THE BLACKOUT AUDIO GUIDE TO: Compression...

[MARKEG]

Someone posted a great guide to compression on this forum a while ago but I cant find it. In fact, there's been loads of posts on this subject. I think alot of our members need a concise guide on 1 thread!!!

Post all general links/advice on compression on this thread!

Also if you remember that anyone has posted something somewhere, please copy/paste/give the link on this thread.

[djTequila]

Here's something I found on IRC...

Compression Corner

Punch, apparent loudness, presence... just three of the many terms
used to describe the effects of compressing and limiting on an audio
signal.

The terms compression and limiting have been in the audio vocabulary for years,
yet there is some confusion over their definitions. The confusion arises from the
fact that both the compressor and the limiter are devices that restrict the dynamic
range of a signal, and the difference between them is one of degree, with the
limiter having the most effect. To simply define each:

Compressor: An amplifier, whose gain decreases as its input level is increased.

Limiter: A compressor, whose output level remains constant, regardless of its
input level.

Both definitions are valid only after the signal being processed reaches a certain
level. Therefore, one more definition needs to be thrown out for consideration:

Threshold: The level at or above which the compressor or limiter begins
functioning.

In a situation where input and output are idealized for a combination
compressor/limiter, as the input level increases from -10dB to 0dB, the output
level, likewise, increases from -10dB to 0dB. Here the device is functioning as a
simple unity gain amplifier, with no effect on the signal level.

Compression

Once the signal level exceeds the compression threshold of 0dB, the output level
will follow the compression curve, assuming a compression ratio of 2:1, as the
input increases 10dB; the output will yield only 5db more gain.

Limiting

In a limiting situation, with a limiting threshold of +20dB, once the input level
reaches +20dB, there is no further increase in output level. Hence, the device is
operating as a limiter,. In actual practice, compression ratios of greater than 10:1
are considered as limiting. Once the limiting threshold of +20dB has been
reached, the output level remains at +10dB, despite further increases in input
level. Therefore, it should be understood that the limiter threshold does not
necessarily indicate the maximum allowable output level of the device. Rather, it
indicates the input level at which the limiter begins working.

Variable Thresholds

It should be noted that the same compression ratio of 2:1 mentioned earlier will
have different effects on the overall dynamic range depending on the point at
which compression begins (threshold). Also, the positioning of the compression
threshold will influence the point at which limiting must begin, if a certain
maximum output level is not to be exceeded.

Variable Compression Ratios

Most state-of-the-art compressors offer audio engineers a variety of compression
ratios from which to choose. Assuming that an audio signal must be kept below
+10dB, using higher compression ratio settings will allow for a greater dynamics
range of the signal being processed.

Pumping and Breathing

It is relatively easy to determine the compression threshold and ratio needed to
prevent a wide dynamic range signal from exceeding a specified output level.
However, it should be realized that - especially at high ratios - the action of the
compressor might become audibly obtrusive. To understand why, remember that
the compressor is a variable gain device. The higher the compression ratio the
greater the change in gain. A constant high level signal, say +10dB will cause
more gain reduction. When the high level is removed, the amount of gain
reduction decreases as the compressor returns to unity gain. If the gain reduction
fluctuates rapidly, it may be quite audible as the background noise goes up and
down in time with the compressor action, i.e. attack /release times, causing a
breathing like sound. This can be used sometimes as an effect producing some
killer results.

Example: The use of extremely short attack times and longer release times may
create a backward-like sound, especially on percussive instruments. The fast
attack immediately drops the signal, and then as the signal naturally decays, the
release time setting brings up the gain, working against the normal decay. This
effect is particularly noticeable on a drum set, and particularly on cymbals.

Program Limiting

Often compression may be applied to the overall program rather than to an
individual instrument. Known as program limiting, this practice will prevent
cumulative levels of the various instruments from getting too high or falling too
low. This type of gain control must be approached with care, since the adverse
effects of compression are heard on the entire program.

Program limiting is often used to raise the apparent loudness of a record. Since
the ear averages the sound level over a period of time, a low level program with
occasional high level peaks will not seem as loud as an average level program
with no high level peaks. (Confused yet?). In the quest for louder sounding
recordings and broadcasts this type of loudness boosting is often overdone,
much to the detriment of the finished product. Meaning, what you hear coming
over the radio or television is often much different than the original producers,
engineers and mastering people had in mind.

Stereo Program Limiting

When a stereo broadcast program is limited, the gain regulating sections of the
left and right track compressors must be electronically interlocked, so that the
compression in one track causes an equal amount of compression in the other
track. This keeps the overall left-to-right stereo program in balance.

