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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<vlt>
finsternis: What does `ffprobe input.avi` look like?
<vlt>
finsternis: A not so elegant solution (ignoring the actual reason for the color change) might be crop and pad.
<vlt>
finsternis: black might be default for pad. Or don’t pad at all.
<vlt>
finsternis: To get the cropping right, you could try with something like "-t 30" first and "-preset ultrafast".
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<welder>
Hi, how can I preserve the bitrate when trimming an .opus file with -ss <time>? It's different than the original
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<aaabbb>
welder: are you doing -c:a copy?
<aaabbb>
also opus uses a variable bitrate. the part that you trimmed out might have a lower or higher average bitrate than the remaining part, so the average bitrate may change a little anyway
<aaabbb>
but if you aren't doing -c:a copy, then do it. the best you can do is ffmpeg -ss $time -i in.opus -c:a copy out.opus
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<shinjite>
Hello, not sure why I am banned from #ffmpeg-devel but I thought thats where you would ask questions regarding libav*** use
<shinjite>
Anyways, what is correct way to really deal with a byte array to avcodec alone - no format context
<nemo_magneet>
people here dont like questions
<nemo_magneet>
exept A3B3, he is good
<shinjite>
I have RAW h265 frames already demuxed by someone else passed into me. It goes like this https://pastebin.com/xD5WeJEA
<shinjite>
i think... its cause im not draining it but that last bit of code i added drains it...
<shinjite>
Iwrote that just now after thinking about it for several hours last night
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<vlt>
Hello everyone! To use my DSLR as a webcam, I use `ffmpeg -f mjpeg -framerate 25 -i - -codec:v rawvideo -f v4l2 /dev/video` to capture gphoto2’s stdout. That works perfectly well. For one event I wanted an additional recording of the stream. What I *should* have done is run a second ffmpeg with "-i /dev/video0" like a normal person, but what I did instead was to add a second output stream to
<vlt>
the capturing process: `... -codec:v ffv1 rec.mkv`.
<vlt>
That worked kind of, but the framerate of the resulting video is all over the place.
<vlt>
Is there any way to reconstruct the PTS of the frames in that recording?
<vlt>
Does FFV1 support this?
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<amdbcg>
hi
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<dev-lizard>
I would like to develop on ffmpeg, I read the rules and style guides. any suggestions would be appreciated.
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<JEEB>
dev-lizard: build git master first, then whatever is your pet peeve, since you usually have most interest towards something you care about
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<dev-lizard>
JEEB: master is built to 7.0 with make && make install, pet peeve is from : "Stream %d, codec %s, is muxed as a private data stream "
<dev-lizard>
"and may not be recognized upon reading.\n , I found it . The original looks like : Stream 2, codec scte_35, is muxed as a private data stream and may not be recognized upon reading. I'm pretty sure private data stream means "isn't in the list of codecs" , I'd like to add codec for scte_35 so it keeps the formatting
<dev-lizard>
and such. Not sure how to start that other than add another case statement that leads to somewhere
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<JEEB>
dev-lizard: what container? mpeg-ts? IIRC there were various gotchas regarding scte35 passthrough since the packet contains a timestamp that may have to be adjusted when being written
<JEEB>
which seems to be the latest from july last year
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<JEEB>
it didn't get any LGTMs so it's unlikely it's in master, and the patches contain quite a bit of commentary why having it exact is harder than what the patch does
<dev-lizard>
yes.
<dev-lizard>
is there a way to get a diff or a branch of the patches and apply locally to see what it does?
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<dev-lizard>
will 'series' work?
<dev-lizard>
it has been a year , I wonder if the patch series download still works on master
<dev-lizard>
error: libavcodec/packet.h: patch does not apply
<dev-lizard>
Patch failed at 0001 avcodec: Add new side data type to contain original PTS value
<dev-lizard>
hint: Use 'git am --show-current-patch=diff' to see the failed patch
<dev-lizard>
When you have resolved this problem, run "git am --continue".
<dev-lizard>
If you prefer to skip this patch, run "git am --skip" instead.
