This entry is part 1 of 4 in the series Audio Toolkit algorithms

I’ve started working on adaptive filtering a long time ago, but could never figure out why my simple implementation of the RLS algorithm failed. Well, there was a typo in the reference book!

Now that this is fixed, let’s see what this guy does.

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After my post on HPCToolkit, I felt that I prefered QCacheGrind as a GUI to explore profiling results. So here is a gist with a Python script to convert XML HPCToolkit experiments to callgrind format: https://gist.github.com/mbrucher/6cad31e38beca770523b

For instance, this is a display of an Audio Toolkit test of L2 cache misses:

ATK L2 cache misses profile
ATK L2 cache misses profile

Enjoy!

Convolution is an algorithm that is often used for reverberations. If the equation is easy to understand and to implement, the implementation is costly. The other way of doing it is to use Fast Fourier Transform (FFT), but the direct/crude implementation requires latency. If it is possible to optimize the basic convolution code, it is sometimes more interesting to use a different algorithm, as it is the case here.

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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.

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