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<Osmcorp> Hi everyone
<Osmcorp> I am receiving a multicast UDP TS H264 from an IRD and then I transcode it with FFMPEG to HEVC and output to /dev/null. I don't understand why I keep having these warning logs from time to time.
<Osmcorp> [h264 @ 0x555c3dd0e300] [debug] nal_unit_type: 6(SEI), nal_ref_idc: 0
<Osmcorp> [warning] udp://230.10.2.65:1234?localaddr=10.20.12.28&overrun_nonfatal=0&fifo_size=10000000&buffer_size=134217728&pkt_size=1316: corrupt decoded frame in stream 0
<Osmcorp> [info] decoder -> ist_index:0 type:video frame_pts:3250369 frame_pts_time:36.1152 best_effort_ts:3250369 best_effort_ts_time:36.1152 keyframe:0 frame_type:2 time_base:1/90000
<Osmcorp> [h264 @ 0x555c3dd0e300] [debug] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 1
<Osmcorp> [h264 @ 0x555c3dd0e300] [debug] ct_type:0 pic_struct:3
<Osmcorp> the method h264_select_output_frame is currently setting the flag AV_FRAME_FLAG_CORRUPT
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<Osmcorp> This is my input stream info
<Osmcorp>     Last message repeated 1 times
<Osmcorp> [h264 @ 0x5641e8c6b700] decode_slice_header error
<Osmcorp> [h264 @ 0x5641e8c6b700] no frame!
<Osmcorp> [h264 @ 0x5641e8c6b700] mmco: unref short failure
<Osmcorp> [h264 @ 0x5641e8c6b700] number of reference frames (0+4) exceeds max (3; probably corrupt input), discarding one
<Osmcorp> Input #0, mpegts, from 'udp://230.10.2.65:1234?localaddress=10.20.12.28':
<Osmcorp>   Duration: N/A, start: 69044.878400, bitrate: N/A
<Osmcorp>   Program 212
<Osmcorp>     Metadata:
<Osmcorp>       service_name : C I
<Osmcorp>       service_provider: OSN
<Osmcorp>     Stream #0:0[0xd4]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt470bg, top first), 720x576 [SAR 16:11 DAR 20:11], 25 fps, 25 tbr, 90k tbn, 50 tbc
<Osmcorp>     Stream #0:1[0x138](eng): Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16p, 128 kb/s
<CounterPillow> thanks for flooding the channel
<Osmcorp> Sorry about that
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<Osmcorp> I think it is a bit hard to understand my issue without having the context.
<CounterPillow> Then use a fucking paste site you dipshit
<Osmcorp> but again sorry for flooding the channel
<Osmcorp> CounterPillow You Forgot to get your polite pills this morning buddy
<ecapi> YOU RUINED THE FUCKING DAT Osmcorp
<ecapi> I AM SO ANGRY NOW I WANT TO KILL
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<Cindy> wtf is wrong with both of you
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<vlt> onla: Thanks for providing feedback and a solution. Just curious: Did you also try to replace the `done < "$subfile"` construct with something more robust?
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<onla> nope. One of the guys simplified my script just this way https://termbin.com/fxpwj
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<smandelli> Dear all,I have a very nasty problem: I have a non-IDR h264 stream, and I want to decode that with ffmpeg. How do I have to set the CodecContext in order to set that I'm working with a non-idr stream?
<smandelli> Thx in advance <3
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<BtbN> If you mean a broken stream with the extradata missing... I don't think that can be decoded
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<smandelli> ffplay reproduces the stream well, just a log complaining: nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
<smandelli> In my application, at this point .... IDK how to read the SPS and PPS contained in the extradata .... think so
<BtbN> it just won't decode the beginning until the next IDR frame comes along
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<smandelli> BtbN, yes right. Anyway, I think that I have some problem to extract the extradata. How can I copy the extradata into the data buffer?
<BtbN> the demuxer does that for you typically
<BtbN> Or the decoder does it itself, it it's embedded
<smandelli> I have a super custom solution. I set the CodecContext with the codec and other few informations, and then I send the packets and receive frames.
