Signal processing devices can be technically defined as electronic devices that alter the audio signal in a non-linear fashion. Given that definition, a fader, gain control, or amplifier is not a signal processor. Equalisers, effects processors, phase shifters, compressors, expanders, and noise reduction systems are all examples of signal processors; they all modify the input signal in some fashion. Equalisers modify sections of the frequency spectrum by boosting or attenuating (reducing) certain frequency bands; effects processors are capable of introducing time delay, artificial reverberation through the use of strange phase-related algorithms, or a combination of both; phase shifters modify the phase characteristics of an input signal and add it back to the original, providing for "hollow" sci-fi effects; compressors essentially keep loud input signals from becoming too loud; expanders make soft signals louder; noise reduction units use a combination of compression and expansion to provide a lower noise floor; the list goes on and on and on. Theatrical sound designers generally use processing such as equalisation and delay to properly align and "tune" their individual loudspeaker systems within a given acoustic space, and use other effects, such as reverberation and compression in more specific quantities on particular inputs.
(Here is the detailed version for those interested. for a basic overview of the various outboard effects watch the videos)
Equalisers are probably the most common example of a basic signal processor-- they are found on the input channel of most mixing desks and are also found as standalone units to modify the frequency response of a given input signal. Historically, equalisers were used by the telecommunications industry to make up for frequency-dependent signal loss when transmitting voice over long distances of cable. Special components were used to boost certain frequency bands that were attenuated over the long cables, thus making all frequencies equal in level (amplitude)-- hence, "equaliser."
Equalisers are used extensively in sound recording and reinforcement. They are designed to be used to provide control of the frequency response of a specific portion of the sound system. Most mixing desk input channels provide at least a very basic equalisation section, which may be as simple as a set of high and low filters (like your car stereo), or may be a four-band parametric equaliser which allows very precise control of the input signal. Equalisers are used further down the sound system chain as outboard processors which can affect the frequency response of groups of inputs, or even farther down the chain as system equalisation designed to affect the frequency response of a given set of loudspeakers.
BASS-ICS AND TREBLE-ICS
The very basic form of equalisers, usually found as the "bass" and "treble" knobs on your car stereo, are simply two filter networks. The "bass" knob is a filter circuit centred around a particular frequency-- usually around 100 Hz, and has a given amount of attenuation and gain (+/- 10dB, usually). Turning up the "bass" knob will increase the amount of low-frequency information relative to the rest of the program material and will result in a richer sound, and in the extreme, boominess. When you turn down the bass knob, the amount of low-frequency information relative to the program material is reduced, resulting in a honky or thin sound. The tone controls are usually of the shelving type, which means that frequencies above the treble centre frequency (or frequencies below the bass centre frequency) will also be affected by the filter. If one were to plot a graph of a bass tone control turned fully counter clockwise (cut), the graph would show that the low-frequency response would slope downward such that at 100 Hz, there was 10dB of attenuation. Below 100 Hz, there would continue to be 10dB of attenuation. The point at which the frequency response starts to be affected by the equaliser circuit is called the hinge point. For instance, to provide for a 10dB reduction in amplitude at 100 Hz, the frequency response starts its downward slope at a higher frequency, say 500 Hz, continues to slope downward until it reaches 100 Hz, and then levels out again below 100 Hz. The treble control operates in exactly the same fashion, but mirrored. If the treble control is turned fully clockwise (boosted) and has a centre frequency of 10 kHz, the frequency response will start being affected at its hinge point, probably somewhere around 3 kHz, and the amplitude will increase until it reaches 10 kHz, at which point there is 10dB of gain. Frequencies above 10 kHz will all have 10dB of gain.
The tone-control setup does not provide for much flexibility at all, what with a fixed centre frequency and a shelving curve. What happens, for instance, when an offensive frequency falls between the two hinge points of the tone controls? The tone controls will not affect the band between the two hinge points, so what good is it? Mixing desk manufacturers thus provide somewhat more elaborate controls which allow the operator to choose the centre frequency, the amount of boost/cut, and even more important quantity- the bandwidth.
