Many
of today's digital effects processors offer you considerable
control over the creation of artificial ambiences for your
music, and if you know how reverberation works in real spaces,
you'll be better equipped for designing fake ones
Reverberation is
something that few people are consciously aware of, yet it
is one of the most fundamental aspects of a room's sound character.
If you were to blindfold someone, take them to an unfamiliar
building and lead them through a succession of rooms, clapping
or shouting in each one, they would almost certainly be able
to give a pretty accurate description of the size of each
room. If they were being particularly perceptive, they might
even be able to suggest where they were standing in each room
and probably even give some idea of what there was in terms
of wall coverings, curtains, soft furnishings and so on! In
other words, it is reverberation that gives your brain most
of the information it needs to create an aural picture of
your immediate environment.
Every room has
its own sound or 'acoustic', and part of the job of a recording
engineer is to assess whether a room's characteristic sound
is worth using in a recording. If the dimensions, layout and
fabric of the room enhance the recorded sound quality, all
well and good, but if not, the microphone technique used should
minimise any room sound so that an artificial acoustic can
be added later from a reverb processor.
In this age of
digital technology, artificial reverberation is not only more
affordable than ever before, but can also be stunningly realistic
and very controllable. With a good understanding of the physics
of natural reverberation, and the fundamental operational
principles of reverb processors, it is possible to quickly
create the illusion of any acoustic environment you can imagine.
Remember, though,
that if you choose to use artificial reverb, it is essential
that the recording has the absolute minimum of the recording
venue's room sound. If the original environment can be heard,
adding extra 'fake' reverbs will just result in a cluttered
sound and the mix will often become confused and indistinct.
TIMING
To
understand what a reverberant sound actually is and what information
our hearing system is able to extract from it, you need to
think about how sound waves travel and what happens when they
encounter various surfaces.
The first thing
to consider is how fast sound travels in air. My old school
physics books said 760mph and the new ones probably say 340
metres per second, but I find it hard to relate these numbers
to anything meaningful (other than the average speed of traffic
around the Evesham bypass in the morning....).
A much more useful
figure to tuck away in the dark and dusty recesses of your
mind is that sound travels roughly a foot each millisecond.
Assuming that you're not too young to relate to feet as a
valid dimensional measurement, this rule of thumb will allow
you to calculate and set one of the most critical parameters
of any reverb processor. One quick side note: the speed of
sound varies with the condition of the air. Temperature, humidity
and pressure (ie. altitude) all have significant effects on
the speed of sound, and in certain applications the 1ft/ms
guide is not sufficiently accurate. However, as far as dialling
up room sounds on a reverb unit is concerned, it's close enough.
Imagine that you're standing in the middle of a very large,
brick-walled barn with a deep covering of straw on the ground.
You have a spontaneous urge to clap your hands: what happens?
Well, the very first thing you'll hear is the direct sound
of your hand clap, and it's this direct sound that the brain
uses to pinpoint the direction of the sound source.
Assuming that the
hand-clap radiates sound waves in all directions simultaneously,
the next thing you'll hear will be reflected sound from the
various room boundaries or nearby objects. As the floor of
this imaginary barn is covered with a deep layer of straw,
there will be no significant reflection from the ground (although
in practice this is often a dominant source), so the first
'room sounds' will be reflections from the side walls and
ceiling. If the barn measures 40ft by 60ft and is 20ft high,
and assuming we're standing in the middle of it, the very
first reflection will come from the roof, after about 30 milliseconds.
This time delay
can be estimated by working out the distance the sound wave
has to travel -- if you're a man standing up in the barn,
your hands and ears are likely to be about four and six feet
respectively above the floor, so (20-4) + (20-6) = 30. Sound
travels roughly 1 foot a millisecond, so it will take 30ms
for the sound of the clap to reach the ceiling and return
to your ears. Similar calculations reveal that the side-wall
reflections occur 40 and 60ms after the initial direct sound.
"...artificial
reverberation is not only more affordable than ever before,
but can also be stunningly realistic and very controllable."
The time taken
for these first reflections to be heard goes a long way towards
defining the perceived acoustic size of the room. Short delays
imply small rooms, and long delays large rooms. To mimic this
natural characteristic, artificial reverberation units normally
allow the user to set the time delay between the direct sound
and the very first reflection with a parameter called Pre-Delay.
