Hi Guy, thanks for sharing this and welcome here. Whats' your IRC handle? I'm asking since I'm also very slowly teaching myself audio DSP. DISTHRO Framework was the right choice, I also use it. Btw Gui is great looking, I'll test the plugin soon. cheers Gerald /IRC: JimsonDrift On 01.09.2015 18:09, Guy Sherman wrote: _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev(a)lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev

Welcome to the community, Guy. Congrats on your first audio plugin. My IRC handle is andrewrk. I've been working on Genesis DAW: http://genesisdaw.org/ See you around! On Tue, Sep 1, 2015 at 9:09 AM Guy Sherman &lt;guy(a)guysherman.com <mailto:guy@guysherman.com>> wrote: Hi all, I just found this list after being directed to the IRC channel over the weekend. I'm new here! Lately I've been dabbling in a bit of open-source audio development (I say open-source rather than linux, because I've been dabbling on both linux and windows). Anyhow, amongst other things I've been trying to teach myself about DSP, so I wrote a really (really, really, really) naive distortion plugin. I was wondering if anyone would be interested in taking a look at it and giving me some feedback, and tips on where to go next. I wrote a little about it here: http://guysherman.com/2015/08/30/my-first-ever-audio-plugin/ And the code is at: https://github.com/guysherman/si-plugins I've got some other projects on the boil that I've been talking about with the crew from the Ardour list, which I'll mention here when they take shape a little more. Cheers, Guy. -- Guy Sherman *e:*guy@guysherman.com <mailto:guy@guysherman.com> *w: *http://guysherman.com <http://guysherman.com/> _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev(a)lists.linuxaudio.org <mailto:Linux-audio-dev@lists.linuxaudio.org> http://lists.linuxaudio.org/listinfo/linux-audio-dev

[Guy Sherman] For high-bandwidth input or high-gain clipping, you'll need to run the nonlinear operator at substantially elevated sample rates unless you want synthesised harmonic content to alias audibly. I'm fine with 4x oversampling for electric guitar at 48k but going higher can still improve things depending on circumstances. Also, it's not a bad idea to put a (preferably configurable) pair of high- and lowpass filters both before and after a strong nonlinearity. Asymmetric clipping gives rise to a DC component which most sane people agree needs to be filtered out. Too much bass going in makes a clipper sound farty, and lowpass filters taking some edge off pre- and post-distortion are highly useful for musical purposes. And welcome to the list, Tim

[Guy Sherman] interpolate The principle is indeed the same, and you could use a converter library for this purpose. However, those converters are designed to work over a continuous range of samplerate ratios whereas the ratio chosen for oversampling is usually a fixed integer because this presents ample opportunity for optimisation. The interpolation filters in both cases are usually windowed sinc FIR (much like the Lanczos kernel in image resampling). As you intuit: you sample up, process, then sample back down, ending up with one output sample for every input sample. IIrc, http://quitte.de/dsp/caps.html contains at least two oversampled plugins and comes with source code. Cheers, Tim