FabFilter
Hi, and welcome to this beginner’s guide to bus processing.
Let’s start by defining the term: a “Bus” simply adds multiple signals together.
The most important bus in your project is the master bus, otherwise known as the mix
bus, where all your individual channels are summed together to create one stereo signal.
If you add processing to this master channel, you’re processing the entire mix.
We also use buses when we create send effects.
If I send a few different channels to a reverb effect… these channels all get mixed together,
before being processed by an all-wet reverb effect… and the resulting reverb-only signal
can then be blended with the dry signals.
Technically this is bus processing, as the signals are added together at a bus, and then
processed by the reverb.
But this kind of effect send setup isn’t what we’re usually referring to when we
use the term bus processing, as in this case the dry signal still just goes directly to
the mix bus, and if I turn down the reverb channel we only lose the reverb effect.
The term “bus processing” more usually refers to the use of subgroups.
In this case, the dry signals don’t go directly to the mix bus: first, they get mixed together
with a group of other related channels, and then that subgroup is routed to the mix bus.
The method you use to achieve this will vary depending on the DAW you’re using: in Reaper
I can create a folder track, in this case containing all my drum channels, and the folder
track will automatically become a subgroup.
Other DAWs might use a more traditional hardware mixer paradigm, such as Harrison Mixbus: in
this case I need to change the output routing for my drum channels, so they all go to the
drum bus instead of direct to mix.
This allows us to adjust the level of the whole drum kit using just a single fader…
but also allows us to process the whole drum kit by loading plugins onto the subgroup channel.
Lets consider a few of the most common options.
If I load an EQ onto the subgroup I can EQ all the drums together.
These classic drum machine samples have bags of character, but they all benefit from a
gentle cut in the lower mids to reducing the boxy character… and another to stop this
lower region from conflicting with the bass part… and as the kick drum is the only element with
significant content in the low sub-bass, I can shape the low end of the kick in any way
I like without really affecting anything else.
In this case, EQing the group gives exactly the same results as if I EQed every channel
individually using the same settings…
But using just a single instance on the group is much more efficient.
A single EQ instance uses fewer of your processor cycles than would multiple instances, but
this is not so important as any modern processor can handle many instances of Pro-Q3 without
breaking a sweat.
More importantly, it allows you to work more quickly, which makes you more productive,
and probably leads to better results: the longer you spend listening to a mix in progress
the more you lose your perspective, and the harder it becomes to gauge how your mix will
sound to a fresh pair of ears.
This approach doesn’t preclude also EQing individual channels.
For example, I might try crisping up the sound with a boost around 8KHz… but then cut back
that same region for the hi-hat channel to stop it getting too bright… and this is
still more efficient than loading and adjusting a separate EQ instance for every drum channel.
Starting with broad brush EQ changes applied to subgroups can be a very effective way to
get a good basic mix together quickly, which can then be refined with tweaks to individual
channels if needed, and this approach is often referred to as “top downmixing”.
The reason a single EQ on a subgroup can be equivalent to EQing each individual channel
is because a clean EQ such as Pro-Q3 is totally linear.
In DSP terms this means it behaves the same way regardless of the incoming signal level,
and hence EQing each channel before adding them together is the same as adding them together
first, then EQing the result.
This ceases to be true as soon as you start to use Pro-Q3’s dynamic features, however.
Now we have a nonlinearity because this dynamic band is reacting to the input signal
and compressing the dynamic range for that region.
If I run multiple instances of this setting on individual channels, each instance would
see a different input signal, resulting in different dynamic behavior.
Ok now let’s try compression.
Here’s how it sounds if I compress each element individually… the kick and snare
now sound punchier and more focussed… and we’ve lost most of the dynamics in the hi
hat part.
Now let’s try compressing them all on the subgroup instead: I can achieve a similar extra attack
and focus for the kick and snare parts… but the hi-hats aren’t hitting the threshold
at all with this setting, so we’re not losing the accents of the original part.
But because the compressor gain reduction is applied to the mixed subgroup, the gain
reduction applied to kick and snare hits now also affects the hi-hat.
Changing the release time doesn’t only change the decay of the kick and snare hits: it now
also changes the way the hi-hats ride up in level in the pauses between kick and snare
hits, and can, therefore, affect the groove of the drums.
Compression is nonlinear by definition: it reacts to, and shapes the dynamics of the
incoming signal.
So unlike EQ, compressing a subgroup is never the same as compressing the individual elements
separately.
Here’s a subgroup with all my guitar and synth parts: when I compress them all together
they start to interact, with loud events from one part ducking all the other parts in level,
and this interaction between the parts is often called “glue”.
The gluing effect is strongest when you compress the entire mix on the mix bus, but gluing
together just a subgroup can also be a very useful approach.
Watch my Beginner’s Guide to Compression for a more detailed look at this effect.
Saturation is also a non-linear process: turning up the input level results in more saturation,
and more added harmonics, while turning it down makes the sound cleaner.
And like compression, saturation will behave differently when applied to a whole subgroup
instead of each individual part.
Each part will gain extra harmonics just as it would if processed alone, but as well as
this we’ll also get intermodulation, as each part changes the way all the other parts
hit the non-linear distortion stage.
Intermodulation quickly starts to sound nasty if there’s too much of it, so you’ll usually
need to keep the saturation pretty subtle when processing subgroups or a full mix.
But in small amounts it adds its own subtle kind of gluing effect, helping the parts
to gel together better.
For my money this is the key to the elusive analog sound that’s so sought after: small
amounts of saturation on buses can help to create that expansive, warm yet clear sound
characteristic of many classic analog recordings… while too much of it can recreate the muddy,
cluttered, or abrasive character of many forgotten bad recordings from the same era.
Check out my Beginner’s Guide to Distortion and Beginner’s Guide to Saturation videos
if you want to know more about intermodulation.
Putting them together.
In a real-world mix we’re likely to be using EQ and compression and saturation, which complicates
things a little.
If the EQ on your subgroup comes before any non-linearities such as compression or saturation,
then it will still be equivalent to the same EQ setting applied to all the individual channels
in that group, so long as those individual channels EQs all come after any compressors
or saturators on those channels.
But if the individual channel EQs are running before the non-linearities… or if the subgroup
EQ is running after them… the two will no longer be equivalent, as EQ before a non-linearity
isn’t the same as EQ after a non-linearity.
So what about more creative effects?
Obviously a filter sweep is more dramatic in effect when applied to the whole mix
instead of just a single element…
But equally, you could apply creative effects to a smaller subset of parts, like bass, keys
and guitar for example.
As complex as this filter setup seems, it is in fact almost entirely linear: the LFO
and step sequencer modulation doesn’t care about input levels, and would still behave
the same way if I run multiple instances on the individual channel instead.
Further to the right, we find an envelope follower, which definitely does depend on input levels…
but if we look closer we see that it’s set to listen to the sidechain input, to which
I’ve routed the kick drum, so the resonance for the lower filter is turned down on every
kick drum.
Assuming you set up the same sidechain routing for each channel this would again behave the
same way running as multiple separate instances.
In fact the only real non-linearities in this preset are within the filters themselves.
If I switched both filters to the Clean type then this preset, as complex as it seems,
would be just as linear as a static EQ setting in Pro-Q3.
This Timeless preset is similar: the step sequencer modulation of the dry-wet mix would
be the same if applied to each individual channel… even the delay time modulation
would be the same… but I’m again using filter models which include saturation, and more
importantly, I’m modulating them with an envelope follower, and this time it’s set to listen
to the main inputs rather than the sidechain.
This preset is therefore definitely non-linear, and wouldn’t sound the same if applied to
each channel separately.
Finally here’s a Saturn preset, which modulates the Drive parameter for the Destroy type of
distortion.
This includes some clipping for very loud signals and bit crushing which mostly affects
quieter signals, but the most significant part of the sound is the sample rate reduction,
which doesn’t care about the input level at all.
However, the modulation in this preset has a strong random element, provided by the third
XLFO: if you ran individual instances for each channel, each would have different random
values, and the result would be very different.
If I put them all together… the individual parts get glitched up and modulated together,
creating something altogether new…
In this case the resonance of the Volcano high pass filter gets a bit overwhelming at
times, but rather than turning down the resonance and losing the self-oscillation between kick
drums, I’m instead going to use a multi-band compressor to keep the low frequencies under
control.
Multi-band compression can also be useful on buses, to control the extreme low or high
ends, or tame harshness in the upper mid-range, or selectively glue parts of the mid-range.
But we’re starting to go beyond beginners level now, so I’ll leave it there.
Bye for now, and thanks for watching.
In this tutorial, Dan Worrall explains the basics of bus processing, e.g. the advantages of grouping channels and processing them together, master bus processing, etc. More info and downloads of FabFilter plug-ins at: https://www.fabfilter.com/downloads https://www.fabfilter.com/shop

