[ddots-l] Re: Tom Kingston's Article On Compression
- From: Bryan Smart <bryansmart@xxxxxxxxxxxxxx>
- To: "ddots-l@xxxxxxxxxxxxx" <ddots-l@xxxxxxxxxxxxx>
- Date: Fri, 30 Jul 2010 16:10:14 -0400
This is a great primer.
For those of you that either don't understand, or don't care to understand, the
subtleties of adjusting a compressor, and that have Sonar 8.5, then take a look
at the Transient Shaper effect. The transient shaper is a compressor that has
been modified to allow you to perform many compressor-type tasks, without
having to understand how a compressor operates. You can adjust the decay time
of a sound with a simple knob, while, behind the scenes, the effect's
compressor has its threshold lowered and its ratio increased to make that
happen. You can adjust many compressor-type characteristics in this way. Of
course, nothing will give you detailed control like operating a compressor
directly, but the transient shaper lets you operate a compressor from an
interface that uses controls that make more sense to people that aren't mixers.
Bryan
----- Original Message -----
From: "Phil Muir" <info@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
To: <ddots-l@xxxxxxxxxxxxx>
Sent: Tuesday, July 27, 2010 10:35 AM
Subject: [ddots-l] Tom Kingston's Article On Compression
> Tom Kingston wrote on the MIDIMag list: Well hello maggers, and thanks for
> tuning in. today's show will
> focus on a comprehensive breakdown of dynamic compression and
> what it can do for your recordings, not to mention your delicate
> musician's psyche. grin.
>
> But be forewarned! Don't feel out of the loop if dynamic
> compression initially throws you for a loop. It's one of those
> things that seems simple in concept, yet it can be overwhelming
> and complex when we examine and experience its specific elements
> and final results. Its potentially confusing nature is due in
> part to the incredibly complex nature of sound and our perception
> of it, and in part to the (usually) subtle and transparent nature
> of properly applied compression. So don't be afraid to read this
> over and over again while, and this is the most important part,
> you spend some serious time in your studio doing compression test
> runs, evaluating and comparing the results of various compressor
> configurations and applications. There are no simple rules, only
> basic guidelines to get you going and some specific points to
> keep in mind. Beyond that, it's all up to your ears. But the
> bottom line is this. Proper use of dynamic compression is no
> doubt the dividing line between the sound of a professional
> recording and that of a small project or private home studio.
> Even if you've got all the high end gear, and your recordings are
> full and clean and sparkling with clarity, the lack or improper
> use of compression is usually the root of that intangible feeling
> you get that there's just something different about your
> recording. It just doesn't have the smooth fullness that the
> commercial CD you're comparing it to has. So don't be discouraged
> if your first few attempts deliver you a sonic disaster.
> Befriending a compressor is like any other relationship in life,
> it takes time to get to know and learn to love your new friend.
> HaHaHa! So here goes.
>
> Let's first draw the line between 2 very different types of
> compression, which is what triggered this discussion in the first
> place.
>
> Dynamic compression has nothing to do with the way audio files
> are stored on a computer. The compression done there, like that
> done when converting a wave file into an MP3 file is called data
> compression. And due to the obviously confusing use of such a
> name tag, some people refer to this as data reduction rather than
> compression. Its only purpose in life is to shrink the size of a
> file; for example, compressing a 3 meg file into a 1 meg file. It
> works under the premise of discarding or abbreviating the least
> valuable bits in the stream. In other words, it tries to rid the
> file of the least audible bits and/or encode in a more efficient
> form bits that are redundant in a predictable manner.
>
> Dynamic compression, on the other hand, , is a critical phase of
> signal processing employed at various points in the recording
> process. There is virtually no such thing as a non-compressed
> recording on the market today. It compresses (narrows) the
> dynamic range of a signal. That is to say, it reduces the breadth
> of volume changes put out by that signal. Visually, this narrows
> the swing of audio level meters, while audibly it reduces the
> amount of volume change between the softest and loudest dips and
> peaks of a signal. It's an automatic volume control. Think of a
> compressor as being a device that automates the acts of an
> incredibly nimble engineer sitting at your console. He has set an
> imaginary zone on the meters within which your levels should stay
> in order to keep your sound smooth. When you start your power
> ballad very softly, he will raise the faders in order to project
> your soft sound. But when you slide into the chorus and crank it
> up, that engineer will pull those faders down to keep your
> overall volume changes less startling. Then when you calm down
> and slide back into the soft verse, he will again keep you in
> relative step by sliding the faders back up. But don't get me
> wrong. Compression does not replace fader movement. It simply
> reduces the amount of it we have to do when mixing.
