BtbN changed the topic of #ffmpeg to: Welcome to the FFmpeg USER support channel | Development channel: #ffmpeg-devel | Bug reports: https://ffmpeg.org/bugreports.html | Wiki: https://trac.ffmpeg.org/ | This channel is publically logged | FFmpeg 7.0 is released
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<kmikita>
BtbN, I see simple solution. Parse in libplacebo.pc existing pl_has_glslang=1 pl_has_shaderc=1 pl_has_vulkan=1
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<kmikita>
Nice. Now compilation ffmpeg+libplacebo done successfully.
<kmikita>
but also "Instance API version 1.1.0 is lower than the minimum required version of 1.2.0, cannot proceed!"
<kmikita>
good night to all))
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<aaabbb>
would anyone know what the best channel here to ask about aac (lc)? i am wanting to learn more about it so i can do a bit of an in-depth comparison between it and vorbis
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<aaabbb>
specifically aac from either coreaudio or fdk-aac (i assume the two are not much different in terms of things like pre-echo suppression, transient detection etc )
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<noobaroo>
how can i remove gaps in video (regenerate timestamps) without the hacky way of first going to raw h264 and then work from that ?
<noobaroo>
The only way that kind of works is by extracting the raw h264 stream to a file and then working from
<noobaroo>
the raw stream back to a container with -r 30 before the -i. But if it's a variable framerate video, then this method makes a warped file.
<noobaroo>
i asked in #mpv too since ive been asking here with no luck for a few days now
<noobaroo>
Its also kind of relevant to mpv since playing a web stream with mpv and seeking around through the video, then using dump-cache to file, results in a video with a bunch of empty parts
<noobaroo>
It should be possible to remove empty timestamps from a video without going to raw h264 first and working from a file without any timestamps, then making some up with constant framerate chosen by -r 30 as input option
<noobaroo>
I don't know how to explain it well but in my mind it should just be fairly simple math to move the timestamps over until they are all combined. Keeping the unique variable framerate timestamps and just throwing away all of the empty part of the video
<JEEB>
ffmpeg itself has discontinuity logic, but I guess you tried codec copy and it didn't trigger?
<JEEB>
(that logic can also be absolutely bad - but it would indeed stick things together)
<JEEB>
also I guess depends on the input container whether it gets done or not
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<noobaroo>
Hmm i think converting from mpegts might work all of the time or at least some of the time for this sort of thing
<noobaroo>
All of the input files were mp4v5 was messing up everytime, not sure about matroska, id assume mp4v5 wouldnt be different from any other mp4 since it's just the -brand tag or whatever
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<ciphrCat>
my phone produces mp4 files. Before uploading to peertube, should I transcode that to webm or mkv? Because IIRC mp4 is proprietary
<ciphrCat>
i am thinking this command will make a video more freedom-respecting before uploading, correct? → ffmpeg -i input.mp4 -c:v vp9 -c:a libvorbis output.mkv
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<noobaroo>
What is peertube?
<ciphrCat>
a place to publish videos
<ciphrCat>
.. one that is intended to respect people’s freedom in a decentralized way that does not like Google’s boots
<noobaroo>
I think the patent thing only matters if you are a large company or something... Web browsers only support mp4 and webm, AFAIK. And if you are going to use vp9 and vorbis, why are you using mkv with that?
<ciphrCat>
webm may be more suited, it sounds like
<ciphrCat>
i read something that a server can only publish a certain number of mp4 files (100,000?)
<ciphrCat>
so if I avoid mp4, then I don’t consume any of the hosting provider’s limit
<ciphrCat>
peertube supports the mkv format as an upload, but if it does not work on the web perhaps the server would have to convert it -- in which case I should avoid it so as not to burden the server
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<ciphrCat>
wow... huge filesize difference. A 20mb mp4 file became ~850kb mkv
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<furq>
peertube are just going to convert it back to mp4
<noobaroo>
That's because you are re-encoding it with lesser quality
<furq>
also i'm pretty sure the mp4 container patents all expired years ago
<ciphrCat>
from webm to mp4?