Consider a stereo program in which the right track occasionally needs some
compression. During compression, a center placed solo would apparently drift to
the left whenever the gain of the right track is affected by the compressor. To
prevent this center channel drift, the stereo interlock function reduces the gain in
both channels simultaneously whenever one exceeds the threshold of
compression. This keeps the center placed information from moving from side to
side with each action of one or the other compressor channel.

[djTequila]

Here's another one... A bit more 'hands on' but unfortunately some bits rely on diagrams which I simply don't have access to :(

The Compressor's Secrets

DAVID MELLOR discusses the less well known aspects of compression.

Every studio has one, every engineer uses one, and every popular music recording - almost - dating back to the
1950s and beyond has benefited from one. Of all the many and varied types of outboard in the processing and
effects racks, the compressor is surely the one that is most often used, and one that repays its cost of ownership
countless times over during its working life. So I don't need to tell you anything about compressors then? Maybe
not - if there does happen to be anything that you don't know already then you can easily find it in textbooks and
magazine articles that are often aimed more at the beginner than the seasoned pro. However, the compressor is a
many-faceted instrument, and there are a number of tips, tricks and techniques that are not commonly covered in
print. Are these the compressor's secrets, known to the few and hidden from the many? Like the Masked
Magician, I intend to reveal these secrets to the world.

Merciful Release

A long time ago, when I was a fresh-faced student of sound engineering, I went to a trade show (in the days
when you had to beg your way in, if you weren't in the business already) and alighted on the stand of a company
who had a new and wonderful compressor to show off. 'Listen to this,' said the silver-tongued salesman. I listened
as he demonstrated his amazing box. 'That's 30dB of compression. Does it sound compressed to you?' I looked at
the gain reduction meter, I listened, I looked at the gain reduction meter, I listened. Sure enough, the meter was
showing a full 30dB of gain reduction and the music I was listening to sounded as fresh as a live performance. I
knew something about compressors, and I knew that 30dB of gain reduction ought to be the sonic equivalent of
what an apple looks like after it has been through a cider press. It's a good thing I didn't have any money or I
might have bought it on the spot.

With the benefit of experience I know what happened. I am sure that it was a reasonably good compressor, but
not significantly better than any other. What the salesman had done was to turn the release control to maximum.
Release, as you know, is the time it takes for gain reduction to return to zero after the signal has passed below
the compression threshold. In this case, the signal never passed below the threshold long enough for the level to
begin to return to normal, to any significant extent. The result was indeed 30dB of gain reduction, but not 30dB of
compression. You don't need a compressor to get any amount of gain reduction - just lower the fader.
Compression implies a constantly changing amount of gain reduction, and the gain reduction meter must be visibly
dancing up and down. If it is not, you're wasting your time. How fast it dances up and down is up to you but, if
you want value-for-money compression, a short release time will give you a more audible compression effect
(Figure 1). A longer release will lessen the audibility of the compression, but you won't actually get as much real
compression.

Over Compression

No one reads the manual for a compressor, and if you did you wouldn't get any warning about the effects of over
compression. I don't mean this in the sense of too much compression, your ears will tell you that, but in the sense
of setting a lower threshold than you need to get the job done. This will always make the sound worse, with the
sole exception of percussive sounds where it might sometimes be a useful effect.

Let's assume a scenario where an instrument plays occasionally with silences in between. This is where over
compression is most likely to happen. When setting the threshold, many users have an idea of how much gain
reduction they want to hear and see on the meter. The amount of gain reduction is controlled by both the
threshold and ratio controls. Suppose these controls are set so that the desired amount of gain reduction, let's
say 12dB for example, is achieved. This should be fine shouldn't it? Look again at the gain reduction meter. While
the instrument is playing, does it ever go all the way down to zero? If it doesn't, if it only goes down to 3dB, then
you haven't applied 12dB of gain reduction, you only have 9dB of compressive gain reduction. The other 3dB could
have been achieved by simply lowering the fader. This, in itself, isn't necessarily a problem. The problem is that,
when the instrument starts to play, the compressor has to go all the way from zero gain reduction to the full
12dB. The necessity of covering that additional 3dB will audibly distort the initial transient. Try it, and you will
hear it for sure.

This leads to rule number one of gain reduction - at some point in the course of the track while the instrument is
playing, the gain reduction meter must indicate zero, otherwise the minimum reading obtained shows wasted gain
reduction and over compression, leading to the distortion of transients that follow silences (Figure 2).

