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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<noobaroo>
I'm trying to test out theora and see how it fares, just to see. I've never used it before. At https://github.com/xiph/theora/blob/master/examples/encoder_example.c I can see it says ` -z --speed <n>` is an option and I'm able to talk to the libtheora encoder via ffmpleg by passing `-flags ___` but I don't know the syntax beyond this.
<noobaroo>
I tried `-flags speed=1` and `-flags z1` and even tried quoting it in case it's passed directly raw without parsing `-flags "--speed 1"` It keeps saying Stuff like [libtheora @ 0x55c6714ec500] Unable to parse option value "z=0"
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<EmberCrest>
Howdy, ladies and gents
<EmberCrest>
I've got two episodes of Star Trek ;) TNG, and DS9. DS9 has AC3 audio encoding, 48khz sample rate. TNG has AAC, also 48khz.
<EmberCrest>
TNG encoding to RTSP using libopus is absolutely attrocious, choppy and hollow-sounding.. but same settings for DS9 are perfectly fine. I've got a pretty powerful CPU-- adjusting compression and cpu-used flags does virtually nothing to help the TNG output improve..
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<ePirat>
EmberCrest, how many channels do each have respectively?
<ePirat>
also did you try if the issue is the decoding or encoding, by playing it with ffplay for example instead of encoding it to opus?
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<EmberCrest>
ePirat: I'm not much of an FFwhiz:) I'll see what I can find using FFPlay
<EmberCrest>
ePirat: TNG is stereo, DS9 is 5.1 (I believe.. need to check again to be sure)
<EmberCrest>
ffplay renders the audio just fine-- in my previous infrastructure, I was using RTMP which supports AAC. That worked fine.
<EmberCrest>
but now I have to use MediaMTX for WebRTC, which doesn't support AAC.
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<FH_thecat>
I have recorded video with my phone, where I held the phone in "landscape" position while recording. So the resulting video is rotated 90 degrees, ie in portrait mode
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<EmberCrest>
I uploaded a sample of what's occurring in my above issue: choppy audio when re-encoding AAC to Opus over RTSP. http://www.sndup.net/bg556
<EmberCrest>
tried with vbr on/off and with/without specified bitrate 96k.
<EmberCrest>
might go try another AAC track when I have time today...
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<Wulf>
Hello... So, I've got two versions of the same video. One is 4:3 format in low res, the other got letterboxed (bottom cut off) and high res. There may also be some differences in timestamps. How can I combine the videos to get the high-res with the low-res bottom added?
<Unit640>
Wulf: I can't help, but I know the feeling of wanting to combine two crappy sources' best parts to make one decent version. I've done it manually by using a video editor, but I can't imagine that ffmpeg would be able to tell what you really want to do with them.
<Unit640>
(Or, if it can, the amount of work to tell it the details would take longer to figure out than doing it manually.)
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<Wulf>
okay... the videos run at slightly different speed too. Guess to put them together it would require a tool that was specially made for this :)
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<StephenLynx>
Wulf, so you want the high res video. why you want the bottom of the low rest btw?
<StephenLynx>
you want to do that effect that people use where they have a blurred version of the video filling up space to fit the format?
<Wulf>
StephenLynx: the high-res is missing the bottom. So I could take the bottom from the low-res to get the complete picture (pun intended.).
<Wulf>
StephenLynx: it's not a blur with useless information, it would contain actual video, although in blurry quality.
<StephenLynx>
ah, I see.
<StephenLynx>
those guys must have REALLY loved letterboxes then.
<StephenLynx>
to cut off the bottom just to put them in.
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<EmberCrest>
My AAC -> Opus issue is *solved*! -async 50 has completely fixed the audio problems.
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<Unit640>
Wulf: In a good video editor (meaning costs money), one would just start by adding the hi-res one as the base, then put the bad (but with more aspect ratio) underneath it and then "drag it out" to match the same duration.
<Unit640>
Sadly, all video editors I've tried for Linux have been horrific, or won't even run at all.
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<Marth64>
You will need to map it so ffmpeg knows to carry it
<Marth64>
Try -map 0 before your -c:a
<StephenLynx>
in my experience
<StephenLynx>
you don't need anything special but setting frames to 1 dontdieych2
<StephenLynx>
wait, i might have misunderstood what you want.
<Marth64>
might need a -c:v copy as well
<StephenLynx>
nope.
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<StephenLynx>
input is audio, output is jpg
<StephenLynx>
frames to 1.
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<StephenLynx>
you can do it just like you would thumbnail a video for the first frame.
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<StephenLynx>
ffmpeg -i {$path} -y -vframes 1 -vf scale= is the command I use. then scale can either be -1:-1 or you could completely omit it i think. or you could set values if you want to resize it.
<Marth64>
that unnecessarily encodes the art, user can copy directly by mapping the attached_pic stream
<StephenLynx>
well, I might have misunderstood what they wanted after all.
<dontdieych2>
without `-map 0` ffmpeg ignores album art video stream
<StephenLynx>
probably it is telling to map that stream.
<StephenLynx>
im not too savy on the command line to be able to tell.
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<dontdieych2>
looks like that
<dontdieych2>
btw, ffmpeg and manul is quite rabbit hole.
<dontdieych2>
enjoying it lol
<dontdieych2>
manul/manual
<StephenLynx>
yeah, it is a very complex software. using the libraries is even harder.
<dontdieych2>
looks like whole world's media things are depending on ffmpeg
<StephenLynx>
btw Marth64 did you had the time to look up the weird issue I had?
<StephenLynx>
dontdieych2, yeah, no one would reinvent a wheel that takes so much labour to produce.
<StephenLynx>
is like linux.
<StephenLynx>
being written in C probably helps fitting it in every case.
<StephenLynx>
despite making the libraries harder to use.
<Marth64>
StephenLynx: the gif encoder?
<Marth64>
sorry I don't recall. I've been tied up at dayjob
<StephenLynx>
sec, let me paste it
<StephenLynx>
when I moved from av_image_alloc to av_frame_get_buffer, my gif stopped being animated. even the filesize is shorter. https://pastebin.com/R7hDiMc4 the whole code. I prepare the frame at line 165.
<BtbN>
I somehow doubt just changing between those functions has any impact on that.
<StephenLynx>
literally the only difference.
<BtbN>
You'll need something more minimal to reproduce it then
<StephenLynx>
i dont think I can make it more minimal. gif comes in, gif comes out.
<BtbN>
Working through those 300 lines is gonna take significant time, and the inconsistent indentation doesn't help either.
<StephenLynx>
i can reindent it.
<BtbN>
It's very unlikely for there to be a bug in ffmpeg in that regard, so best advice still is to refer to the documentation, headers and the ffmpeg code itself
<StephenLynx>
btw, I'm using 4.2.
<BtbN>
Well, then we can't really help you here anyway, that's as dead as it gets
<BtbN>
test with latest master first
<Marth64>
++1 BtbN advice. also I see a lot of return paths with no logging. it might be helpful to add some logs until you get to the root of why it is stopping
<StephenLynx>
it didnt stop.
<StephenLynx>
actually.
<StephenLynx>
let me test something.
<StephenLynx>
im assuming it didn't stop, but I didn't test that.
<Marth64>
you are in a while(true) so surprises can occur
<StephenLynx>
yup, it finishes.
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<StephenLynx>
it doesn't crash, segfault or errors out.