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<furq>
h264 still has active patents which is what you just encoded it to if you used mkv and didn't specify a codec
<furq>
you also don't need to pay for an h264 license unless you charge the viewer for it
<ciphrCat>
furq: did you see the command? ffmpeg -i input.mp4 -c:v vp9 -c:a libvorbis output.webm
<furq>
you said mkv
<furq>
you should also be using libopus for webm
<ciphrCat>
vp9 is the codec. not sure why the quality would significantly be reduced when I did not specify a lower resolution, but it seems fit for purpose nonetheless
<furq>
because you didn't specify any encoder options and the defaults aren't very good
<ciphrCat>
i switched to webm because of what noobaroo said.. it's better for web viewing
<furq>
at any rate the thing you said about servers only being able to publish 100k mp4 files is not true at all
<furq>
so you might as well just keep the mp4
<ciphrCat>
libopus, is that for sound?
<furq>
yes
<ciphrCat>
it's bizarre that the default parameters are not to retain the original parameters. Kinda seems to break the principle of least astonishment. If someone wants to change resolution or something they would say so.
<noobaroo>
furq Why should he be using opus? According to https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio , while not as good as opus, vorbis is equal to or superior to the proprietary fdkaac. And everything that plays opus should play vorbis. My old TV supports vorbis but doesn't support opus
<ciphrCat>
anyway, the defaults are good in my case because the output file is good enough quality. I'm quite happy that the video is under 1mb
<furq>
i assume your tv doesn't support webm so that's academic
<noobaroo>
That's correct, but what does academic mean?
<ciphrCat>
i actually decided to remove the audio (-an) because it's a nature video.. no interesting sound anyway
<furq>
not practically relevant
<furq>
also i'm pretty sure modern ios supports opus in webm but not vorbis
<furq>
vp9 and opus is what everyone uses for webm so it wouldn't surprise me at all if there was more stuff that didn't support vorbis
<furq>
opus and flac are the only xiph codecs that got any kind of mainstream traction
<noobaroo>
I hadn't looked at the page in a while, but I checked and it says vorbis is the same as opus now. It used to have a > symbol and now it's a ~ symbol
<furq>
that would be surprising considering vorbis hasn't been updated in four years
<noobaroo>
I'm also surprised
<furq>
i'm not that surprised that people just write whatever they feel like on a wiki
<noobaroo>
Newer studies can always be conducted, however.
<furq>
ok and were they
<noobaroo>
Ask the person who changed the trac.ffmpeg/wiki page, I just discovered this change myself, I am as surprised as you
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<furq>
i am comfortable just assuming it's wrong
<furq>
it's much simpler to do that
<noobaroo>
I'm guessing it's a higher bitrate thing though, i have no doubt that at under 128k opus sounds better. But I can definitely believe that at higher bitrates vorbis retains more of the spectrum. I've read some things about this before in audiophile forums
<noobaroo>
Who edits the wiki? You don't even know?
<furq>
apparently some guy called MasterQuestionable
<furq>
wasn't someone in here complaining about his edits to another page
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<furq>
anyway unless someone cites an actual listening test that has more than one participant then it's not worth anything
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<furq>
so just use whatever normally goes in that container
<furq>
which for webm is opus
<noobaroo>
If i have slow internet and i want to use ffmpeg to seek and cut a section of a huge file over https://, is there any options i can pass to minimize any unnecessary info downloading?
<noobaroo>
Something like the output option `-map_metadata -1`, but maybe like some input option version?
<furq>
not sure what kind of info downloading you mean
<furq>
unless it's mpegts or flv you will always need to download headers or you won't have the seek index
<furq>
and then you have no idea what range you need
<noobaroo>
Im trying to extract only the audio from a 4k video and its going at 0.1x
<noobaroo>
I guess its downloading the whole file even though i passed `-vn` and `-map 0:1`
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<furq>
the audio packets are interleaved with all the other streams
<furq>
-map and -vn are output options so they wouldn't have any effect anyway
<furq>
but there wouldn't be any way to do it in general
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<noobaroo>
-vn is input option
<noobaroo>
Are you in #yt-dlp by any chance, I just said something kind of long in there by accident, it was meant for here
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