>
> If you want to experience a good example in contrast of
> recordings employing lots of and very little compression, all you
> have to do is listen to any heavy metal tune, listening for
> volume changes, then compare that to an orchestral recording.
> Anyone who has ever listened to much orchestral music has no
> doubt experienced the need to continuously turn the volume up and
> down throughout the recording. This is because the classical
> genre prefers a purist approach wherein we hear pretty much
> exactly what they play, wild dynamics included. Rock-n-roll on
> the other hand prefers to play the loudness ticket and compress
> the heck out of their recordings. For example, while the tonal
> changes remain true to form, this results in very little volume
> change regardless of what the singer is doing, whispering or
> screaming. Why? Because our perception of volume works more to
> the average signal level rather than the actual hills & valleys
> of volume. Plus, at least in contemporary pop music, we perceive
> volume as a seemingly quantifiable indicator of clarity and
> overall quality. That's why the oldest trick in the book at hi-fi
> shops is to play the speakers they most want to sell you at just
> a little higher volume. Not much mind you, because it doesn't
> take a real perceptible up tick in volume to convince your sub-
> conscious that those speakers offer more clarity and definition.
> So this is why hard-rockers want their CD to be just a little bit
> louder than that of others. If you drop their disk into your
> multi-disk player and hit random play, their (louder) tune will
> sound just a little bit better.
>
> But is this increase in volume an actual clarifier? Well, yes.
> The reason is that a loudspeaker, with all the high-tech faces it
> holds today, is still a rather primitive device when compared
> with the audio output of our real world. No loudspeaker can come
> close to reproducing natural sound efficiently when trying to do
> so under the extraordinary burden of such extreme dynamics. While
> the sound of a gentle breeze is actually very similar to that of
> a gale force wind, that is when heard at equal volumes, the
> natural volume change between them is outrageous. And no speaker
> can efficiently reproduce the entire sweep of this dynamic
> spectrum. We could design a speaker specific to each sound that
> would actually work pretty good, but it's virtually impossible to
> design one that can handle such a spread in volume. This
> relatively narrow range of efficiency is why many people can't
> understand why their 5000 dollar 1200 watt speakers don't sound
> as good as their neighbors 500 dollar 120 watt speakers. It's
> because the smaller speakers can run at their most efficient
> (cleanest) levels while the super dooper power towers next door
> operate basically in a mumble mode. And while we're talking about
> this, I'd like to point out that this is in the forgiving world
> of home stereo speakers where they go to great lengths to expand
> the efficient range of speakers, simply due to the wide variance
> in usage. Professional PA speakers on the other hand are much
> more narrow in their efficiency design. This is why high end PA
> amps have no volume control. You should match the amp to the
> speakers and match the entire system to the output levels you
> need. This will get you the best sounding system in town. Many
> performers sound lousy only because they have a way too powerful
> system running in mumble mode. So keep that in mind.
>
> But back to the topic at hand. As I said earlier, be it that our
> perception of volume lies mostly in its average level,
> compressing a signal fools our ears into perceiving a virtual up
> tick in volume that solidifies and clarifies the sound we're
> listening to because we're not asking the speaker to work beyond
> its practical means. Therefore, while a compressor is actually
> narrowing the spread of volume changes, rather than simply
> cranking everything up, it is perceived as a volume increase
> because it raises the average level of a signal. And if we
>revisit for a moment the comparison between a rock band and a
> symphony, we can again learn why compression is more important to
> a rock band. This is because the typical mix of a rock band is
> very congested regarding instruments that crowd the same tonal
> frequencies. Guitars, piano and keyboards, and the voice are all
> competing for the same space. Conversely, an orchestra by
> definition is almost like a frequency map when you listen to each
> section. From the lower brass and cello, up to the piccolo and
> violas, they each have their own slot in the tonal spectrum. And
> classical composers work with this in mind, being careful not to
> have similar sections fighting for audible space. But when a
> frequency crowded mix is the case, compression can help alleviate
> one of the problems it creates. This is referred to as masking.
