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 firstname.lastname@example.org
This page last modified 2003-07-29 - Copyright © 2000-2003 ACE