In our tone-control example, frequencies above the "treble" centre frequency are summarily affected in the same way- all frequencies above the 10 kHz point are boosted or cut. But this shelving arrangement isn't very practical, even if we're allowed to choose the centre frequency. What happens if we want to centre our equaliser at 500 Hz with a shelving equaliser? Either everything below 500 Hz or everything above 500 Hz will be similarly affected by our filters, which isn't a very helpful arrangement. Opposite to the shelving type of equaliser is the notch or peaking equaliser, which allows the operator to centre around a particular frequency and leave frequencies outside of a given bandwidth largely unaffected-- thus, we can centre on a 500 Hz frequency, apply attenuation or gain, and leave frequencies beyond, say, 600 Hz and below 400 Hz unaffected. If were to plot the frequency response of a 10dB boost at 500 Hz, we'd see that low frequencies remained at 0dB, started to slope upwards somewhere around 400 Hz, reach a 10dB boost at 500 Hz, then slope downwards around 600 Hz, leaving higher frequencies at 0dB. This is a very common type of equaliser; when multiple notch filters are combined into one unit, each affecting a different portion of the frequency spectrum, we call this a graphic equaliser.
A graphic equaliser is a multi-frequency bandpass/reject filter that functions as a standalone unit, usually used to equalise specific loudspeaker systems. While mixing desk input channel equalisers operate on three or four bands, a graphic equaliser can simultaneously operate on at least eight different frequency bands. The most common graphic equalisers in sound reinforcement align their centre frequencies according to ISO standards which fall on one-third octave centres, and can be covered in thirty-one bands. There are more expensive and cumbersome one-sixth octave and one-twelfth octave devices, but these units are rarely used in sound reinforcement. It is generally accepted that one-third octave graphic equalisers are sufficient for most venue tuning and feedback reduction.
Further developments in equalisers provided a bandwidth control in addition to the boost/cut and centre-frequency selections. Adjusting the bandwidth controls the range of frequencies affected by a given filter. In graphic equalisers, the bandwidth of each frequency band is pre set by the manufacturer and cannot be changed. This configuration does not provide the flexibility that some designers and operators require-- in certain circumstances, one may wish to have a broad EQ curve, which begins gently and very gradually builds up to the maximum peak with regard to frequency. On the other hand, one may wish to have a very specific frequency cut or boosted without adversely affecting adjacent frequencies. The broadness or sharpness of the EQ curve is specified by a quantity called Q, or quality factor. Quality factor is the measure of the sharpness of a resonant peak, and many people often use Q interchangeably with bandwidth, which is not entirely correct. The quality factor of a filter can be said to actually determine the bandwidth of the filter; the higher the Q, the sharper the curve, and thus, the narrower the bandwidth.
Equalisers that provide control over Q, centre frequency, and boost/cut, are known as parametric equalisers, so named because they allow control over all necessary parameters of equalisation. Parametric equalisers are found on input sections of mixing desks and also as standalone units. Input channels can offer anywhere from one to three different bands of parametric equalisation, which allows for precise manipulation of the input signal. Outboard units usually come in five-band versions; each band is set for a different frequency band, although there are overlaps. There are three potentiometers per band; one selects the frequency, the other selects the bandwidth, which is usually measured in octaves (.1 is a very tight bandwidth, and 1.1 is a wide bandwidth), and the last selects the amount of boost/cut produced by the filter. Further information on equalisation and sound waves can be found in these two videos.
TO EQ, OR NOT TO EQ
Any equalisation creates phase shift in the audio signal, which can lead to distortion, reduced headroom, and comb filtering in multiple loudspeaker systems, if allowed to run rampant. Equalisation; should kept to a minimum, and errant frequency response should be corrected by proper loudspeaker positioning or acoustical treatment of the room. Most designers never use system equalisers for frequency boost; boosting particular frequency bands can result in reduced amplifier headroom, distortion, and more potential for feedback. Most designers prefer instead to attenuate only, and use equaliser gain only in very specific instances, such as at the input channel.
Equalisation can reduce the resonant peaks and dips in loudspeaker systems as they interact in a given acoustic environment, which can be used to reduce the possibility of acoustic feedback. As the overall gain of the sound system is turned up, feedback occurs first at that frequency where the system of microphone + system + loudspeaker + room has a peak. It is heard as a slight ringing and can become a loud howl. If the offending resonant frequency appears due to the loudspeaker + room interaction, an equaliser set to attenuate that first peak can be used on that particular loudspeaker system in order to increase the available gain before feedback. Subsequent frequency peaks can be eliminated in the same way. Graphic equalisers provide a quick and dirty method of equalising a system within a room, while parametric equalisers provide more control over the attenuation of frequencies.
