Parallel compression is a powerful mixing technique, but it’s often misunderstood. Read on to find out what it really does — and how it can help you make better mixes.
Hugh Robjohns
When talking with sound-engineering students of all ages and experience, I often find that one area where most struggle is compression. Perhaps that’s not surprising, since the concepts are fairly challenging on their own — before even contemplating the subtleties of different kinds of physical implementation (or modelling). Compressors are available in myriad different forms, using feed-forward or feedback control paths, with RMS- or peak-weighted side-chains, and even more variations of audio-attenuation device, such as opto attenuators, diode rings, vari-mu valves, solid-state VCAs… and the list goes on!
Let’s leave such complexities aside for the moment, though, because in this article I really want to try to clear up some common misunderstandings about the increasingly fashionable technique of parallel compression. Of course, when discussing an inherently complicated topic such as compression it’s vital that we establish a common framework of terminology, so that we all understand what we’re talking about — so that’s where I’ll begin.
The first thing to establish is what a compressor of any form actually does, and the answer is that it reduces the dynamic range of the input signal. Whether it’s configured to make the loud bits quieter, or the quiet bits louder, fundamentally it exists to reduce the overall dynamic range from something large and unmanageable to something smaller and more appropriate for the intended application.
The term ‘dynamic range’ can be interpreted in a variety of ways, too, but for the purposes of this article I’m using it to refer to the range of average signal levels between the quietest musical element and the loudest — the difference between the ppp and fff sections of a performance, if you like.
Downward Compression
The vast majority of compressors apply ‘downward compression’ which means, in essence, that loud stuff is made quieter. More specifically, signals below the threshold level are left alone, while those above are ‘squashed’ by an amount determined by the ratio setting.
The ‘ladder diagram’ (Figure 1) is a convenient way of describing the relationship between signal levels at the input of a compressor, and those at the output. The ladder at the top left shows the situation in bypass — the input signal appears at the output at exactly the same level, so all the ‘rungs’ are horizontal. The ladder to its right shows the effect of downward compression, where signal levels below the threshold remain unchanged, while those above are reduced in level proportionally, according to the selected ratio (in this case 2:1).

Figure 1: In these diagrams you can see the difference in effect on the dynamic-range reduction and output level of the various different basic types of compression discussed in this article. As you can see, ‘upward compression’ and ‘uplift compression’ do not achieve the same result, even if they do achieve a similar degree of dynamic-range reduction.
Figure 1: In these diagrams you can see the difference in effect on the dynamic-range reduction and output level of the various different basic types of compression discussed in this article. As you can see, ‘upward compression’ and ‘uplift compression’ do not achieve the same result, even if they do achieve a similar degree of dynamic-range reduction.
The more conventional way of illustrating compression is with a ‘transfer plot’, which is a graph with the input level on the horizontal axis and the output level on the vertical axis. The graph in Figure 2 was obtained by measuring the amplitude response of a ‘hard-knee’ compressor plug-in in a DAW, using an Audio Precision test system. The dotted straight red line at 45 degrees shows the response with the compressor bypassed — clearly illustrating that what goes in comes out, unchanged in level!
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