Compression By Stealth

One of the best-known uses of compression is to increase the apparent loudness of a mix, or an individual voice or
instrument for that matter. Compression, as you know, works by reducing the high signal levels, bringing them
closer to the low-level passages, and then applying make-up gain. Thus the low-level signals are brought up and
the whole thing sounds louder. This is fine in theory, the trouble is that the effect of compressing the high-level
signals is very audible, necessitating great care in the set-up of the compressor and judicious compromise
between getting enough compression and not spoiling the overall sound. Ray Dolby told us this when, in the early
A-type noise reduction system, he left high levels completely alone and modified the gain only of signals below
-40dB. What we need is a compressor that only operates on low-level signals. Is there such a thing? Yes there is,
and it's in your rack already. You just have to use it in a different way. Since in this situation the object is to
bring up the lower levels of the track, what we need is a way of making the quiet sections louder without
affecting the loud sections.

The answer is to mix the uncompressed signal with a compressed version of the same (Figure 3). At levels below
the compressor's threshold the two signals will combine to produce a 6dB increase in level. Above the threshold
the compressed signal will be progressively reduced and add hardly any additional level to the mix. The result is a
form of compression where you can get more dynamic range reduction with fewer audible side-effects. I'm not
going so far as to say that it is always the best way, but it's certainly worth a try. Maybe some enterprising
company will bring out a gadget to do just this, in a convenient rack mounting package. By the way, if you try
this with a digital compressor you will get a lesson in the delay involved in digital processing. You will get comb
filtering and it will sound dreadful.

Compression Vs. Clipping

While I'm on the subject of increasing apparent loudness, I don't know whether it is as widely appreciated as it
should be that compression is only half the answer. Compression is a long-term type of gain reduction, working at
the very least over periods of tens of milliseconds. If you try to achieve very fast acting compression by using
very short attack and release times, you may well end up with distortion of low frequencies where the compressor
actually changes the shape of the waveform. There comes a point in maximizing apparent loudness where the
compressor has given all it has got to give. Clipping, on the other hand, works on a very short timescale.
Transistorized circuitry reacts within microseconds to any level that is too great for the power supply to cope
with and cuts it short, creating harsh harmonics, but at the same time extra loudness. The soft clipping of valve
and valve-emulating designs rounds rather than clips the peaks but, once again, operates on a short time scale.
The problem with soft clipping, if used alone, is that it only works on high-level signals. Clip-worthy peaks only
occur in quantity in high-level signals and low-level signals, although they may indeed have the occasional
clippable peak, are largely unaffected. The answer is to use a compressor and a soft clipper in series.

The compressor evens out the general level of the signal but, since it works over a comparatively long time scale,
the peaks are not clipped but simply brought to a more uniform level. The clipper then has more material to work
on. A useful alternative is to use a series-parallel configuration as shown in Figure 4. Here, the compressor
smooths out the levels, the valve emulation device soft clips the peaks, and the result of that whole process is
added to the uncompressed signal. The result is controllable enhancement over a wide range of levels. If you
want to go further then you might add an equalizer after the compressor so that you can choose the frequency
range that will be affected to add just the right hint of distortion without going over the top, particularly in the
mid-range.

MS Compression

Here's an interesting curiosity. As you know, when compressing a stereo signal, a two-channel compressor must
have its sidechains linked, otherwise heavy compression in one channel will cause an image shift in the stereo
sound stage. Both channels must, at all times, be compressed equally. Of course, this assumes that you are
handling stereo as left and right channels - let's call this LR stereo. Not as popular but certainly very useful is
mid-side or MS stereo, where the M channel is the mono sum of the whole sound stage and the S channel
represents the difference between left and right. MS is a useful microphone technique and is sometimes used at
other points in the signal chain for modifying the width of the stereo image. (It's a funny thing that proponents of
MS often forget that you can do that to LR stereo signals with the pan controls.) But what about compressing a
signal in MS format? Is it possible? Does it have anything new to offer?

Yes, it is possible to compress MS signals without converting them to LR. Just pass the M signal through one
channel of the compressor and the S signal through the other. Once again, you will need to link the sidechains or
funny things will happen, but it will all work perfectly. Some might say that it works better than compressing LR
stereo since, even when sidechains are linked, it is not guaranteed that analog compressors will handle both
channels absolutely equally and some image shift may persist. But, if you compress in MS domain then any
disparity between the channels will result not in an image shift, but in a variation in the width of the stereo image,
which is arguably less obtrusive. But why not take this a stage further and do something really wacky like
compressing the S signal only. What happens now? If you compress the S signal only, then anything panned
center is unaffected and compression only affects signals panned left or right, or signals that are out of phase.
Loud signals in these modes will cause a momentary reduction in level of the S channel resulting in a narrowing of
image width. I can't say that I recognize any useful function for this myself, but in the hands of more creative
people, who knows?

