A few days ago, I’ve released my first VST plugin. Now it is time to analyze how it works.
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: https://sourceforge.net/projects/qtvst/files/QtSimpleOverdrive/
If you find any issue, please submit it on Github: https://github.com/mbrucher/qtvst
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).
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.
I’ve recently burnt the tone circuit of my Cort GB-74, so I’ve changed it for a Seymour Duncan STC-2A. The basic wiring diagram does not include all the options I had on the original diagram, here are the (small, really small) modifications and some impressions on the STC.