This is generally adjustable in millisecond increments (or
finer) over an astonishingly wide range.
So the first reflection
comes from the closest surface, and is followed by others
from the adjacent room boundaries, the whole ensemble being
known as Early Reflections. The timing, relative amplitudes
and timbre of these individual reflections are determined
by three things: the placing, angles and nature of reflective
surfaces; the location of the sound source; and the position
of the listener. Moving any of these will change the relative
timing of the early reflections quite significantly, but our
hearing system is remarkably good at extracting even the most
subtle information. Consequently, these initial reflections
and their relative timing are very important in defining an
imaginary acoustic space.
The better reverb
units allow the user to alter not only the value of the pre-delay,
but also the number, grouping, timing, amplitudes and tonal
qualities of the first reflections. In some cases, these parameters
are preset by the manufacturer and are simply selected from
a list of programmed options, such as Hall, Chamber, or Plate,
although occasionally they are derived from measurements taken
in genuine acoustic environments. The most sophisticated machines
allow the user to specify the cubic volume of the imaginary
room, or even its precise dimensions, together with the source
and the listening positions!
TAILS
So first we hear the direct sound; then, a short time later,
a number of discrete reflections return from the various surfaces
in the room. However, these reflections don't just stop when
they reach the listener -- they continue until they reach
other surfaces, where they instigate more reflections. These
reflections start even more reflections and the sound density
becomes too great to allow us to distinguish the reflections
as separate events.
At each reflective
surface, some of the sound energy is absorbed, and more is
lost as the sound travels through the air -- which is why
reverberation gradually dies away. This reverberation 'tail'
may last for anything from, say, 0.3 seconds (for a dead-sounding
room), through to several seconds in a church or big concert
hall.
The length of the
reverberation tail is usually specified in terms of its 'RT60'.
This is defined as the time taken for the reverberation to
fall by 60dB in level below the original direct sound. Every
reverb unit allows this time to be adjusted, normally through
a parameter called Decay Time.
It's important
to note that the reverberation tail lasts for different durations
at different frequencies. High-frequency sound waves have
a lot of trouble persuading air molecules to vibrate quickly
enough to pass the sound energy onwards. Consequently, high-frequency
sounds tend to die away, as they travel, much faster than
mid-frequency sounds. On top of that, high-frequency sounds
are absorbed by soft furnishings (which includes people and
even wallpaper!). On the other hand, high frequencies reflect
strongly from a wide range of surfaces, such as windows, sound
desks, equipment racks, and so on. At the other frequency
extreme, low frequencies are only reflected by large and very
solid objects, so there may be little LF in the reverberation
at all in some circumstances, but a definite bass 'bloom'
in cave-like rooms!
To help provide
this level of realism, most reverb units allow you to adjust
the reverberation time for high (and sometimes low) frequencies
relative to middle frequencies, and introduce some kind of
overall equalisation to the reverberation tail.
KNOBS
Let's
recap on the parameters that today's digital reverb processors
are likely to offer for the simulation of real acoustic spaces.
Firstly, although I haven't previously mentioned it, there
is usually a means of balancing the direct sound against the
reverb. The direct sound is often referred to as 'dry' and
the reverberation as 'wet', so a wet/dry control will probably
be in there somewhere. Some of you will know how unpleasant
it sounds when an analogue-to-digital (A/D) converter is overloaded,
so a critical control on digital reverbs is the input level
control, and its associated headroom meter.
The first control
which defines the reverb character is pre-delay, which effectively
defines the distance of the first reflective surface. This
will be followed by one or more parameters for controlling
the number, timing, amplitude, and timbre of the other early
reflections. Some machines provide controls called 'Pattern',
'Level' and 'Room Size'; others might simply offer preset
venue simulations ('Hall', 'Chamber', 'Jazz Club', and so
on).
After the early
reflections, the reverberation tail is set by a control for
overall decay time. This is normally accompanied by a parameter
that adjusts the relative decay time at high (and perhaps)
low frequencies. There's usually also some means of setting
overall tonal characteristics, although this may be little
more than a simple bass and treble equaliser.
