[ddots-l] Re: Tom Kingston's Article On Compression
- From: "neville" <neville@xxxxxxxxxxxxxxxx>
- To: <ddots-l@xxxxxxxxxxxxx>
- Date: Tue, 27 Jul 2010 10:52:34 -0400
I read the article and it was very informative. I tried the settings on my
vocal using the sonidus compressor. The results were not very good at all.
The vocal was very smashed and it pumped up and down like mad.
May the peace of God which passes all understanding guard your heart and
mind in Christ Jesus. God bless you!
Music soft sacred and soulful
Website http://www.nevillepeter.com
email neville@xxxxxxxxxxxxxxxx
phone 407-222-4488
-----Original Message-----
From: ddots-l-bounce@xxxxxxxxxxxxx [mailto:ddots-l-bounce@xxxxxxxxxxxxx] On
Behalf Of Phil Muir
Sent: Tuesday, July 27, 2010 10:35 AM
To: ddots-l@xxxxxxxxxxxxx
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
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