Regardless of the type of equaliser used, they are a very powerful tool that should be used carefully. Inputs on a mixing desk may require some sort of equalisation, be it radical frequency-specific tuning, or just a low-cut roll off filter to reduce boomy microphone noise. Most system designs require an equaliser on each signal feeding a loudspeaker system, which is usually installed just prior to the amplifiers or electronic crossover systems. In recording and broadcast situations, the equalisation that is required is often radically different from that required in a reinforcement system; instead of equalising frequency response peaks and dips in loudspeaker systems as they interact with the room and other loudspeaker systems, equalisation may be needed to alter the frequency response to accommodate requirements for a particular recording medium. Use it wisely. It's a good thing.
Some frequencies that might be useful are found in this PDF
Other Signal processors
There are many other types of processor. here are a few of them.
One of the first things you'll notice when sound engineering is that people tend to sound really bad when singing through a microphone. Its not just bad singing though, even great vocalists will sound bad when you compare them to a recorded track, with the sound appearing uneven and varying in volume. Whilst good microphone technique will reduce this, a major factor in making recording sound good is the use of a compressor. A compressor essentially reduces the volume of loud signals, 'squashing' the signal so that the loud and quiet bits sound more even.
Imagine that you are listening to someone singing through a microphone and that when you hear them getting above a certain volume you start to turn the volume control down. The louder the singer becomes, the more you turn the volume down. By doing this, you will be able to stop the singer from becoming too loud and make the volume more even.
The effects of threshold and the compression ratio on the output signal of a compressor.
There are two main controls that all compressors possess:
Threshold - The point at which a compressor will start to take effect. Signals below this threshold will be unaffected but ones above it will be reduced. In our analogy, it corresponds to the volume at which you would start to move the volume control.
Compression ratio - The amount the signal is compressed by expressed as a ratio of the output and original gains (above that of the threshold). A 1:1 gain will have no effect on the sound but as the ratio increases past 3:1 and up to 10:1, the volume differences are getting reduced and the sound is getting squashed. When the ratio reaches infinity:1, the signal can no longer get above the threshold level and the compressor has become a limiter.
Other controls which are usually included:
Attack time - When the signal reaches the threshold point, the compressor can wait for a certain amount of time before it reduces the signal. This is the attack time and is usually set very short (0.05s) when dealing with very quick peaks caused by handclaps and drums etc.
Release time - As the signal falls below the threshold point, the compressor can be set to wait for a period of time before it stops reducing the output signal. This is the release time and is usually in the range of about 0.05 to 5s.
Uses of a compressor
Compressors can be used on a wide variety of instruments but are most widely used on vocals. They can compensate for a singer who tends to sing at a wide range of volumes and and distances from the microphone. Since the loudest sounds from the microphone has been reduced, the overall gain on the channel can be increased, meaning that quiet sounds can be heard more clearly. Another use for a compressor is that by setting the threshold high and the ratio to infinity any extremely loud signals can be stopped before they are sent downstream to the amplifiers where damage can be done to the speakers.
Noise gates and expanders
Noise Gates and their older brothers, expanders, work on the opposite end of dynamic range.
If you actually listen to an instrument like a drum you'll find that its significantly different to what you hear on your CD at home. You might feel that the kick drum hasn't enough 'kick' or that the snare drum hasn't enough 'snap' and the reason is that during mixing, the signal will have been fed through a noise gate before being used. Essentially this piece of equipment only lets through loud bits of sound and cuts off quiet parts.
Imagine that you have a door with a spring on it. Hit the door softly and the door won't open but hit it hard and you'll be able to open it, letting you through. A very similar thing happens inside a gate where a signal above a certain strength (the threshold) will be able to pass through.
The effects of threshold, attack and release on the output signal of a gate.
There are three main controls that all gates possess:
Threshold - The point at which a gate will let the sound pass. Any signal under this strength will not be let through and so if the threshold is set too high, it is possible for there to be complete silence.
Attack - The time between the signal reaching the threshold srength and the gate actually opening to let sound through. Don't confuse this with delay, the signal isn't paused, the gate just waits before acting.
Release - The time between the signal falling below the threshold and the gate actually closing to prevent sound passing through. A higher release time usually produces a softer sound since you can hear the drum vibrating freely.
Traffic light display
Many gates use a series of three LEDs to indicate what it is currently doing to the signal:
Status of the gate
Input signal is below the threshold point. The gate is reducing the output level and so nothing will be coming out from the speakers
Input signal falling below the threshold level or rising above it. The gate is using its hold or release circuits.