The Beginner’s Guide to Bus Processing video was embedded from Youtube channel “”. Video source

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FAQ FabFilter

Add audio files on both the main track and the side chain track. You can simply drag audio files from a Finder/Explorer window onto a track. Select the main track and add ProC 2 in the first Insert effect slot. Open FabFilter ProC 2’s interface, enable Side chain expert mode and set Side Chain to Ext.

Ableton LiveAt the left top of Ableton Live’s interface, click on the Plug-in Device Browser icon (third icon from the top). From the plug-ins list, double-click FabFilter/FabFilter Pro-Q 3, or drag it onto the track.

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You can uninstall FabFilter Total Bundle from your computer by using the Add/Remove Program feature in the Window’s Control Panel. When you find the program FabFilter Total Bundle, click it, and then do one of the following: Windows Vista/7/8: Click Uninstall.

You can simply download the latest installers from www.fabfilter.com/download . If you’d like to update/install multiple plug-ins, you can use the Total Bundle installer; during the installation process you will be able to choose the plug-ins that you like to install.1

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FabFilterFabFilter was founded in 2002 by Floris Klinkert and Frederik Slijkerman and is based in Amsterdam. They are known for making exceptional tools for audio processing that look great, such as the award-winning Pro-Q 2 plug-in.

Can FabFilter plug-ins be authorized to an iLok? No, our plug-ins don’t need an iLok. Instead, we use a fairly simple license key system. Just copy/paste the license key from your account or purchase e-mail into the license dialog and you’re done

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At FabFilter, we make the best possible tools for music production and audio processing. To achieve this, we continually rethink and challenge industry standards: we’ve never been afraid of reinventing the wheel. Considering every little detail, we tune our algorithms and interfaces until they sound perfect, look amazing and feel great to work with.

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