> All it means is that when 2 signals are competing for the same
> frequency slot, the loudest one will always prevail, basically
> eliminating the other from the mix. And on very dynamic signals,
> this can be very frustrating because the dueling pair will
> perform a maddening dance in and out of the mix, depending on
> which one happens to be a bit louder at any given point. And this
> (masking) by the way is why any soloed sound may sound like it's
> just what you're looking for, but when you drop it into the full
> mix, it seems to go pale. What to do. Stabilizing these signals
> with a compressor will give us the ability to make the
> adjustments needed to give each signal its own space. This
> usually amounts to reshaping the battling signals with an
> equalizer, reconsidering the timing of one, or simply re-writing
> the part for one of them.
>
> But wait! there's one more reason why compression becomes more
> necessary in a pop mix. And this is due to the manner in which we
> record. Don't let the purists down at symphony hall fool you.
> Even though they scowl at compression as if it's the work of the
> devil, the way in which they record their beloved orchestra
> employs natural compression. Even though they may mic a solo
> instrument, or mic the orchestra by section, they rely heavily on
> ambient mic's. These are mic's placed at a considerable distance
> from the orchestra, be it overhead or scattered out at various
> points in the auditorium. The purpose of this approach is to
> capture the ambient sound you would experience if you were in
> that concert hall. But the added benefit of space between the
> sound source and the microphone is that it creates a buffer zone
> that compresses the sound. Like with any other sound, distance
> dulls the dynamics. So this natural compression is the form they
> choose to use. But that's not to say that compression is never
> used on such recordings either. It's just that they for the most
> part use very slight compression, just enough to nip the stray
> transient peaks in the bud. Or they may reserve it for the final
> mastering phase of the album.
>
> But what the heck does that have to do with why pop music needs
> more compression! It has a lot to do with it. The way in which
> pop tunes are recorded is the complete opposite of the scenario I
> just outlined. We record everything direct, so there is no room
> for natural compression to have any effect. We work in inches
> when placing mic's in front of a singer, acoustic instruments, or
> guitar amps. We'll even mount microphones literally inside of
> drums or pianos. And if it's not a mic we're using, we plug the
> keyboard, drum-machine, or guitar amp line-out directly into our
> recorder or console. If it's a sound-card and recording program
> you're using, you again have the same basic setup where your
> sound-card outputs are plugged (routed) directly into your
> virtual console. And most sampled sounds, be they from a
> keyboard, sound module, or sound-card, are sampled (recorded) in
> a very direct manner. So it's all the same no matter how you
> slice it. And regardless of how you choose to look at this
> methodology, more or less true a form of recording, the bottom
> line is that the dynamics and tonal characteristics are amplified
> to the enth degree due to the proximity effect. That is, because
> the mic is on top of the sound source, or it is a direct feed,
> every little change in volume is tracked with much more precision
> than when the mic is placed at a greater distance, hence the
> natural compression effect. This is why it is actually harder to
> tweak a live vocal. Due to the fact that the singer must chew on
> the mic in order to keep the gain down and lessen the chance of
> extraneous signal bleed or feedback, this usually requires much
> more fiddling with EQ and compression to take the jagged edge off
> of the vocal sound. Stepping into a studio environment on the
> other hand offers 2 advantages. The first is that because the
> singer is using headphones, the engineer can crank up the mic
> gain and allow the singer to back off, thereby expanding the
> pickup of the mic and smoothing out or lessening the exaggerated
> dynamics. But even though this studio approach to vocals is
> indeed an improvement over the live environment, it still offers
> nowhere near the smooth response or natural compression inherent
> in a mic placed 30 feet away. So our only choice is compression.
> Oh yes, and EQ as well, but that's a whole other discussion.
> grin.
>
> So where when and how do we apply compression?
> Wow! what a loaded question! HaHaHa!
> And not an easy one to answer, but here goes.
