Mixing consoles used for dubbing are often large and intimidating, with exceptional amount of controls. These controls luckily consist of a great deal of duplication among them, so by studying just one area of a console, one learns nearly all of the areas. The mixing console can at the end of the day be broken down into two ingredients, processing and configuration. Sound processes are the devices that is used to manipulate the sound, all the way from simple internal level controls to sophisticated outboard reverberation units whereas the configuration issues consist out the signal routing from the input to the output of the console through the various processes.

Digital Audio Workstations (DAW’s) today have many mixing features which potentially have more processes available to that of a major console. The distinction between DAW’s equipped with a control surface and large consoles is usually that, if the console is digital, it will consist of dedicated digital signal processors for each channel and is thus designed not to “overload” under the burden of signal processing and which may possibly crash or lose signals. DAW’s can dynamically assign resources like digital audio signal processing power and so could run out if many signal processes were in simultaneous use. The upside of this is that this can be solved fairly easily by just plugging in more hardware into the DAW, but the downside would be that the cost of this endeavour may approach that of a major console.

Outboard Gear and Plugins

Outboard gear (analog) and software plugins (digital) can be inserted to the signal path in order to extend processing possibilities. Outboard gear and plugins fall into two main categories:

• Processors – these devices are normally connected in series to the signal path, so the input signal is replaced with the processed signal (e.g., equalizers).
• Effects – while an effect can be considered as any unit that affects the signal, the term is mostly used to describe units that are connected in parallel to the signal path and therefore they add to the existing sounds, but do not replace them. Examples would include reverb and delay.

Multiple Level Controls in Signal Path

A single signal can pass through a large number of level controls – such as an individual channel fader, subgroup master fader, master fader and monitor volume control. Problems are created according to Holman due to the multiplicity of the controls. Each and every console has their own dynamic range – so it is important to utilize this correctly in order to avoid excessive noise or distortions. Attacking this problem – of the correct setting of the variety of controls can be accomplished relatively easily. Holman refers to the scale of the control as a clue for the solution of this problem. With 0 dB being the nominal setting of the controls, many have a “gain in hand,” which goes above 0 dB. This means that one can turn it up from the nominal setting to have something that sounds clear. As for other controls such as sub masters and master level controls are used for slight trims to the overall section by section balance or for the main fade-ins and fade-outs of the overall mix.

Processes that Affect Audio Levels

• Faders – used to attenuate or boost the level of signals.

• Pan pots – A fundamental part of configuration in recording console is panning. Pan pots are devices that place sound among the channels: L, C, R, LS, and RS. They are also used to pan signals to the left or right and in surround, to the back or front.

• Compressors – Every track has a volume range. The moment when tracks are combined in mixing, a problem of unintentional masking of one signal by another arises. For example: we have a dialog recording with a volume range and a music recording with its own volume range. The problem is that although most of the time the music will lie underneath the dialog, at a certain point in time the peaks of the music may correspond to the minimum level of the dialog and the dialog will be obscured.

 
To solve this problem we could decrease the gain level of the music during its higher level passages and increase it during its softer ones to maintain a more even level behind the dialog – but this process will be time consuming. This process can easily be automated by utilizing a device called a compressor – this device is equipped with a number of controls to vary the volume range over which the action of the compression occurs, the amount of the compression, and how fast or slow the compressor acts.

• Expansion – The Expansion device does exactly the opposite of what the compressor does. It increases the volume range of a source and may do so across a wide dynamic range or may be restricted to a narrower region by control functions. Restricting expansion to only low-level sounds helps to minimize noise. This function is often referred to as downward expansion, noise gating, or keying and turns the level down below a threshold set by a specific control. Noise gates have numerous audible problems. If for instance one has a dialog recording with air conditioning noise in it – the threshold of the noise gate can distinguish between the dialog and the air conditioner because the air conditioning noise is lower in level that the dialog. The problem arises with the audible air conditioning noise behind the dialog and the absence thereof in between lines. This exaggerated difference will be much more unnoticeable by being just left unprocessed

• Limitors – A limiter acts on signals above a certain threshold. Above that threshold, the level is controlled so that for each dB of increase on the input, the gain is reduced by the same amount. Therefore the output level above the threshold will stay exactly the same, regardless of any increases in the input level. Limiters can be used to catch occasional events that might not otherwise be controlled, to bring them into a range in which the recording medium can handle the signal linearly.

These items discussed thus far affect the level of audio signal. The most commonly used process is level control, which is used even on the simplest of mixers.

Processes that Effect the Frequency Response

Processes that primarily effect the frequency response of the signal are generally seen as second in importance to level control. These processes clean the audio signal, enhance interchangeability between other signals, adjust for the loudness effect, and generally create a much more pleasant or deliberately worse sound. There are two principle frequency response processes – equalization and filtering.

• Equalizers – The simplest description of EQ is the process of altering the frequency response in a manner similar to what tone controls do on a stereo system. Professional EQ’s dissect the audio spectrum into three or four parts which may be called the low-bass, mid-bass, mid-treble, and high frequency controls.

