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Comment by cyphar | Hacker News Reader

It would be cool if the hardware did some Fourier analysis to resample the buffers it gets (so it could make them run for longer if the buffers run dry), but it probably would cause some kind of latency issue. I reckon that just avoiding buffer underruns is less overhead.

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I had to cut off an audio file abruptly once, and I found that I could smooth out the abruptness by quickly fading in a reverberated version of the audio, just as the original audio was about to end, and then letting it ring for a fraction of a second after the original had ended.

(I say "fading in", but it might have been that I had the reverb applied but dry, and transitioned to wet just before the signal ended.)

A colleague of mine actually patented this technique: http://www.google.com/patents/US8538038

:-/

Yep, this should not be a patent..It's a trivial solution to the problem that could be devised by any sound engineer!

If you have any evidence of prior work to that patent (I really hope you do), then you can topple that patent. If your code to do what you said isn't free software I'd recommend you release it as free software now.

I can't check right now, but it's possible that the patent predates my use of the technique. Even if not, I don't know how I could prove it.

By the way, it wasn't done in code; I did it manually in Ardour [0].

[0] <https://ardour.org/>

This seems like the most sensible thing to do, and was actually what I was getting at :)

What do you mean by latency issues? Why would there be any?

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