Request for errata: Minimal delay for block size adaptation (LAC 2026) (Matthias Geier)

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> Hi all.
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> 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"
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> The PDF of the paper is available at =
https://lac2026.sciencesconf.org/722511
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> The video recording of the talk available at
> https://tube.mucs.club/w/7a837NuQo6radV2NpmNvEJ
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> In the comments of the live stream there were a few questions and
> complaints that I would like to discuss here.
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> 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]".
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> Reference [3] is: St=C3=A9phane Letz, =E2=80=9CCallback adaptation =
techniques,=E2=80=9D
> Technical report, GRAME, 2001, https://hal.science/hal-02158912v1.
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> 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
> St=C3=A9phane Letz, which we mentioned both in the presentation and =
the
> paper.
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> 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!
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> 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!
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> 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.
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> cheers,
> Matthias
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> 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.
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So what is the final conclusion ? Nothing new under the sun 25 years =
after the 2001 paper ? Or a better expressed formula ?

Thanks.

St=C3=A9phane Letz