[FFmpeg-devel] [PATCH] avfilter/alimiter: Add "flush_buffer" option to flush the remaining valid data to the output

[Paul B Mahol]

On Tue, Apr 5, 2022 at 8:57 PM Wang Cao <wangcao-at-google.com at ffmpeg.org>
wrote:

> On Mon, Apr 4, 2022 at 3:28 PM Marton Balint <cus at passwd.hu> wrote:
>
> >
> >
> > On Mon, 4 Apr 2022, Paul B Mahol wrote:
> >
> > > On Sun, Mar 27, 2022 at 11:41 PM Marton Balint <cus at passwd.hu> wrote:
> > >
> > >>
> > >>
> > >> On Sat, 26 Mar 2022, Wang Cao wrote:
> > >>
> > >>> The change in the commit will add some samples to the end of the
> audio
> > >>> stream. The intention is to add a "zero_delay" option eventually to
> not
> > >>> have the delay in the begining the output from alimiter due to
> > >>> lookahead.
> > >>
> > >> I was very much suprised to see that the alimiter filter actually
> delays
> > >> the audio - as in extra samples are inserted in the beginning and some
> > >> samples are cut in the end. This trashes A-V sync, so it is a bug
> IMHO.
> > >>
> > >> So unless somebody has some valid usecase for the legacy way of
> > operation
> > >> I'd just simply change it to be "zero delay" without any additional
> user
> > >> option, in a single patch.
> > >>
> > >
> > >
> > > This is done by this patch in very complicated way and also it really
> > > should be optional.
> >
> > But why does it make sense to keep the current (IMHO buggy) operational
> > mode which adds silence in the beginning and trims the end? I understand
> > that the original implementation worked like this, but libavfilter has
> > packet timestamps and N:M filtering so there is absolutely no reason to
> > use an 1:1 implementation and live with its limitations.
> >
> Hello Paul and Marton, thank you so much for taking time to review my
> patch.
> I totally understand that my patch may seem a little bit complicated but I
> can
> show with a FATE test that if we set the alimiter to behave as a
> passthrough filter,
> the output frames will be the same from "framecrc" with my patch. The
> existing
> behavior will not work for all gapless audio processing.
>
> The complete patch to fix this issue is at
>
> https://patchwork.ffmpeg.org/project/ffmpeg/patch/20220330210314.2055201-1-wangcao@google.com/
>
> Regarding Paul's concern, I personally don't have any preference whether to
> put
> the patch as an extra option or not. With respect to the implementation,
> the patch
> is the best I can think of by preserving as much information as possible
> from input
> frames. I also understand it may break concept that "filter_frame" outputs
> one frame
> at a time. For alimiter with my patch, depending on the size of the
> lookahead buffer,
> it may take a few frames before one output frame can be generated. This is
> inevitable
> to compensate for the delay of the lookahead buffer.
>
> Thanks again for reviewing my patch and I'm looking forward to hearing from
> you :)
>

Better than (because its no more 1 frame X nb_samples in, 1 frame X
nb_samples out) to replace .filter_frame/.request_frame with .activate
logic.

And make this output delay compensation filtering optional.

In this process make sure that output PTS frame timestamps are unchanged
from input one, by keeping reference of needed frames in filter queue.

Look how speechnorm/dynaudnorm does it.


> --
>
> Wang Cao |  Software Engineer |  wangcao at google.com |  650-203-7807
> <(650)%20203-7807>
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