> Just don't forget, all of what follows is no more than a set of
> basic guide-posts to get you going. There are no hard & fast
> rules for compression. Trust me. If you had the priceless
> opportunity to sit down and chat with 20 of the most sought after
> engineers in the industry, and asked them each to outline their
> compression technique and philosophy, I guarantee you would hear
> 20 very different answers. This is why you, when comparing 2
> albums, may prefer the overall sound of the first, while you
> actually like the tunes more on the second. Naturally, this goes
> beyond compression and into EQ and recording technique as well,
> but compression is a big part of the overall feel of any music.
> But that's enough with the psycho-acoustics for now! Let's move
> on to the technical heart of the matter.
>
> When to compress.
> The basic rule of thumb is to employ light to moderate
> compression on signals going to tape. You just want enough to
> give the recorder a good strong signal to work with and reduce
> the chance of your soft signals flirting with the noise floor.
> But, having had said that, the more accustomed you become to
> compression, the more you will discover how much you need on what
> signals. So over time you may be able to be more aggressive with
> compression at the input phase and reduce or at least make easier
> the next phase of compression. The closer you can get the
> incoming signals to their final state, the quicker and easier
> your mix will fall into form. And this will make your mixing
> chores considerably easier.
>
> Which leads us to our next layer of compression. Mix-down.
> Here we can do more compression to fit each track more into the
> mix the way we want it. Let's say that your vocals need just a
> little more steadying, or the bass doesn't quite fill the groove
> the way you want it to, or you'd like to just take the edge off
> of the guitar a bit, or, maybe you want to push the drums a bit
> further back in the mix. No problem. Compression can work all
> these wonders. So you fiddle to no end and get everything
> sounding just the way you want it. Now you mix down your entire
> tune.
>
> But wait! not done yet.
> The next step of compression is part of what's called the
> mastering phase. This, though usually done by a mastering house,
> can be done to a lesser degree if you've got the tools to do it
> with. All it amounts to, and that's the understatement of the
> century, is the final phase of very discrete overall compression
> and EQ applied to the entire mix. It usually employs what's known
> as Split Frequency Compression. All this means is that the tune
> is broken up into frequency blocks, like highs mids and lows,
> each of which is compressed separately, because these different
> frequency blocks react somewhat differently to compression, which
> we'll get to in a bit. But suffice it to say that these frequency
> zones of your material are compressed just a little bit more and
> then reassembled back into the whole on the final master tracks
> to be used for duplication.
>
> OK, so that's a basic overview of when compression is applied.
> Now let's get to the nuts & bolts of actually applying it. First,
> I'll give an extensive breakdown and description of the typical
> controls on a compressor, sometimes referred to as a
> compressor/limiter. I'll then give some generic examples of
> compressor settings for various signals.
>
> But before we dive into this, there's one more thing I must first
> clarify. And that's the metering system being used in my
> examples. The digital console and recorder I use employs the
> digital metering system which is a bit different from the VU
> metering of a standard analog deck. It operates on a negative
> numbering system where 0 Db indicates the clipping threshold of a
> signal. All this means is that in practical terms, a digital
> signal cannot go over that 0 Db peak limit. Unlike on an analog
> deck, there's no such thing as soft distortion, that is, the act
> of pushing the signals so that they distort just enough to
> conjure up some nice fuzzy warmth, or even a smooth audible
> distortion, common on guitars. Digital systems are not capable of
> creating this type of harmonic distortion in this manner. It's
> all a numbers game in the digital environment, and the 0 Db peak
> limit is a non-flexible absolute. Beyond this point, the actual
> wave form has its peak clipped off as if it had tried to go
> higher but slammed into a wall. And if you clip a large enough
> portion of a signal, it will scream at you in pain. So this is
> the level I'm referencing when I speak of any metering, such as,
> threshold and peak levels.
>
> And now for the controls.
>
> 1 Threshold. Typical range: -60 to 0 Db.
> This sets the level at which compression kicks in. When a signal
> crosses over the threshold, the compressor takes notice and makes
> its move in accordance with the other settings you've configured.
> Average setting: -20 Db.
> Tips.