Serious Sidechain

Everyone knows how to direct a high-frequency boosted signal to the sidechain to perform a crude type of
de-essing - now superseded by more sophisticated stand-alone de-essers such as the Drawmer MX50. But what
about applying EQ to the sidechain in general, rather than this one specific application? If you have never done it,
do it now. Parallel a signal so that it enters the normal input of the compressor, and at the same time is
connected to the sidechain input via an equalizer. Now play some signals through this set-up. We all know that
different compressors have different sounds, but this little trick allows the compressor that's in your rack right
now to have an incredible range of sounds going far beyond the normal differences between models, when used in
the standard configuration. You will find that the compressor becomes another type of EQ but, instead of simply
cutting or boosting different frequencies, you allow different frequency bands to control the amount of
compression applied. When you are in search of that elusive 'phat' sound and simple EQ and compression are not
getting you there, EQ'ing the sidechain might just do it for you. In fact, I would go so far as to say that all serious
compressors should have sidechain EQ built in. Once you have really tried it you won't want to do without it.

The sidechain can do more. Everyone knows that different compressors sound different, and that soft-knee types
are more subtle than hard-knee, which go immediately from uncompressed to compressed at the exact threshold
level rather than the gentle blending of the soft-knee type. The precise knee curve of a compressor is an
important factor in its sound, but few compressors allow you to modify the knee curve in any way. So can it be
done? Well of course it can, otherwise I wouldn't have mentioned it. Here's the deal: set up a sidechain
configuration as above, but this time, instead of an equalizer, insert a distortion processor. A guitar effects unit
such as the SansAmp GT2 would be fine (Figure 5). Remember that you are not going to hear any signal coming
out of the sidechain, unless there is some internal crosstalk within the compressor, so the output signal isn't going
to be distorted. What the GT2, or similar device will do is apply soft or hard clipping, which will bend the shape of
the knee curve of your compressor. What effect this has depends on the compressor itself, on such factors as
whether peak or RMS detection is used for example. However, the result will be that you will feel as though you
have a totally different compressor in your rack. In fact, when different settings are used on the distortion box
you will feel as though you have installed a whole rack full of different compressors.

Another option for the sidechain is to insert an advanced version of the signal to control the level of the signal
itself. One of the enduring problems of compressors, and gates for that matter, is that they can only react to
whatever information they receive, they can never anticipate what is going to happen and prepare for it. Well
now they can. Using a digital tape or hard-disk multitrack it is commonly possible to delay individual tracks with
respect to the others. Even if it isn't possible to advance a single track, you can always delay the rest, and
perhaps make a delayed copy of the track you want to process. Armed with this you can connect the advanced
version of the track to the sidechain - just 50 to 100 milliseconds should do - and the delayed version to the
normal input. Now you will find that the compressor anticipates the amount of gain reduction required and
transients in particular are rendered very much more realistically than doing things the normal way. In fact, you
can do it the other way around too - delay the sidechain so that the compressor takes a moment to react. 'Why
would you want to do this?', you might ask. The answer is that percussive sounds often benefit from a relatively
slow attack, allowing the initial transient to come through unaltered before the 'body' of the sound is compressed.
This is just a different way to do it, but this time with a little more control.

Radical Ratios

When is a compression ratio not a ratio? I could give you a straight answer but instead I would like to ask another
question. Whoever said that it should be a ratio? Some scientist I don't doubt. Virtually every compressor on the
market offers logarithmic compression, such that, once the knee curve is passed then, for example, at a 2:1
compression ratio a 10dB increase in level at the input will result in a 5dB increase in level at the output.

This is all very tidy, but I wonder whether this is always going to be the right approach? How about a compressor
where, once the signal exceeds the threshold, it is subjected to a knee curve leading to logarithmic compression,
as tradition dictates, but beyond that the compression is lessened and the curve reverts to a straight line,
meaning no compression (Figure 6). Here, signals of a certain level are compressed, but louder transients are
substantially unaffected.