Finally, having
skilfully set all of these parameters to create a wonderfully
believable artificial acoustic(!), you can usually store your
fake room in one of a number of user memories.
CREATING
SPACES
You cannot create an artificial acoustic space if you don't
know what a real one sounds like to start with, and the only
way you can find out is to actively listen to sounds in as
wide a range of environments and circumstances as possible.
Everyone has a very detailed subconscious knowledge of how
different rooms sound, and although few are able to analyse
the reverberation structure, most spot incongruities in artificial
reverberation very easily.
It's very revealing
and informative to consciously listen to the sound of different
rooms as you move around in a building -- even in places with
which you thought you were familiar. Try to analyse in your
own mind what sort of pre-delay, decay time, early reflections
and high-/low-frequency decays naturally occur to create the
'sound' of that room. Don't just listen to indoor reverberations,
either -- try to assess the reverberant features of the local
high street, the great outdoors, a wood or forest, or wherever
you happen to be. You will find reverberation in places you
didn't expect it, and may be surprised to discover that places
you assumed to be reverberant actually are not!
MONO
AND STEREO
The very nature of genuine reverberation is that it tends
to come at you from all over the place, but particularly from
the sides of the room. This has significant effects on the
compatibility between stereo and mono versions of your mix,
since the mono listener is effectively denied any information
from the sides of the stereo image.
To see how this
happens, consider a simple M&S (Middle & Side) stereo
microphone technique being used to record something in a reverberant
room: the stereo listener hears the full acoustic in all its
glory, but the mono listener hears only the forward-facing
'M' microphone, not the sideways-facing 'S' microphone --
and guess which one picks up the bulk of the room sound? This
absence of reverberation in mono afflicts artificial reverb
processors as well as natural acoustics. In practice, the
amount of reverberation heard in mono may be substantially
less than that in the stereo balance, and if mono listeners
are likely to be an important part of your music audience,
always check for mono compatibility. In general, you almost
always have to compromise the balance in some way because
either the mono will be too dry, or the stereo will be too
wet!
"You
cannot create an artificial acoustic space if you don't know
what a real one sounds like to start with...""You
will find reverberation in places you didn't expect it."
Tricks worth trying
include reducing the stereo width of the reverb (turn the
pan-pots in a bit towards the centre instead of having the
reverb returns running out to full left and right), or mix
in a small amount of reverb from another reverb processor,
panned centrally. The extra reverb should be set up with the
same parameter values as the stereo reverb, although a slightly
shorter pre-delay and longer HF decay time often work well.
The balance between the dry sound, the mono reverb, and the
stereo reverb needs to be adjusted carefully, while you continuously
switch between mono and stereo listening to find the most
uniform results in the two modes.
In matrix surround
systems (such as Dolby Surround), real or artificial stereo
reverb tends to spread across the rear channel quite naturally
as a result of the way in which the rear-channel information
is encoded and decoded. Altering its stereo width controls
the front-back balance, narrowing the reverb pulls it to the
front, and increasing the width pushes more to the sides and
rear.
Many stereo digital
reverb units have a single input and a stereo output, and
this often causes people to wonder how the reverberation can
be 'true stereo' with only a mono input. The answer is simple
if you consider the real situation of a sound source within
a reverberant space.
If someone claps,
there's only one sound source, yet the reverberation will
come from all directions and could be captured by a simple
stereo microphone array -- a mono input to the room and a
stereo output from it. Of course, in a more complex situation
with, say, a string quartet in the room, there are multiple
sound sources and each will have slightly different pre-delays
and early reflection patterns, but this is usually a very
subtle distinction, and in practice the mono-in, stereo-out
system of most digital reverb units works perfectly adequately.
Something few people
ever check is the line-up of a stereo reverb unit. However,
it is a stereo source and should be treated in just the same
way as any other stereo signal, which means making sure that
the left and right reverb outputs have the same gain and equalisation
through your mixer. I find that a quick, easy and reliable
method of doing this is to simply dial up a 3- or 4-second
decay time and send a brief burst of signal into the machine.
Listen carefully to the dying reverb tail: it should decay
centrally, possibly even becoming narrower in width as it
goes (although this depends on the particular algorithm).
If the reverb tail appears to collapse towards one side or
the other, your return channels have different gains and should
be adjusted.