> A higher threshold (-16 to -12) is used when all you want to do
> is smooth out a signal, such as an entire mix. An even higher (-
> 12 to -8) threshold may be used when you want to simply grab a
> signal (limit it) at that point, and boost it up to a constant
> level, sometimes used for bass guitar or vocals. A lower
> threshold (-28 to -38) allows you to compress the entirety of a
> signal. This for example has a lot to do with how breathy a vocal
> is, or how loud fingering noise is, because the lower the
> threshold, the more the lower or softer parts will also be
> compressed.
>
> 2. Attack. Range: 0 to 250 milliseconds. Average setting: 10.
> This determines how long a signal must stay over the threshold
> before compression actually begins. The reason for this delay is
> that we often want the leading edge, or peak transient of a sound
> to have a chance to make its point before we call in the troops
> and beat it back down. A slower attack time will allow sound
> elements such as, the initial strike of a drum or piano, the
> pluck of a guitar string, or the emphasis of the voice to lead,
> or articulate, the sound before it's compressed.
> Tips.
> The lower the threshold, the more critical this setting becomes
> because you're then working at the start level of a sound. If you
> compress it before it has time to ramp up, you're going to dull
> the heck out of that sound. And this may be exactly what you want
> on a fat bass guitar or drum sounds. But it's probably not what
> you want on an acoustic guitar because it will dull the overall
> clarity of the sound. This also has a much more apparent effect
> on high frequency sounds, or the high frequency components of
> otherwise low frequency sounds, such as: the finger or fret noise
> of a bass. Because high frequency components are usually the
> leading edge of most sounds, subtle adjustments of the attack
> time can have enormous effect on the perceived placement and
> sharpness of a sound.
> For example, I was just recently playing around with the entire
> setup for my snare drum, trying to tweak more of the shell sound
> out of the drum. And when I started playing around with the
> compressor, I discovered a 5 millisecond window of complete
> control over that drum. When I quickened the attack time, the
> compressor held back just enough of the initial strike snap to
> allow more of the body tone to come through. But if I really
> slammed down a quick attack, I heard the drum move to the back of
> the studio as if it had been pushed 10 feet back from its mic. So
> always play around with the attack time.
>
> On the other hand, longer (30 to 50 millisecond) attack times are
> used when doing overall compression of a mix because it lets the
> tune swell and sway within reason, just pulling in the reins when
> a sustained surge comes, such as ramping up and in to a chorus.
>
> But for the most part, this is the one setting that should really
> be played around with for each signal. It can easily make or
> break the sound you're looking for, or maybe already have.
>
> 3 Release. Range: 5 to 2000 milliseconds. Average: 100.
> This determines how long a signal must fall and stay beneath the
> threshold before the compressor actually lets go of it.
> Tips.
> Having too short a release time on a signal can cause a pumping
> or breathing effect because the compressor is trying like mad to
> chase a punchy signal, or it's letting go during slight breaks in
> the music, ramping the noise floor up, then letting it fall back
> down. . Too short a release can also cause distortion in low
> frequency signals, such as the bass guitar and drums. But the
> good side of a shorter release is that it can keep the punch in
> the music. A longer release, on the other hand, smoothes out the
> overall flow of the piece, but at the expense of some definition.
> So it all depends on the tune, or even the section of the tune
> you're working on. For example, ballads usually use a longer
> release while punchy tunes have a shorter overall release.
>
> 4 Ratio. Range: 1 to 1 up to 100 to 1 infinite.
> This determines how much the signal is compressed once it hangs
> out over the threshold for the duration of the attack time. The
> actual ratios available may go something like this.
> 1 to 1, 1.5 to 1, 2 to 1, 2.5 to 1, 3 to 1, 4 to 1, 5 to 1,
> 6 to 1, 7 to 1, 8 to 1, 10 to 1, 20 to 1, 30 to 1, 40 to 1,
> 50 to 1, 100 to 1, infinite.
> Example: If a 4 to 1 ratio is being used, for every 4 db your
> signal moves over the threshold, the compressor will only allow
> it to move 1 db. So if it peaks at 8 db over the threshold, the
> compressor will only allow it to peak 2 db over the threshold.
>
> Tips.