With traditional compression, it is usually the transients that cause the problems, so once you have got the
general run of signal sounding pleasant, along comes a transient and the whole thing goes crazy for a second.
Why not just let the transient through so it can be on its way, and concentrate on the parts of the signal that
will really make a difference. You can always limit the transient later if you need to. There is actually a range of
compressors that do depart from the traditional logarithmic curve. I'll give you a clue - they are all bright green in
color.

But there's a whole world of options waiting to be explored, by users and by designers. Compression is boring in
comparison with what it could be. Why not have a bit of fun and experiment? Most of the ideas I've outlined here
won't cost you a penny, and you may never have to buy another compressor again because you'll be getting all
the fun you need from the compressors you already own!

[Ritzi Lee]

wicked!!

[dj vorny]

nice one :D

[tioneb]

it doesnt affect anything in the way to make a track sound properly but ther is something wrong in the first quote.

the dB scale is not linear, but logarithmic. If you consider the power of signal in W (PW), then you get the power in dB (PdB) like this :

PdB = 10*log(PW) , where log means decimal logarithm.

To be more simple, if you multiply a power by 2, the gain is 3dB. So if we consider the 2:1 ratio compressor, if the input is over the threshold and increases by 10 dB, then output will increase by 7 dB (and not 5dB !!!). Thats because the dB scale isnt linear that if you look at your comp's curve it doesnt look like a straight line.

That is also useful to consider this when going in parties, just thnik that if the sound level is 6dB above what youre used to hear, its 4 times louder ... Quite painful for the ears imo...

[BombJack]

Cheers Guys - very useful material indeed...

BombJack

[MangaFish]

is there a chance this thread could be made sticky so newbe's can find it easy?

[DJZeMigL]

will do..

Z

[DJZeMigL]

sorry don't have the tools 2 make it a sticky... Mark help please!


Z

[messyfuture]

bump**

its a good thread

[NooNoo]

Very helpfull, especially the 1st one. Nice to see things explained in laymans terms for a change.

[Mindful]

*BUMP*

[rounser]

Here's a trick which I learnt recently that seems to be giving me a road map when using a compressor:

To start, set ratio to max (inf:1), threshold to max (0 db), attack to zero, gain to 0 db, release to something you'll hear, but not too long...

If you now adjust threshold downwards, you can hear the point it just begins to bite, and you can also hear what your release setting is doing because of the extreme pumping...when happy with threshold, adjust release until the length of the release is musical for your track*.

When happy with the threshold and release, lower the ratio to something more sensible and increase the attack to something that flatters the kicks, perhaps.

I'd welcome input from more experienced folks as to whether this approach is any good...or whether it's obvious and what I should have been doing all along.

*: If you're having trouble hearing what's musical, I found what could be a bit of a gem of advice, potentially useful for delay and reverb settings too:

converting bpm to milliseconds per beat is just maths...
specifically, unit conversion.
(anyone here who remembers elementary physics
will begrudgingly admit to knowing what i mean.)

given that you have:
60 seconds per minute
a tempo of 180 (adjust accordingly with each tune)
tempo is in beats per minute
1000 milliseconds per second

we want:
milliseconds per beat

60 (seconds/minute) / 180 (beats/minute)
= 1/3 (seconds/beats)

1/3 (seconds/beats) * 1000 (milliseconds/second)
= 1000/3 (milliseconds/beats)

or...

333.33 milliseconds per beat @ 180bpm.

as far as what values to use for the attack + release,
that's entirely up to you.
but if you make it multiple or divisor of your tempo,
the compressor or limiter will "breathe" on the beat,
instead of at some interval/rhythm of its own.

[dj vorny]

very nice read this nice and usefull :clap:

[RDR]

For milliseconds per beat there is a slightly quicker calc. I posted a topic on this a while back.

60000/tempo = ms per qtr bar. :)

from there you can go anywhere including running particular tempos that equate to frequency.

Anyway i wont go back into it cos this is for compression.

:clap: for the GREAT posts, i love compression and am always looking to learn more!

[loopdon]

there's a nifty little program for all yer calculation needs:

Scal



Scal is a calculator dedicated to musical tasks. Delay time, lfo frequency, timestretch amounts, bpm determination plus many more are available.
Still available is the original SCalV4. Soon to be release are the VST version of Scal (for plugged in calculation inside you sequencer), plus Java versions for MacOs, Linux and any other Java compatible device

http://www.expdigital.co.uk

[RDR]

Aww fooey! i prefer the hard way!

:bash:

[rounser]

Ah, cheers to you both!

It's kind of conspicuous that this kind of basic formula isn't exactly common knowledge, and that utilities like SCal aren't requested much.