Input signal is above the threshold level. There is now sound coming from the speakers.
Uses of a gate
There are two main uses of a noise gate:
Change the sound of an instrument such as a drum to give it more impact. By only letting the loudest part of a drum strike through into the mix, the drum will sound quicker and stronger.
Reduce the noise. People talking, interference and other quiet and unwanted sounds can be removed from the mix by using a gate. By setting the threshold so that the gate only lets intended sounds through, there should be silence at all other times. Remember that it is best to remove noise at the source rather than use a gate and that a gate can't remove interference down-stream of it.
Noise gates are often used for automatic microphone mixing situations. Dan Dugan, a pioneer in the field of automatic mixers, developed a system in which microphones were automatically muted if no primary signal was detected, which reduced the amount of background noise in the system and also increased the available gain-before-feedback in a reinforcement system. In recording, noise gates are often found on individual drum inputs or electric guitars to provide a clean, undisturbed signal.
Expanders find their homes in most noise-reduction and broadcast systems. Upon recording, the noise reduction system compresses the audio signal in order to fit the original dynamic range into a smaller dynamic range for recording. On playback, the noise reduction system decompresses (expands) the compressed signal, which restores the original dynamic range of the recording and "pushes down" any inherent tape hiss or noise below the program noise floor.. Systems that integrate compressors and expanders for this purpose are often referred to as companders.
Other effects (FX)
Reverberations the natural phenomenon that occurs when sound pressure waves are reflected and reinforced and blended with the original, direct sound. All concert halls and theatres exhibit different reverberation characteristics. The size and shape of the venue coupled with any obstacles or sound-absorbing material all affect the way in which the sound waves interact. Concert halls designed for classical music often have somewhere in the neighbourhood of 2.50 seconds of natural reverberation time; churches may have as much as 5.00 seconds. "Dead" theatres may only have about 1.00 seconds of reverb time.
Reverb processors are electronic devices which emulate this effect. They have found a home on the reinforcement side to give more "fullness" to program material or to be utilised as special effects, giving the listener the illusion of distance or space. Reverberation is often confused with echo and delay, and while both of these effects comprise reverberation, they are not the same thing.
Most modern reverb processors can do more than just supply digital reverb-- many incorporate other digital effects such as chorus (which replicates the original sound and frequency-shifts the copies), or pitch-shifting, or distortion, - the list goes on. Special effects can be operated and processed in real time using such devices, and, used judiciously, can provide a variety of acoustic environments by which the audience's perception can be altered.
Delay units do precisely that-- take an input signal and delay the output by a user-defined amount of time. The recording industry uses delay units as special effects; in sound reinforcement, we use digital delay devices to delay the audio signal feeding a given loudspeaker system such that sound waves from two spaced loudspeaker systems will arrive at the same location at the same time-- but we'll cover that in another section.
Phasers and flangers produce the same-sounding effect via different means. The final effect of a phaser or flanger is a hollow-sounding sound caused by phase differences between copies of the same input signal.
Flangers originated from the use of two tape playback machines playing the same signal. By mixing the two outputs and alternately slowing down one machine, then the other, different phase cancellations occurred. The slowing down of the machines was acheived by applying slight hand pressure to the flanges of the tape supply reels hence the name. Modern flangers use electronics to simulate this effect.
Phasers, or phase-shifters, use one or more deep, high-Q filters. The input signal is split two ways: one bypasses the filter, and one flows through the filter circuit. Because of the capacitative nature of all analog filters, phase shift occurs when a signal is passed through a filter on either side of the notch frequency. By varying the notch frequency through the audio frequency spectrum and mixing the resultant signal back with the original, a series of phase-cancellations results.
Exciters are units designed to add "punch" and/or "air" to the overall program material without changing system gain. The exciter circuits often use a set of filters and equalisation coupled with harmonic manipulation to produce a "phatter," more intelligible sound. Exciters are sometimes used in the penultimate mixdown process of popular music and are often used in dance-club reinforcement systems. Only rarely are they used in theatrical sound design.
Be aware that any frequency-manipulating device will introduce phase-shift into the audio circuit, which may have detrimental results on the frequency response of loudspeaker systems in the venue! On the other hand, frequency-manipulating devices and other effects processors can aid the designer in creating an aural soundscape that transports listeners into another environment. Let your ears be the judge!