<dev-lizard>
To restore the original branch and stop patching, run "git am --abort".`
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<dev-lizard>
I'll use a nopaste next time, sorry about the length
<JEEB>
yea, multi-line stuff onto a pastebin-like site of your preference
<JEEB>
but now you can at any point `git am --abort` to go back to your previous state before applying anything. but you can either try fixing it yourself, or try running git am with the `-3` three-way option
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<JEEB>
(3way works if the thing that patch set starts off of is something that exists in the repo - which in theory it should be)
<JEEB>
if it fails tho, then you will have to go back to manual fixing :)
<JEEB>
you should never start off a branch to be honest :P it's interesting on which version a patch does or doesn't apply, but at the end of the day anything that would go in would at this point be on master
<JEEB>
but anyways, someone needs to talk regarding the issues mentioned with that set, and whether some of them can be improved and whether others can be addressed later
<dev-lizard>
it's pretty much what I want solved, but it's got a few compile time issues and doesn't compile with master. I could try manually applying all the changes to see if that solves it. line numbers probably changed from 6.1 to master.
<dev-lizard>
*manually applying changes to master
<dev-lizard>
thanks for the help/references
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<JEEB>
np
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<GrandPa-G>
I have a portrait short video that is 2160x3840. I want to make a new video of it with the same video duplicated on its side so together the video is 4320x3840. Would anyone give me a command that would do this (or even a web process). Please don't just say use ffmpeg, I need the exact command line. I have tried many options and none seem to really work.
<jkqxz>
"ffmpeg -i input -vf split,hstack output"
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<dev-lizard>
JEEB : the code looks like its compiling on master for the patch I manually applied. 1 down 4 to go. How do I update or provide a comment for the patch so its easier to apply to master for the next developer?
<GrandPa-G>
jkqxz, what does split, hstack refer to?
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<slobodan>
Hey guys
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<slobodan>
I wont to burn divx file to cd so is Codec : isom (isom/av01/iso2/mp41) Divix codec, container is .mp4
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<dev-lizard>
how do I prove that my build is recent build of ffmpeg? is there a way to see git hash of build?
<another|>
ffmpeg -version
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<bcn_>
Is there a linux AV enthusiasts group somewhere? I'm looking for a handy tool to synchronyze multiple video and audio streams (mostly just multiple audio) into one MKV file so that the viewer can change audio channels and hear different places in the room
<bcn_>
Traneptora: that would work if I knew with certainty, that all tracks began at the same milisecond in time. (I don't)
<nemo_magneet>
Like concert videos?
<Traneptora>
oh, well you'll have to do that manually then
<Traneptora>
cut them manually and then mix them together
<Traneptora>
you can always put -ss before -i to cut stuff from the beginning if you want to test the final product
<Traneptora>
mpv, the media player, can switch audio tracks if you push '#' (that's Shift+3) for testing
<bcn_>
Traneptora: hmm. I;m hoping there is a way to use audio thumbprints of some sort to automatically sync different tracks' audio. All my files have audio and I would think like the same waveform present in all of them would queue it to sync
<bcn_>
then again I do understand no two mics record identically. but we have tech today that identifies the song being played in a roo
<bcn_>
i also understand that any time a microphone moves, it will receive different sound sources at different times. but they're mostly always in the same room, paper airplane distance from eachother'
<Traneptora>
automatically synchronizing the two to millisecond accuracy is a hard problem
<Traneptora>
especially when the microphones aren't identical distances from the source, and the speed of sound is notably finite
<Traneptora>
if you pull up the waveforms on a small scale they won't look close to each other at all because phase will be different
<Traneptora>
since sound has a positive wavelength, there's no guarantee that the compression of one corresponds to the compression of another and not the rarification or somewhere in between
<Traneptora>
you'd need identical distances from the source on a microscopic scale to guarantee that, which you don't have
<bcn_>
notably so! I just.... I think pros use one of those "action!" slapper things because it gives you a spike of sound on all mics, and a moment in video you can correlate. But I'm recording in a public place. I think doing that. Even clapping, would be obnoxious
<Traneptora>
that makes it easier when there's a momentary spike in amplitude, sure
<Traneptora>
if you can't create on you may have to find one naturally in the thing you're recording
<bcn_>
for instance, filming a public talk/town hall meeting, i might want to use a parabolic microphone to get better audio in a noisy room. I'd still record trash audio from my video camera becausae why not. and sometimes it will be good enuogh. but i might want to fade over the parabolic audio when its in focus and better
<Traneptora>
my instincts are that you have to do it manually. automatic detection of that is *HARD*
<dev-lizard>
anotherl : where's the git hash on there?
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<another|>
e.g. `ffmpeg version N-116781-gd89930f866` => g(it)<hash>
<another|>
so d89930f866
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<losth1ghway>
can anyone help me figure out why the following command is rendering a video 2sec longer than the corresponding audio file: https://pastebin.com/raw/XggJkHSM
<losth1ghway>
*input audio file
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