CHOOSING
AND USING REVERB
In general, two reverb units will meet the needs of pretty
much every recording situation. One machine would normally
be set for a short, bright sound (perhaps a plate setting)
for percussive sounds, whilst the other would be set to a
longer, warmer patch, providing a 'lush' quality for vocals
and solo instruments. You could also try passing some instruments
through both reverbs (percussive one first) for a third alternative.
Some engineers
like to use several reverbs to create a layering effect, but
I generally find that this approach causes a loss of definition
and adds confusion to the overall sound. Going back to the
idea that the artificial reverb is merely replacing the poor
acoustics of a less-than-ideal recording venue, it could be
argued that there should only be one reverberation sound for
everything, as would be the case if the musicians all played
live in the same reverberant room!
The next issue
to address is how much reverb to use. The classic mistakes
of the novice are using too much reverb return on everything
and allowing reverb tails to be too long. Reverb generally
needs to be subtle, and ideally only the loudest musical peaks
should cause obvious tails. Even the biggest halls rarely
have a reverb time in excess of four seconds, and often a
two-second decay time is easily long enough.
The choice of reverb
parameters is dependent on both personal taste and the nature
of the programme material, so it's impossible to give specific
recommendations, but try to create life-like environments
wherever possible. Most reverb units offer a number of special
effects, such as gated or reverse reverbs, and these are best
used sparingly, so that they keep their impact. While we're
on the subject of special effects, it's worth trying out the
pseudo-reverberation programs too. Algorithms such as 'Ambience'
or 'Alive' can often add extra definition and life to dull
vocals, or spice up closely-mic'd solo instruments without
your having to resort to using those horrible exciters (just
a personal opinion, of course...).
Normally, reverb
sends are taken post-fader, so that direct signal level adjustments
are reflected in their reverb returns. However, it can often
be useful to send pre-fader, and not allow any direct signal
into the final mix at all. This is particularly effective
with sustained keyboard string sounds and the like, where
it helps to make less-than-ideal synth sounds blend a lot
more smoothly.
Another useful
trick is to set up a reverb specifically for the keyboard
sounds, and route the reverb returns through a chorus unit.
This provides a completely different kind of sound to chorusing
the keyboards directly and adds an interesting 'swirling'
quality which can be very effective if used discreetly.
Although reverb
processors are most important during mixdown, they're also
vital during recording, especially when recording vocals.
Many singers have enormous trouble pitching properly without
reverberation and it's essential to have the ability to route
reverb returns to the headphone monitor mix. The reverb setting
for the cue monitor is not particularly critical to the performance
(provided it is broadly appropriate) and need not be recorded,
although some engineers do like to record voice and reverb
together (occasionally as a complete mix but more usually
on adjacent channels on the multitrack machine). This is particularly
useful if the reverb plays a part in the performance (through
timing or percussive vocal effects, for example).
As artificial reverb
becomes more and more elaborate, there's a trap which many
engineers find themselves falling into. It's possible to become
so engrossed in adjusting each parameter minutely, trying
every possible combination along the way, that you lose sight
of the original idea. The best way of getting the sound you
want, quickly, is to understand the nature of real reverberation
and apply that knowledge to creating the acoustic space you've
imagined. It's far better to think for a minute or two, and
then dial the right numbers in, than to sequentially try every
preset on the machine, hoping to stumble across something
that sounds OK.
ONE
ALGORITHM OR TWO?
A word of warning -- not every reverb processor is as flexible
as it might seem. Particularly with multi-effects units, it
is quite common to find that there is actually only one reverberation
algorithm. The wide range of supplied preset environments
(Hall, Room, Plate, and so on) is actually composed of variations
in the delay, decay and EQ settings of a single algorithm.
In these cases, you'll find that no matter how you adjust
the reverb parameters, all settings sound very similar: the
overall character of the room does not seem to change, and
this is because the pattern of the early reflections remains
fixed. The better machines have a number of different algorithms
and a variety of early-reflection patterns, which allow a
larger range of different room types to be created, each with
distinct and individual sonic characters.
