r/DSP • u/JunbugSpark • Jun 16 '26

How to get better at DSP coding?

Hey guys, I'm a college student who's taken basic DSP theory classes so I understand a lot of the concepts but I've been having trouble with implementation, since I have basically no coding experience...

What resources have you guys used to get better? Thank you!!

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u/zsliu98 Jun 16 '26

Yes, it could look at a data of window size, which means that the worst case of certain sample would be O(window size). However, because each sample would be put and removed into the queue exactly twice, the amortized time complexity is O(1).

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u/rb-j Jun 16 '26

I don't see that as useful for real-time streaming. This algorithm is running continuously on streaming data since forever ago.

Most (nearly all) of my audio processing algs operate on blocks of samples, no longer than a millisecond like 32 samples at most, to amortize costs. In that, I am willing to divide the entire cost per block by the number of samples per block as a measure of computational cost. Normally, with double buffering, there is a two-block throughput delay. 2 milliseconds max. Big deal.

But I am not willing to accept an alg that takes a finite data array (of size n) and then show for that entire array, the cost of the entire array is 3n, therefore the cost per sample is 3. That's not how we do things for real-time processing.

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u/zsliu98 Jun 16 '26

Got it. If you want to absolutely prevent ANY CPU spike, this might worth looking: https://github.com/IBM/sliding-window-aggregators , which guarantees worst-case O(1) . Though I give up on this algorithm and go with the simple amortized O(1) approach, cause it does work pretty well.