal lines will still have gaps between phrases, and instruments may come and go according to the arrangement of the song. The outcome is that the overall level of a typical pop mix still fluctuates according to what is and what is not playing at any given time.

Because the dynamic characteristics of a complex mix can vary considerably over the period of a track, a compressor which automatically sets suitable attack and release times is often easiest to use in this application. If your compressor doesn't have an auto mode, try an attack time of around 20mS and a release time of around 300mS, but experiment with these values, because every make of compressor responds differently. Use a low threshold in conjunction with a low ratio to trim a few dB off the original dynamic range and you should find that the impression of energy and mix integration increases. What's really happening is that the pauses between vocal and instrumental lines, as well as the gaps between drum beats, are compressed just a little less, which means that the level of the backing track is constantly adjusting itself to maintain a more even overall level. If this were overdone, there would be audible gain pumping, but kept down to two or three decibels, the subjective result can be very musical and can often help prominent parts, such as vocal lines, sit better within a mix.

Many Bands Make Light Work

When processing complex mixes using a conventional compressor, you can easily reach the point where gain pumping becomes audible, with high-energy, low-frequency sounds affecting the gain of the whole mix. Multi-band compressors were designed to avoid this problem, by treating different sections of the frequency spectrum independently. In such systems, the audio is split up into separate frequency bands, usually three, by means of a crossover circuit, each band then being treated with a separate compressor. At the output, the various bands are again combined to provide a full-range signal. Figure 2 shows a block diagram of such a setup.

The clear advantage of this system is that a loud, low-frequency sound will only instigate gain reduction in the low-frequency compression band, so any mid-range and high-frequency sounds occurring at the same time will be unaffected. This is in contrast to the conventional full band compressor where a loud kick drum will pull down the level of any simultaneous hi-hat and snare beats. Essentially, the ability to apply more compression without audible side effects is the main benefit of a multi-band compressor.

However, in a system where each band's compressor can be adjusted separately, there's a lot more you can do. For a start, the output gain of each compressor can be adjusted to alter the overall tonality of the mix. For example, if you feel the mix needs more mid-frequencies, you can simply turn up the output level of the mid-band compressor by a few decibels. You can also increase the perceived bass level by using more compression in the bass band than in the mid and high bands ? just set a higher ratio or a lower threshold. If you use the make-up gain control to compensate for this extra gain reduction, the average level of the bass will have been increased without increasing the peak levels, therefore making the mix sound more powerful at any given playback level. Similarly, if the top end needs a bit more sizzle or enhancement, it can be compressed a little harder too, in much the same way.

Most multi-band compressors also allow you to move the crossover points, and in most circumstances these need to be set in such a way as to separate the main bass and treble sounds from the mid-range. Kick drums and bass instruments need to be mainly in the low band while the mid band should be wide enough to accommodate the entire vocal range except for perhaps the highest harmonics and breath noises. This is important, as placing the crossover point of a multi-band compressor in the middle of the vocal range can compromise the vocal sound. In the top band, you should be aiming to capture cymbals, the bright edge of acoustic guitars and so on. For a pop mix, a low crossover point of 150 to 200Hz and a high crossover point of 5 to 8kHz would be typical.

When working with other material, listen to the mix and try to pick out the different ranges covered by the various instruments and sounds, then adjust the crossover points accordingly. Choosing the right compression settings when mastering takes a little experience, but the first step is always to identify the problem ? is it just a question of balance or is part of the mix more dynamic than it needs to be? Once you've pinpointed the problem, try to fix it using the least amount of processing.

EQ Before Compression Or After?
Compressors are often used in conjunction with equalisers, especially in mastering applications. However, there's a big difference in the results achieved, depending on whether you put the EQ before or after the compressor, especially if the compressor is a full-band type. Let me give you an example: let's assume that a mix needs more low-end energy, so we add some bass boost at 80Hz. If we then feed the EQ'd signal through a compressor it will respond most to the loudest signal peaks, which in all probability will occur exactly where we applied the boost ? in other words the compressor will attempt to turn down the level of the sounds we've just tried to emphasise. Sometimes this will produce a musically useful effect, but where you want the EQ to be unaffected by the compressor, you're better patching the compressor first in line. If you have a hardware EQ and a compressor, or corresponding plug-ins, I'd recommend you try a few experiments to demonstrate just how great a difference can be made simply by moving the EQ before or after the compressor.

In a mastering situation, having independent control over each band can really help to sort out a problem mix. One popular strategy is to use a higher ratio and higher threshold to sort out low-end peaks while using the gentler low-ratio and low-threshold approach in the other frequency bands. On the other hand, using more compression in the mid-band can often help lift the vocals out of a problem mix. Some multi-band compressors, such as TC Electronic's Triple*C, don't have independent control over the bands other than for level, but they do provide templates for different types of music where these more advanced settings are preset within the templates. In most cases, the appropriately named template will be the right one for the job in hand, but don't let that put you off trying different templates to the obvious ones just to see what happens.

However, even if your mastering compressor provides full and independent control over each frequency band, it's usually a good idea to use similar attack and release settings across the three bands unless you have a very clear reason for doing otherwise. Generally the attack will need to be as fast as possible without making the compression process sound too obvious, though in some situations, you may want to increase this a little to allow brief transients to stand out a little more. If the attack and release times are set too differently, then the attack of a transient sound may be disturbed, with some parts of the spectrum coming in before others or coming in with more initial intensity. In extreme cases, badly mismatched attack and release settings can even have a detrimental effect on the apparent timing of the music.

The Final Word

Compression is a much more subtle process than adding an effect such as delay or reverb, so you may have to play around more before you feel you have enough experience to get the results you want. Dynamic control is a key element in modern music production, whatever the style, so give yourself time to learn, and be aware that different types of compressor can produce very different subjective results. If you have a digital compressor with factory presets, look at the way the presets are set up and try to figure out why the designers chose those parameter values ? can you imagine what effects those settings will have on the type of signal they were designed for. You can also learn a lot from listening to commercial records to see how they were mixed and mastered. Most importantly though, learn restraint. Overprocessing is almost always more damaging to a piece of audio than underprocessing

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