On Sat, 5 Dec 2020, Michael Niedermayer wrote:
On Sun, Nov 15, 2020 at 01:14:55AM +0100, Marton Balint wrote:
On Fri, 6 Nov 2020, Michael Niedermayer wrote:
On Wed, Nov 04, 2020 at 10:44:56PM +0100, Marton Balint wrote:
On Wed, 4 Nov 2020, Michael Niedermayer wrote:
we have "millisecond" based formats, rounded timestamps
we have "exact" cases, maybe the timebase being 1 packet/frame per tick
we have "high precission" where the timebase is so precisse it doesnt matter
This here though is a bit an oddball, the size if 1 PCM frame is 1 sample
The timebase is not a millisecond based one, its not 1 frame either nor is
it exact nor high precission.
Its 1 video frame, and whatever amount of audio there is in the container
which IIUC can differ from 1 video frame even rounded.
maybe this just doesnt occur and each frame has a count of samples always
rounded to the closes integer count for the video frame.
The difference between the audio timestamp and the video timestamp for
packets from the same DV frame is at most 0.3929636797*frame_duration as the
specification says, as Dave quoted, so I don't see how the error can be
bigger than this.
It looks to me you are mixing timestamps coming from a demuxer, and
timestamps you calculate by counting the number of demuxed/decoded audio
samples or video frames. Synchronization is done using the former.
But if for example some hardware was using internally a 16 sample buffer
and only put multiplies of 16 samples in frames this would introduce a
considerable amount of jitter in the timestamps in relation to the actual
duration. And using async to fix this without introducing more problems
might require some care.
I still don't see why timestamp or duration jitter is a problem
as long as
the error is below frame_duration/2. You can safely use async with
min_hard_comp set to frame_duration/2.
Thats exactly what i meant. an async like filter which behaves differently
or async with a different value there can mess this up.
IMHO such mess up is ok when the input is corrupted or invalid. OTOH
here it is valid and correct data.
In general, don't you find it problematic that the dv demuxer can return
different timestamps if you read packets sequentially and if you seek to the
end of a file? It looks like a huge bug
yes, this is not great
but even with your patch you still have this effect
when seeking to some point in time a player has to output video and
audio to the user at an exact time and that will differ even with async
from linear playbacks presentation
which is not fixable if you insist
on sample counting...
I think you misunderstood me, or maybe i didnt state my opinion well,
iam not saying that i consider what dv in git does good. Rather that there
is a problem beyond what these patches fix.
Some concept of timestamp accuracy independant of the distance of representable
values would be usefull.
if you take teh 1/25 or whatever they are based on dv timestamps and convert
that
to teh mpeg 90khz based ones thats not making it that accurate.
OTOH if you take 1/25 based audio where each packet is 1/25sec worth of samples
that very well might be sample accurate or even beyond.
knowing this accuracy is usefull for configuring a async like filter or also in
knowing how to deal with inconsistencies, is that timestamp jtter ? or the
sample
rate jittering / some droped samples ?
Its important to know as in one instance its the timestamps that need adjustment
while in the other the samples need adjustment
ATM its down to the user to figure out on a file by file base how to deal or
ignore this. Instead it should be possible for an automated system to
compensate such issues ...
OK, but the automated solution is far from trivial, e.g. it should start
with a analysis of the file to check if the sample rate is accurate or
not... And if it is not, is the difference constant througout the file? Then
there are several methods to fix it and the user might have a preference.
E.g consider audio clock "master" and duplicate/drop video frames. Or keep
all video frames, but stretch audio (with or without pitch correction - and
which filter you want for pitch correction? atempo? rubberband?). So making
it automated is not trivial at all.
Anyhow, is it OK to apply this patch then?
yes
Thanks, applied.
Regards,
Marton
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