The Sound
Effects Tutorial
Sound effects are the icing over the cake; they
can be the difference between a dull and uninteresting sound and a lively, more
enjoyable one. A simple delay can transform a muddled sound, where the
instruments seem to step over each other, into a sound where each the
instrument has its own space, greatly adding to their expressiveness. In this
tutorial we will take a brief look at the sound effects implemented by KarVCD.
Just as a chorus is a group of singers, the
chorus effect can make a single instrument sound like there are actually
several instruments being played. It adds some thickness to the sound, and is
often described as 'lush' or 'rich'.
The
algorithm behind the chorus effect isn't a spectacular or amazing trick - it's
actually fairly simple. What happens when two people play instruments in
unison? Well they are not always playing in precise synchronization, so there
is some delay between the sounds they produce. In addition, the pitch of the
two instruments can deviate somewhat, despite careful tuning. These are the
functions that your chorus effect is reproducing.
Reverberation (reverb for short) is probably
one of the most heavily used effects in music. When you mention reverb to a
musician, many will immediately think of a stomp box, signal processor, or the
reverb knob on their amplifier. But many people don't realize how important
reverberation is, and that we actually hear reverb every day, without any
special processors.
Reverberation
is the result of the many reflections of a sound that occur in a room. From any
sound source, say a speaker of your stereo, there is a direct path that the
sound covers to reach our ears. But that's not the only way the sound can reach
us. Sound waves can also take a slightly longer path by reflecting off a wall
or the ceiling, before arriving at your ears. A reflected sound wave like this
will arrive a little later than the direct sound, since it travels a longer
distance, and is generally a little weaker, as the walls and other surfaces in
the room will absorb some of the sound energy. Of course, these reflected waves
can again bounce off another wall before arriving at your ears, and so on. This
series of delayed and attenuated sound waves is what we call reverb, and this
is what creates the 'spaciousness' of a room.
It's very
tempting to say that reverb a series of echoes, but this isn't quite correct.
'Echo' generally implies a distinct, delayed version of a sound, as you would
hear with a delay more than one or two-tenths of a second. With reverb, each
delayed sound wave arrives in such a short period of time that we do not
perceive each reflection as a copy of the original sound. Even though we can't
discern every reflection, we still hear the effect that the entire series of
reflections has.
The delay is one of the simplest effects out
there, but it is very valuable when used properly. A little delay can bring
life to dull mixes and widen your sound. The delay is the also a building block
for a number of other effects, such as reverb, chorus, and flanging.
Simply put,
a delay takes an audio signal, and plays it back after the delay time. The
delay time can range from several milliseconds to one second. When there is no
feedback involved, the delay only produces a single copy of the input, and thus
is often referred to as an echo device. With feedback, you control the time
that a sound is recycled on the delay unit and therefore the number of echoes
produced.
Delays are
very useful for filling out an instrument's sound. Using the delay unit with a
short echo, say 50 to 100 milliseconds, creates a doubling effect, as though
two instruments were being played in unison.
The Delay
unit is also very important when building a mix of instruments in a stereo
environment. It can enhance stereo placement of instruments, and making the mix
sound 'bigger'. A little delay can be more effective than panning for spreading
tracks out in the stereo field. Just a simple delay on the order of 20
milliseconds can make a big difference.
As the name implies, compression reduces the
dynamic range of a signal. It is used extensively in audio recording,
production work, noise reduction, and live performance applications, but it
does need to be used with care. It's commonly said that compressors make loud
sounds quieter, and the quiet sounds louder, but this is actually only half
correct.
A
compressor is basically a variable gain device, where the amount of gain used
depends on the level of the input. In this case, the gain will be reduced when
the signal level is high which makes louder passages softer, reducing the
dynamic range.
KarVCD has
two compressors, one dynamic and one static. They perform two very different
kinds of sound processing.
The static
compressor works on instantaneous values of the output: As soon as the output
value becomes greater than a given threshold, the output is clipped is a soft
manner, so as not to cause audible distortion. This compressor is controlled by
the Compressor level slider control. This sets the threshold that determines
the point where the clipping starts to take place. When this slider is set to
the minimum (pushed full to the left) the static compressor becomes inactive.
The dynamic
compressor looks ahead in the output buffer and when it finds that a sound peak
is about to happen, it starts to reduce the gain of the output stage, in a
gradual manner, so that when the peak is reached, the gain is set so as not to
cause distortion. After the peak is over, the gain is gradually set back to the
normal value.
The dynamic
compressor is always active and is controlled indirectly by the Master Volume
slider. This means that to increase the effect of the dynamic compressor you increase
the Master Volume so that the sound peaks will be higher and thus the effect of
the compressor more pronounced. Conversely, if the compressor is generating
unpleasant sound dynamics, you should reduce the Master Volume.
The amount
of time that the dynamic compressor looks ahead is about 3 seconds. This is
also the time it takes to recover back to normal gain after a peak.
Flanging has a very characteristic sound that
many people refer to as a "whooshing" sound, or a sound similar to
the sound of a jet plane flying overhead.
Flanging is
created by mixing a signal with a slightly delayed copy of itself, where the
length of the delay is constantly changing. This isn't difficult to produce
with standard audio equipment, and it is believed that flanging was actually
"discovered" by accident. Legend says it originated while the Beatles
were producing an album. A tape machine was being used for a delay and someone
touched the rim of a tape reel, changing the pitch. With some more tinkering
and mixing of signals, that characteristic flanging sound was created. The rim
of the reel is also known as the 'flange', hence the name 'flanging'.
In KarVCD,
the flanger effect is provided by the Delay unit. For this, you set the delay
time between 1 and 10 ms and the modulation value between 2 and 6 ms.
Optionally you can set a small amount of feedback. A large amount of feedback
can create a very 'metallic' and 'intense' sound.
The phase shifter (or phaser) achieves its
distinctive sound by creating one or more notches in the frequency domain that
eliminate sounds at the notch frequencies (the flanger also makes use of
notches, and it is actually one specific type of phasing). The notches are
created by simply filtering the signal, and mixing the filter output with the
input signal.
In KarVCD,
the phaser effect is provided by the Delay unit. For this, you set the delay
time to 3 ms and the modulation to 4 ms. If you set a small amount of feedback
the phasing effect will be more intense.
Comments, suggestions and bug reports are welcome and should be sent to fadevelop@clix.pt
This page last modified 2003-07-29 - Copyright
© 2000-2003 ACE