> > Hi all. > > A bit more than a week ago, Matthias Rath and I presented a paper at > the Linux Audio Conference in Maynooth: > "Minimum required delay for realtime block size adaptation in digital > audio signal processing" > > The PDF of the paper is available at https://lac2026.sciencesconf.org/722511 > > The video recording of the talk available at > https://tube.mucs.club/w/7a837NuQo6radV2NpmNvEJ > > In the comments of the live stream there were a few questions and > complaints that I would like to discuss here. > > One of the complaints was that this problem was solved years ago and > used in a lot of code, which we acknowledged in the presentation. In > the paper we wrote: "While the problem of block size adaptation has > for sure been tackled and solved multiple times over the past decades > within the implementations of several realtime applications, the only > paper where we found it discussed in any depth is [3]". > > Reference [3] is: Stéphane Letz, “Callback adaptation techniques,” > Technical report, GRAME, 2001, https://hal.science/hal-02158912v1. > > The problem is that when I looked for papers/reports/websites/code, I > could only find the PortAudio code we showed in the presentation (and > which is also in the paper). Later I was pointed to the paper by > Stéphane Letz, which we mentioned both in the presentation and the > paper. > > Other than that, I couldn't find anything, so if anyone reading this > knows any additional sources (books, papers, websites, code bases, > ...), please share them here! > > Apart from this, we were also told that we got something wrong, > without specifying what exactly it is. So if anyone spotted anything > that we might have gotten wrong, please share it here! > > We might not be able to publish formal errata to the paper, but at > least this mailing list thread can provide errata which will then > hopefully become discoverable for anyone searching for that topic. > > cheers, > Matthias > > P.S. the abstract of the paper, for quick reference: When a > block-based realtime audio application invokes a signal processing > component which uses a different block size, this requires some > buffering of audio data between invocations of said signal processing > component. This is sometimes called reblocking. Depending on the two > block sizes, this buffering may introduce a delay. This paper answers > the question of the minimum such delay required for any given > combination of the two block sizes. >
So what is the final conclusion ? Nothing new under the sun 25 years after the 2001 paper ? Or a better expressed formula ? Thanks. Stéphane Letz
