Category Archives: Digital filters

Announcement: QtSimpleOverdrive 1.0 (QtVST)

I’m pleased to announce the release of my first VST plugin (Windows 32bits), based on the simple overdrive prototype.

It is a mono filter, with an oversampling of 2 to 32, based on polyphase filters, and the undersampling is done after an 8th order Butterworth lowpass filter with a cut frequency of 22kHz.

The source code will be available (under the GPL) in the future if there is interest in the plugin and its support. The exact way it works will be explained in a future blog post.

The audio plugin is available on Sourceforge:

If you find any issue, please submit it on Github:

Please note that the oversampling can be quite CPU intensive (on my laptop, an Intel Core2 T7200, using an oversampling of 4 at 96kHz uses the full power of one core).

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Electronic: The purpose of an oversampling filter

A few months ago, I’ve posted a note on an overdrive. The main issue of this kind of non-linear filter is aliasing, a process that adds digital acoustic content. The best way to solve the issue is to oversample the input before processing the signal.

Continue reading Electronic: The purpose of an oversampling filter

Electronic: Simulation of a simple overdrive

There are some effects that are simpler than other. Digital ones are generally easier than analog ones, and purely digital filter are also easier than digitally-transformed analog ones. Linear filters such as passband, cutband, … are easy to digitally design, chorus can be achieved through some spectral computations, delay and reverbation are computationnally expensive but easy to code.

It said that analog devices have a unique sound that digital devices cannot achieve. In fact, much is due to the simplications that occur when digitizing an analog device. One of the most blatant examples is the overdrive, which I took from Simulanalog.
Continue reading Electronic: Simulation of a simple overdrive