> In general, the lower the threshold, the lower the ratio, and the
> higher the threshold, the higher the ratio. This is because with
> a low threshold, you're compressing more the entirety of the
> sound, some of which may be noisy elements. A good example of
> which would be the breathiness of vocals and even the crackling
> of lips. While the breath tones may be just what you're looking
> for, you'll soon discover that it's all or nothing. Therefore,
> the unwanted pops and even your movement in front of the mic will
> also be much more apparent. So this usually requires considerable
> playing around with the settings in order to find the best
> compromise.
>
> Conversely, you'll want to hit a high threshold harder because
> you have less room to play with on the meters. But all rules are
> made to be broken. You may just want to soften up the peaks just
> a little bit, so you use a gentle 1.5 to 1 ratio on them at a
> high (-6) level.
>
> But let's now look at the other extreme of ratio settings.
> In practical terms, a 10 to 1 or higher ratio is considered a
> limiter. In other words, it is limiting that signal (stopping it)
> dead in its tracks at the threshold level. This, as I said
> earlier, may be used to grab a signal and crank it up to a rock
> steady level, by limiting it let's say at -10 Db, then boosting
> it up 4 db to sit it right there at -6 Db. This method may also
> be used to add just a slight overall boost to a finished mix by
> placing the limiter at -3 db and boosting it up 2 db. Limiting
> however is a tough call. Some people like it, and some think it
> strips the signal of its life. The call is yours.
>
> Limiting can also be used as a protective type of compression. It
> can be used on signals just to stop any stray peaks from ruining
> an otherwise good recording, or, it may be used as an overload
> protector for your speakers in a live environment. In either
> case, you're simply placing a limiter at the point beyond which
> you want no signal to go.
>
> And finally, back to the other end of the spectrum and the 1 to 1
> ratio. This is a bypass ratio that makes the compressor do
> nothing. It may be used for quick comparisons while working out
> your settings, or, it may be used when all you want to use is
> another feature of the compressor, such as, a noise gate or
> downward expander.
>
> 5 Gain. Range: 0 to 12 or 15 Db.
> This is how you compensate for the gain reduction being done by
> the compressor. In general, it's a simple corollary to the ratio
> you're using. For example, if using a 4 to 1 ratio, adding 4 Db
> of output gain will pull the compressed signal back up to the
> correct level. Sometimes, particularly on vocals, and when using
> a low threshold and ratio, even more gain may be used to power up
> the vocals. Although this is usually done with 2 compressors in
> series where the second unit is simply a high threshold limiter
> waiting to keep those power-house vocals in check.
>
> But the most important use of the gain knob is simply to
> compensate for the compression. No signal likes to be squashed,
> so forgetting to compensate correctly will result in significant
> degradation of signal quality.
>
> That's it for the basic control knobs.
> But a few additional switches may also be available.
> 1 Bypass: pretty self explanatory.
> 2 Stereo link: This allows you to control both channels from one
> set of controllers. And this is very important when compressing
> stereo signals. Lack of in sync settings can make sounds wander.
> 3 Noise gate or Downward expander.
> Consists of at least a threshold knob, and perhaps attack and
> release knobs as well.
> This tries to cut low signal noise during silent parts. When a
> signal drops beneath the threshold, a noise gate acts as a simple
> gate and closes, basically cutting the signal off. Then when the
> signal comes back above the threshold, the gate reopens and lets
> it come through. The attack and release time work just like on
> the compressor. The signal has to stay below the threshold for
> the duration of the attack time before the gate will close and it
> has to come back up and stay above the threshold for the duration
> of the release time before it will actually be let through.
>
> The difference in a downward expander is that it's a more smooth
> processing of the signal. The gate in effect is just a simple
> switching device. It's either on or off. But the expander employs
> a gradual slope to signal reduction. It also has a ratio setting,
> so it works basically like a compressor, in reverse. This
> alleviates the potential audibility of a noise gate that simply
> flips open or closed.
>
> And now for some practical examples.
>
> Generic vocals:
> -28 threshold, 4 to 1 ratio, 0 attack, 500 release, +4 gain.
> Power vocals:
> -38 threshold, 2.5 to 1 ratio, 12 attack, 180 release, +5 gain.
> Note that compressing vocals too much can be difficult for the
> singer because they will not be able to hear their louder parts.