Fortunately, there
is an easy way to find out which category a particular machine
falls into. Select two, theoretically diverse, programs --
perhaps a Hall and Plate. Set the delay, decay, EQ and any
other parameters to identical values and store the new settings
in a couple of user memories so that they can be recalled
easily. Next, listen critically to the quality of the reverberation
while switching between the two presets. There should be an
obvious difference in the character of the room acoustic if
the machine uses different algorithms, with different early
reflection patterns. (Try closing your eyes and imagining
the dimensions and furnishings of the fake room.) If you cannot
spot any differences, the chances are that the machine uses
the same algorithm for all its reverb programs.
THE
MUSEUM OF ARTIFICIAL REVERBERATION
Artificial
reverb has been around for more or less as long as people
have been performing in non-ideal acoustic environments. In
more recent times, however, various electronic and electro-mechanical
methods have been developed, although none was as effective
as the current generation of digital designs.
One of the simplest
and most obvious systems was the echo room -- literally a
room, often tiled and full of ceramic sewer pipes to provide
a wealth of reflective surfaces. A loudspeaker generated the
direct sound, and one or more microphones collected the resulting
reverberation. The echo room has the advantage that the reverb
is naturally very complex, but it is also difficult to adjust,
and requires a large and quiet room!.
One of the first
electro-mechanical systems (and one which remains popular
to this day) is the plate. This employs a large sheet of metal
(typically 6ft by 4ft) suspended on springs within a sound-deadening
case as a reverberant space. A vibrating transducer feeds
the direct sound into the metal plate and a pair of pick-ups
extract the reverberation as the vibrations bounce off the
plate's edges. A motorised damping plate parallel to the main
one can be remotely positioned at varying distances to control
the duration of the reverb. The plate has a characteristic
metallic, bright, sound quality which has become intimately
associated with pop music. Virtually every digital reverb
I have used includes a simulation of the humble plate -- which
is a very good indicator of just how popular this mechanical
system remains!
Another enormously
popular electro-mechanical system is the spring-line reverb.
This technique has been around for a very long time, and I'm
sure everyone has come across guitar amps with spring reverbs
installed. The operating principle is similar to the plate,
in that a transducer sets up vibrations in a spring, which
rattle back and forth, to be extracted by a pick-up at the
other end. The character of a particular spring reverb unit
is fixed (other than the wet/dry balance), but can be optimised
for the sound source at the design stage by careful choice
of the number, length, diameter and compliance of the spring(s).
All manner of record-replay
systems have been developed to provide a reverb effect, but
none have survived the digital revolution. The earliest ideas
simply used a three-head tape machine, where the direct signal
was recorded onto tape and the replay signal provided the
reverberation. The tape speed and head spacing determined
the pre-delay and if some of the replay signal was mixed with
the direct signal, a pseudo-reverb could be created. The results
are hardly realistic, but the system was popular at a time
when the alternatives were too expensive or impractical.
The record-replay
theme was further developed into machines like the WEM Copycat
and the Roland Space Echo, which used tape loops and multiple
replay heads, with the ability to adjust the contribution
and feedback of each head -- but then solid-state technology
arrived...
Bucket Brigade
systems became popular for a brief time (fortunately) just
before the first true digital reverb machines hit the market.
Bucket Brigade delays were a halfway house between analogue
and digital systems, but were no more realistic or flexible
than the earlier tape-loop products -- and were often a lot
noisier!
The advent of digital
technology really revolutionised artificial reverb, basically
because the time-domain signal processing of digital audio
lends itself very well to the kind of sound manipulation needed
to create realistic reverb. Creating a pre-delay is simply
a case of storing sound in a memory until the required time
has passed. The early reflections are created by replaying
the direct sound repeatedly at suitable moments, with level
and equalisation changes as necessary. The main body of the
decay is created by cycling the direct sound through a complex
set of short feed-back and feed-forward delays, configured
to introduce the desired equalisation characteristics.
Digital reverbs
are available to suit all pockets from a large number of manufacturers,
including Yamaha, Sony, Digitech, Klark Teknik, ART, Ensoniq
and Alesis, to name but a few. However, if you ask any professional
sound engineer to name their favourite reverb machine, chances
are you'll hear one name above all others. Lexicon are probably
the most popular manufacturer of digital reverbs, and their
product line extends from push-button preset units to the
most complex state-of-the art systems.
Published
in SOS October 1997. |
