Timbre: Alternately tone quality or tone color, this is basically the percieved personality of a sound that cannot be measured.

Different plug ins might have similar goals, there are endless plug ins to emulate a piano for instance, but you may still wish to keep different plug ins of similar types for their individual personality or timbre.

Plug-ins will come generally in two flavors, each with two subdivisions.

  • Instruments & Generators
    • A synthesized instrument is one that generates sound entirely from algorithms stored in the plug in. It keeps no recordings of actual instruments. These types of plug-ins are typically more processor intensive (they require faster computers).
    • There are also "sample-based" generators which rely on pre-recorded sound files. These often work by recording one or several octaves worth of notes and then modifying the sound on playback to achieve different pitches or other effects.
  • Processors & Effects
    • As discussed before, some processors and filters might just adjust the inherent properties in a signal.
    • Others are "additive" or modify the signal based on itself from different positions or others channels entirely.

Plug-in Formats

There are a few different types of plug-ins. None are "Better" than the others as these types simply describe how the application interfaces with a host and not how the actual sound is generated or its content.


Virtual Studio Technology. A plug in that either emulates real world technology such as sound pedals or modulators or modulates it in a purely digital way.

Please note; "VSTi" refers to an instrument, if you see just "VST" then it probably is an effect meant to be applied to a sound channel instead of a generator itself.

The actual interface spec for the VST format makes it compatiable with Windows, Mac and Linux.


AKA the DirectX Instrument plugin, this format is probably not as widely used as the others.

Generally DX plug-ins are only available on Windows machines because of their reliance on DirectX.


"Real Time AudioSuite" is a plug-in format first created by Digidesign, now avid technology, primarily for their own use.

However, as of version 11, AVID Pro Tools no longer supports the RTAS format. The given reason is that the RTAS format only has a 32-bit spec while modern computers have moved on to 64-bit architectures. AAX has been the primary plug-in format for Pro Tools since.


AAX (Avid Audio eXtension) is the latest format introduced by AVID specifically for their ProTools machines. It is fully 64 bit. It generally has two forms as stated by Avid:

  • AAX DSP: "compatible with Pro Tools|HDX only (Pro Tools|HDX does not support TDM)"
  • AAX Native: "compatible with any system running Pro Tools/Pro Tools HD 10 or higher."


Linux Audio Developer's Simple Plug-in API. Basically a Linux VST alternative.

Isolating effect sounds

Sometimes when using effect plug ins you'll come across a need to adjust not just the effect parameters but the amount of the effect itself. Usually a good plug in will save you some trouble and include the aforementioned dry/wet dial. But what if it does not and you do wish to isolate the sound of an effect away from the original input sound?

If an effect does not have a dry/wet dial then it is possible to simulate one by splitting the output of a track into two send/insert channels and then reversing the polarity of one we can isolate the effect itself. This of course is easier when the mixer panel has a reverse polarity option, otherwise a simple phase inverter will do.

Let's look at the basic, non-program specific, steps.

  • Assign the instrument to an input channel.
  • From that input channel mute the signal to the master output and send it to two other unused input channels
  • Within the first of those two input channels reverse the polarity, either via a toggle or by a phase inverter plug-in.
  • In the second apply the effects you actually want to isolate.

As long as both of the two channels the original input was sending two are going to the master output then you should only hear the effect sound since one input channel will be phase canceling information in the other.

It's not a perfect approach as it will be subject to the phasing effects of the effect you want to retain, but it typically works well.

Keeping the bass low

In music, making a bass that can be loud enough to drive the composition but not overpower the other instruments can usually be as simple as filtering the mid and higher frequencies.

This is typically as simple as taking any given kick (or drum) sample and applying a low pass filter. The sample below is filtering all frequencies about 75 hz (with a somewhat soft knee).

A simple series of notes playing a kick sample.
The same notes, but with a low pass filter applied, removing all frequencies above 75 hz.

It doesn't sound like much on its own, but it will allow the other instruments to "breathe", and allow them to be clearly heard on their own.