
I’ve recently found the phrase “best dnb plugin” in the set of terms people searched for in this blog. Well, reading that, it’s pretty hard for me to pass by without commenting on it.
When we decided to seriously start producing DnB (after a couple of years just playing around and getting a feel for electronic music production in general), we were wondering, too, what that plugin was every huge producer was using to get that (then) fat sound we repeatedly failed to achieve. In the first couple of months, we just stacked loads of sounds on top of each other, pushing each element through endless numbers of every plugin we could get our hands at, hoping to one day find the one we were looking for. Obviously, there is none. How come?
Basically, plugins generate or transform audio. Left aside those generator plugins like soft synths, let’s focus on the huge number of transformation plugins. Having written my university’s master thesis about Audio Fingerprinting including the design and implementation of a generic, DirectX-plugin-based processing framework, I know quite a lot about the internals of plugins. Let me make this clear: There is no magic involved.
Plugins transform audio input in small fragments of about 10ms length, each using an algorithm which uses parameters the user can tweak via the plugin’s user interface. These algorithms can work in two “domains”:
- The time domain, which is the base of the typical waveform data view. Examples: delay, reverb, compressor
- The spectral domain, which allows for transforming frequencies. Examples: EQ, pitch-shift
(To be precise, I’ll have to add that quite a lot of modern time-domain-based plugins also work in the spectral domain, and that the two domains are connected views on the same facts, but let’s keep it simple for now.)
To properly produce a well-designed tune, you need plugins of both types. But, honestly, you won’t need stacks of them. Also, there is no “best dnb plugin”. What you want plugins for, is mainly one reason: Imagine your tune as a seamless sequence of real-time-data, just like freezing every split-second when you listen to a track. Now, you’ve splitted the time domain’s data into little pieces. In our virtual fragment of the audio stream, when switching to the spectral domain, we encounter a lot of frequencies. You’ll experience them as bass, mids, highs, but not clearly separated, but along an axis, ranging from 30Hz to about 20KHz. You may have seen that, when switching to your audio editor’s “spectral view”, most of which visualize high amounts of energy within a frequency band (meaning a part of the spectral axis) as red/warm and low amounts as blue/cold.

The human ear micro-mechanically perceives this spectral data within the virtual frozen time step as sound. Music in general works by expressing data in certain ranges along the spectral axis in form of more or less typical patterns. Non-musical sounds express different patterns, like white noise, which equally gouverns spectral data along the whole axis. For well-designed music, space is the place…eh..key: If you put in sounds which clutter a certain part of the spectral band, it’ll become too crowded, making it sound awkward and undefined. To better design your tune, you have to focus on certain spectral key areas, while suppressing others. This is not only true for the spectral, but for the time domain! Be sure to separate sounds as good as possible. How?
An EQ provides the primary mechachism to achieve that. By using it, you can cut spectral parts of a sound while emphasizing others. Make sure you process EVERY sound element of your tune with an EQ, cutting everything that’s not absolutely neccessary. You’ll quickly hear how your tune becomes more clearly defined. Now focus on key instruments, like kick and snare, pushing them in certain spectral bands. Try using A/B comparison with similar, well-known tracks by other artists to determine the degree of change. When processing elements which work in the same spectral area, like bass and kick, try to separate them as good as possible, so they both have “air to breathe”.
Also, try using the second important plugin, the compressor. It primarily works in the time domain and helps you dynamically cut out peaks in the waveform, therefore emphasizing the overall. One could also say, by using it you can push up the energy while skipping the peaks. But be careful not to over-do it, as it’ll easily rip all life from your sounds, like taking away every punch. I’ll post on that in a different post.
All other plugin types are secondary. You’ll need them for certain tasks, but in the end, they’re not as important as EQ and compressor. Concerning the plugin’s vendors, I found it’s not really that important which ones you use, as long as you use them at all. We sometimes use special EQ and compressor plugins for certain tasks, but for the most time, we just use the standard tools provided by the sequencer. There just is no “best EQ/compressor”, and, as shown in the plugin performance post, you just have to take care of the amount of CPU usage each plugin consumes. Just first learn to set the parameters right, then start looking for better sounding, more power-consuming plugins and use them when appropriate.