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Also known as MB or MBC.

These divide the incoming audio signal into multiple bands, with each band being compressed independently from the other.

The beauty of this is that with full band compressors the whole signal is treated, so when a peak is detected, the whole signal is compressed and so other frequencies are also subjected to compression.

Multiband compression only compresses the frequency bands chosen, so a more fluid and less abrupt result is gained. Instead of having one peak trigger the compressor into compressing the entire signal, the multiband allows for individual bands to be compressed. On some compressors, you even have the option of selecting bands that will not undergo any treatment. In essence, a multi-band compressor comprises of a set of filters that splits the audio signal into two or more frequency bands. After passing through the filters, each frequency band is fed into its own compressor, after which the signals are recombined at the output.

The main advantage of multi-band compression is that a loud event in one frequency band won’t trigger gain reduction in the other bands.

Another feature of the multiband compressor is that you are offered crossover points. This is crucial, as you are given control over where to place the frequency band. Setting these crossover points is the heart of the compressor and crucial in processing the right frequency spectrum with the right settings. For example: if you are treating the vocals in the mid range but put your low end crossover too far into the middle range, then the low end compression settings will also affect the mid range vocals.

Multiband  compression can either be a friend or enemy. It all comes down to how you use it and when. It can be a great compressor for controlling problematic frequencies, or for boosting certain ranges in isolation to others. I tend to use them to rescue poor stereo mixes and with the aid of new features like crossover frequencies and threshold and ratios for each band I can have more accurate processing.

However, use with care.

This is the level difference between the signal level and noise floor. The best way to describe this is by using an example that always works for me. Imagine you are singing with just a drummer. You are the signal and the drummer is the noise (ha.ha). The louder you sing or the quieter the drummer plays the greater the signal to noise ratio. This is actually very important in all areas of sound technology and music. It is also very relevant when we talk about bit depth and dynamic range.

Imagine using 24 bits. That would allow a dynamic range of 144dB (generally, 6 dB is allocated for each Bit).
Bearing in mind we have a limit of 120dB hearing range (theoretical) then the audio signal would be so much greater than the noise floor that it would be almost noiseless.

People still find it confusing to distinguish between signal-to-noise ratio and dynamic range, particularly when dealing with the digital domain.

Signal-to-noise ratio is the RMS (Root Mean Square) level of the noise with no signal applied (expressed in dB below maximum level). Dynamic range is defined as the ratio of the loudest signal to that of the quietest signal in a digital system (again expressed in decibels (dB)).

In a typical professional analogue system, the noise floor will be at about -100dBu. The nominal level is +4dBu, and clipping is typically at about +24dBu. That basically equates to about 20dB of headroom and a total dynamic range of about 120+dB. Clipping in an analogue system equates (when used in small stages) to harmonic distortion. This is why ‘driving’ the headroom ceiling would sometimes make the audio sound more pleasing.

Digital systems operate in finite and critical terms, and ‘driving’ the ceiling cannot be done. As digital works off a linear system, once the quantising scale is reached clipping takes place (enharmonic distortion).

Luckily, converter technology has improved so much that we now have 24 bit delta-sigma converters offering 120dB of dynamic range, similar to what we had/have in analogue consoles. And by using the same methodology, by leaving ample headroom, we are able to have great dynamic range and a strong S/N offering a negligible noise floor.

In practice, this equates to the following:
Working with a nominal level of -18dBFS (EBU) or -20dBFS (SMPTE/AES), we can attain approximately 20dB of headroom whilst keeping the noise floor about -100dB.

Digital systems cannot record audio of greater amplitude than the maximum quantising level (please read my tutorial on the Digital Process). The digital signal reference point as at the top of the digital meter scale is 0dBFS, FS standing for ‘full scale’.

In the US, the adopted standard of setting the nominal analogue level is; 0dBu equals -20dBFS, thereby tolerating peaks of up to 20dBu. In Europe, 0dBu equals to -18dBFS, thereby tolerating peaks of up to +18dBu.

This all sounds complicated but all you really need to be concerned with, as far as the digital world is concerned, is that we have a peak meter scale of 0dBFS. Beyond this and you have clipping and distortion.