> The voice in their head will drown out that in their monitors. So
> it is sometimes necessary to use minimal compression while
> recording and add more during mix-down.
> Guitar:
> -20 threshold, 3 to 1 ratio, 20 attack, 200 release, +3 gain.
> Bass guitar: -24 threshold, 4 to 1 ratio, 10 attack, 250 release,
> +4 gain.
> Piano: -36 threshold, 1.8 to 1 ratio, 3 attack, 100 release, +2
> gain.
> Note that piano compression in particular seems to be very
> noticeable. Obviously, some people don't mind this because I can
> hear it in many commercial releases. I hear the strike, then it's
> as if it's yanked back by the compressor. So that's why I use a
> low threshold and ratio, with a quick attack. This lessens the
> audibility of the compressor.
> Drums: Very tricky!
> It would be easy to write an entire book on drum compression. The
> problem is that the typical drum set includes too many very
> different types of sounds, from the bass to cymbals, all that
> react very differently to compression. And this is why drums
> really need individual compressors. Bass drums need a longer
> (200) release in order to prevent flutter distortion, but cymbals
> need a shorter release (100) in order to prevent wavering of the
> cymbal decay. Plus cymbals usually require the quickest attack,
> while toms and the bass can tolerate flexibility in this
> department, depending on what you're looking for. And the snare
> is a whole other game, particularly when setting the attack time.
> This can completely change the sound of that drum. And all this
> changes depending on how much or less you're trying to compress
> the drums.
>
> But for the sake of this discussion, I would assume that most of
> you are probably not working compression on a drum-by-drum basis
> anyway. So the best thing I can say is to just start your overall
> drum compression with the standard -20 threshold, 4 to 1 ratio,
> 10 attack, 100 release, and +4 gain. Then spend some serious time
> playing around with the attack time as this will have the most
> profound effect. Then fiddle a bit with the release time, paying
> particular attention to the low end of the groove. Then you might
> even try lowering the ratio a bit while leaving the gain up. This
> can sometimes pump up the sound a bit.
>
> As far as full mix compression goes, this is a tough call as well
> because it has a lot to do with the type of tune you're working
> with. But here's a generic configuration to try.
> -16 threshold, 3 to 1 ratio, 30 attack, 50 release, +3 gain.
> Note that the low frequency sounds are not as suseptable to the
> distortion problems caused by a short release when the threshold
> is higher. But you should still keep an ear on this.
>
> And if all else fails, try everything in between. grin. You're
> sound is out there somewhere. Now go and find it!
>
> So that's about it. Aren't you glad! HaHaHa!
>
> In conclusion, all I can do is remind you to take all of those
> settings with a grain of salt. The simple fact of personal
> preference and opinion in sound makes it impossible to define
> what the right settings are. Plus, there are many very different
> compressor designs in use these days, so they all tend to vary to
> some extent on how they react under the same conditions. And the
> final thing is that there is a line that has to be drawn
> depending on the quality of compressor you're using. A high-end
> 2000 dollar compressor is going to be virtually transparent no
> matter what you ask it to do. So the sky is the limit if that's
> what you have to work with. But if it's a 99 dollar compressor
> you're working with, be aware of this, and be conservative on
> what you ask it to do. Heavy compression with a low quality
> compressor can suck the life right out of your otherwise great
> recording. So be careful and listen hard. Do a lot of comparing
> and make sure that there is indeed a sustainable improvement in
> your compressed material.
>
> And finally finally finally!
> This all goes back to the thread that started this discussion.
> The reason I suggested compressing your material more when
> converting into MP3 format, even beyond the norm for playback in
> a conventional system, is that narrowing the dynamic gaps and
> raising the overall level will give less space for the noise to
> poke through. In effect, you're masking the noise. So the bottom
> line is simply to give your MP3 encoder the hottest possible
> signal to work with.
>
> Good luck, Tom.
>
> Regards, Phil Muir
> Accessibility Training
> Telephone: US (615) 713-2021
> UK +44-1747-821-794
> Mobile: UK +44-7968-136-246
> E-mail:
> info@xxxxxxxxxxxxxxxxxxxxxxxxxxx
> URL:
> http://www.accessibilitytraining.co.uk/
>
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