• Filters – Filters are used to essentially eliminate certain frequencies from the output. Filters strip off the any part of the audio spectrum. There are various types of filters:

 
High-pass filter (low-cut): used to remove excessive room noise at low frequencies. Low-pass filter (high-cut): used to help isolate a low frequency instrument playing in a studio along with others. Band-pass filter: a combination of high- and low-pass filters, also known as a telephone filter – because a sound lacking in high and low frequencies resembles the quality of sound transmitted and received by telephone.

Processes that Affect the Time domain

• Reverbs – Reverbs are used to simulate boundary reflections created in a real room, adding a sense of space and depth to otherwise ‘dry’ recordings. Another use is to help distinguish among auditory objects; all sound having one reverberant character will be categorized together by human hearing in a process called auditory streaming. This is an important feature in layering sound in depth from in front of the speaker to behind it.

 
For example: Before the advent of electronic reverb and echo processing, somewhat more basic, ‘physical’ means were used to generate the effects. An echo chamber, a large reverberant room equipped with a speaker and at least two spaced microphones. Signals was sent to the speaker and the reverb generated in the room was picked up by the two microphones which constituted a “stereo return”.

Downmixing

Downmixing is making a stereo mix from a 5.1 surround mix. It is done in the user’s home theatre receiver. In the down-mixing circuit, the left and right surround channels are blended with the left and right front channels. The centre channel is blended equally with the left and right channels. The LFE channel is either mixed with the front signals or not used. Downmixes made this way seldom create a well-balanced stereo mix, so be sure to check the 5.1 mix for stereo compatibility. Surround monitoring systems should have a downmix button so one can hear how surround mixes will sound when down-mixed to stereo by consumer receivers. It is best to do a separate stereo mix and record it on tracks 7 and 8 of an eight-track mix-down recorder. This stereo mix can be put on DVDAudio discs or Super Audio CDs along with the surround mix.

Consumer electronics may also downmix automatically. For example, a DVD player or sound card may downmix a surround sound signal (four or more channels) to stereophonic sound (two channels) for playback through two speakers.

Mixing in surround

Any device having a number of multiple bus consoles (typically having eight or more buses) can be used to create a surround-sound mix, but the important question of the day is this: How easily can signals be routed, panned and affected in a surround environment to create a 5.1 mix without going nuts with frustration?

Whether you’re working in an analog hardware, digital hardware, or DAW “in-the-box” mixing environment, the ability to pan mono or stereo sources into a surround soundscape, place effects in the 5.1 scape and monitor multiple output formats without difficulty can make the difference between a difficult, compromised mix and one that lifts your spirits.

Mixing in surround is very similar to mixing in stereo except that there are more speakers, placed to “surround” the listener. In addition to the horizontal panoramic options available in stereo, mixing in surround lets the mix engineer pan sources within a much wider and more enveloping environment. In a surround mix, sounds can appear to originate from many more or almost any direction depending on the number of speakers used, their placement and how audio is processed.

There are two common ways to approach mixing in surround:

• Expanded Stereo – With this approach, the mix will still sound very much like an ordinary stereo mix. Most of the sources such as the instruments of a band, the vocals, and so on, will still be panned between the left and right speakers, but lower levels might also be sent to the rear speakers in order to create a wider stereo image, while lead sources such as the main vocal might be sent to the center speaker. Additionally, reverb and delay effects will often be sent to the rear speakers to create a more realistic sense of being in a real acoustic space. In the case of mixing a live recording that was performed in front of an audience, signals recorded by microphones aimed at, or placed among the audience will also often be sent to the rear speakers to make the listener feel as if he or she is actually a part of the audience.
• Complete Surround/All speakers are treated equally – Instead of following the traditional ways of mixing in stereo, this much more liberal approach lets the mix engineer do anything he or she wants. Instruments can appear to originate from anywhere, or even spin around the listener. When done appropriately and with taste, interesting sonic experiences can be achieved, as was the case with James Guthrie’s 5.1 mix of Pink Floyd’s The Dark Side of the Moon, albeit with input from the band. This is a much different mix from the 1970s quadrophonic mix.

 
Naturally, these two approaches can be combined any way the mix engineer sees fit. Recently, a third approach, or method of mixing in surround was developed by surround mix engineer Unne Liljeblad.

• MSS – Multi Stereo Surround – This approach treats the speakers in a surround sound system as a multitude of stereo pairs. For example, a stereo recording of a piano, created using two microphones in an ORTF configuration, might have its left channel sent to the left rear speaker and its right channel sent to the center speaker. The piano might also be sent to a reverb having its left and right outputs sent to the left front speaker and right rear speaker, respectively. Additional elements of the song, such as an acoustic guitar recorded in stereo, might have its left and right channels sent to a different stereo pair such as the left front speaker and the right rear speaker with its reverb returning to yet another stereo pair, the left rear speaker and the center speaker. Thus, multiple clean stereo recordings surround the listener without the smearing comb filtering effects that often occur when the same or similar sources are sent to multiple speakers.

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