The desk has two main sections: Inputs and Outputs. Although most sound desks look terrifying initially, they can be mentally broken down into simple sections. Once you understand the basic concepts, you'll see that all mixing desks are very similar.

Inputs are where the sound signals come into the desk, and outputs are obviously where the sound leaves the desk.

Any input signal (in the case above, there are 8 possible inputs) can be routed to any of the four outputs.

The inputs sent to each output are mixed together, and once the output channel is raised, the signal is sent to the output connector.

The machine that actually records musicians' performances during a session. The multi-track recorder is kept in the control room and operated by the engineer(s). Often, the process of recording onto a multi-track recorder is called tracking or laying tracks.

Each track can be used to record a different instrument or vocal performance. Because the tracks are synchronised, performances that have already been recorded may be played back and listened to while additional performances are recorded.

Consequently, a "perfect" recording can be achieved without requiring all participating musicians to perform perfectly together. If one musician makes a mistake, he can listen to the recording of the others while re-performing his own part. The most common multi-track recording method is using the computer program 'ProTools' through a Macintosh computer.

This machine is similar to the multi-track recorder except instead of many mono tracks, its just one big 2 channel track; Left and right. Not used very often on its own, but in studio stereo channels are activated with microphones such as overheads; two overhead speakers go into the same stereo channel.

The devices collectively known as signal processors were originally added to the recording studio to allow compensation for frequency response or dynamic range problems in the equipment. When carefully used, they can add to the fidelity of the sound, and occasionally improve on reality. If improperly used, they can seriously degrade the sound or even produce laughable results. It is possible to use signal processors to manipulate sounds, so naturally they are popular with composers of electronic music.

An equalizer is a device that can alter the spectral content of a signal. This can be done with any circuit that has an adjustable frequency response, the most familiar being the tone controls on a home stereo set. These tone controls typically affect the amplitude in two frequency regions, the treble and bass. This is sufficient for the minor changes the end user may wish to make in the program, but the recording engineer needs more flexibility and coverage of the entire audio spectrum. The complete studio will have complex e.q. systems that might include 30 or more regions of control. Many of these machines have sliders to adjust the amplitude of each band, and those sliders are laid out in such a way that their positions visually indicate the frequency response. This feature gives rise to the name GRAPHIC EQUALIZER.

A 30 band equalizer is going to be an expensive device, simply because of the sheer duplication of circuitry. Much of this circuitry is wasted most of the time because typical use only involves adjustment to two or three bands while the others are left alone. An equalizer with bands of adjustable frequency is a more economical approach to the problem, because only three to five circuits are necessary. Such an equalizer is called a PARAMETRIC EQUALIZER. Parametric equalizers typically have three controls for each band, allowing adjustment of frequency, amplitude, and width of the band.

It is also possible for the soft spots of a program to become too low in amplitude, especially in a noisy medium like AM radio. A program with a wide dynamic range may have to have that dynamic range reduced in order to fit a particular audio system. (Surprisingly, the current switch to digital audio is making this problem more common as stations try to fit the wide range of compact discs through the restricted dynamic window of FM transmitters.) A device which reduces the overall dynamic range of a signal is called a COMPRESSOR.

A compressor works by measuring the average amplitude of the input signal and using the information from that measurement to control the gain of an amplifier. If the measurement comes out above an adjustable threshold, the gain is reduced, if the measured input amplitude is below the threshold, the gain is increased. The amount of effect is usually adjustable and is expressed as a ratio of input dynamic range to output dynamic range, such as 2 to 1 or 3 to 1.

The release time is adjustable just as with the limiter, but the adjustment is much more critical, because the compressor is constantly changing gain whereas the limiter only cuts in occasionally. It is impossible to find a release time that is perfect for all situations. A very short release time will cause the circuit to attempt to trace the waveform of low frequency components of the signal, resulting in inter-modulation distortion of high frequency components; whereas a long release time will leave the amplifier in the wrong mode if the signal changes quickly. That second error results in brief bursts of expansion, a very disconcerting effect. (Sounds like heavy breathing.)

Sometimes a program is encountered which has proper dynamic range but which has objectionable background noise. Very little can be done to reduce the noise without affecting the signal, but the spaces where there is no signal can often be cleaned up with a NOISE GATE. This circuit is an expander that only works on the low amplitude parts of the signal. Any section that is below the threshold will be attenuated rather drastically, so the space becomes nice and quiet. You can hear this device in action on old movies on TV. Release time adjustment is very critical in this application also, because if the circuit is too slow, the turning down of the noise will be painfully obvious, but if the circuit works too fast, the reverb at the ends of the sounds will be lost.

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 sounds 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.

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, widen your instrument's sound, and even allow you to solo over yourself. 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 several seconds. Figure 1 presents the basic delay in a flow-graph form. This only produces a single copy of the input, and thus is often referred to as an echo device.

Plugging the guitar directly into the mic or line level input of a mixer is unsatisfactory for several reasons:

Similar problems are encountered with other instruments such as synthesisers, electric pianos, and pickup systems for acoustic instruments.

To connect such an instrument with the mixer, a special interfacing unit known as a DI box (DI=direct injection) is therefore employed. This unit will convert the instrument's output to a low-impedance balanced signal, and also reduce its output level to the milli-volt range suitable for feeding a microphone input. In addition to the input jack socket, it will also have an output jack socket so that the instrument's unprocessed signal can be passed to an amplifier as well. The low-impedance balanced output appears on a standard three-pin XLR panel-mounted plug which can now be looked upon as the output of a microphone.

An earth-lift switch is also provided which isolates the earth of the input and output jack sockets from the XLR output, to trap earth loop problems.

The simplest DI boxes contain just a transformer, and are termed 'passive' because they require no power supply. The transformer in this case has a 20:1 step-down ratio, converting the fairly high output of the instrument to a lower output suitable for feeding microphone lines. Impedance is converted according to the square of the turns ratio (400:1), so a typical guitar output impedance of 15 kilo ohms will be stepped down to about 40 ohms which is comfortably low enough to drive long microphone lines. But the guitar itself likes to look into a high impedance.

The active DI box replaces the transformer with an electronic circuit which presents a constant very high impedance to the instrument and provides a constant low- impedance output. The box is powered either by internal batteries, or preferably by the phantom power on the microphone line.

A headphone distribution amp however is intended to drive a number of headphones, usually in a some sort of a professional situation such as a recording studio or a language lab etc., the emphasis in the design of a distribution amp is usually more on cost per channel that on sound quality although there are some excellent quality distribution amps out there designed for use in recording studios