The main changes for this release are first trials at modulated filters, C++11 usage (nullptr, override and final), and some API changes (the main process_impl function is now const).
Audio Toolkit is now almost ready for its first stable release. Its content will now move toward more advanced DSP algorithms (zero delay filters, amplifiers).
I just released my SD1 simulation, and now it is time to explain why is inside this plugin. Not everything in the original pedal was simulated, and different schemes were used to tackle the different stages.
It’s time for a new release of the toolkit. Much has been done in terms of basic filters, but also to simplify usage (static and shared libraries are compiled, no need to reset the pipeline before calling process…).
I’ve explained in earlier posts how to simulate a simple overdrive circuit. I’ve also explained how I implemented this in QtVST (and yes, I should have added labels on those images!), which was more or less the predecessor of Audio TK.
The main problem with simulating non linear circuits is that it costs a lot. Let’s see if I can improve the timings a little bit.
A few days ago, I’ve released my first VST plugin. Now it is time to analyze how it works.
Continue reading QtVST: how QtSimpleOverdrive is implemented
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).
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.