<chiselfuse> i just recorded a video with my phone, it's 13 minutes long but for some reason 1.6 GB in size. how do i go about figuring out why it's so big? i ran ffprobe on it but i don't know what to look at https://0x0.st/Hy5c.txt
<smandelli> I am wondering if I have to specify something the the codeccontext
<BtbN> If there is no demuxer, you need to get the extradata from "somewhere", and set it before you start decoding
<BtbN> Or make sure it's embedded into the stream
<BtbN> chiselfuse: phones encoders are typically quite bad. So if you want good quality, the solution is "more bitrate"
<BtbN> 17Mbit/s for a 1080p live recording isn't even that excessive
<chiselfuse> BtbN: when encoders are bad they compress the stream badly, as far as i understood. but isn't bitrate something we calculate for the uncompressed stream? what does it have to do with how bad the encoder is?
<BtbN> The worse it is, the more bitrate it needs to achieve passable quality.
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<chiselfuse> BtbN: if i record the same video and then encode it twice, once well and once bad. do we talk about how high bitrate is *after* it's been encoded or *before* it's been encoded?
<BtbN> I don't follow.
<chiselfuse> bitrate is how many bits per second in order to represent the stream, right?
<BtbN> yes
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<chiselfuse> BtbN: if a camera caputres frames, *before* it encodes them, can we measure the "bitrate" or is the bitrate something that we measure after it's been compressed into a stream?
<BtbN> an uncompressed stream also has a bitrate, sure. A quite large one
<BtbN> But I don't understand what you're getting at.
<BtbN> The better the encoder, the more quality it can squeeze out of a given bitrate, or the lower a bitrate it can achieve for a given quality.
<chiselfuse> okay i think i got it
<chiselfuse> is there a way to decode and re-encode this stream into a better compressed one?
<BtbN> yes, but quality only ever goes down, never up
<BtbN> every time you do a lossy re-encode, the quality get worse
<BtbN> But really, 17Mbit/s is not that unreasonable for 1080p in h264
<chiselfuse> of course, but i assume it can be done negligbly
<BtbN> I wouldn't be so sure, depending on what the video is.
<chiselfuse> BtbN: but i have 2 hour long films 1080p that are ~2 GB
<BtbN> they are either incredibly low-complexity then, or the quality is poor
<BtbN> 1080p in h264 is something like 8~15Mbit/s for decent quality and "average" levels of complexity
<chiselfuse> "A 1080p HD 60 fps 2-hour movie averages 6 GB in file size. A 1080p HD 30 fps 2-hour movie averages 3 GB in file size."
<chiselfuse> found this on internet
<BtbN> Sounds like nonsense. Halving the FPS does not result in having the filesize
<BtbN> *halving
<chiselfuse> hmm, i still think that that file size is incredibly high
<BtbN> It's a bit high, but not unreasonably so.
<BtbN> For some shoddy hardware encoder in a phone, it's decent though
<BtbN> assuming it actually looks good
<chiselfuse> BtbN: https://youtu.be/WUvTyaaNkzM this video after i downloaded it at highest quality is 42 MB
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<BtbN> so, it's not h264, not encoded by a phones hardware encoder, and also encoded by youtube, who don't care about quality
<chiselfuse> you can do it yourself with `yt-dlp WUvTyaaNkzM` to test
<BtbN> plus, the video is largely solid black, so compresses _super_ well
<chiselfuse> i must have bad eyes then idk
<BtbN> there is almost no motion, just occasional "slide swaps"
<BtbN> Of course it's tiny
<chiselfuse> regardless, i can't afford to have 1.64 GB for 13 minutes, what could i do?
<BtbN> reduce the quality
<BtbN> or the resolution
<chiselfuse> -crf ?
<BtbN> hm?
<BtbN> On your phone I was thinking
<chiselfuse> oh, but i like the video and i don't want to record it again
<chiselfuse> i was thinking decoding and reencoding
<BtbN> You can do that, yeah. libx264 (if it has to be h264) with a slow preset, and then play around with crf values until you're happy with the result
<chiselfuse> BtbN: it doesn't have to be any specific container/codec, which one could i go with for better result?
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<BtbN> well, av1 is probably the best you can get right now. But encoding will be _slow_
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<BtbN> If you want an "in between", hevc via libx265 is also a good option
<BtbN> I'm not sure which AV1 encoder is currently the best one
<chiselfuse> ffmpeg -i systematic-rule-application.mp4 -c av1 out.mp4
<chiselfuse> does this look good?
<BtbN> no
<BtbN> that will for one try to encode the audio with AV1, which won't go well :D
<BtbN> And then you are not specifying a specific encoder, so it will randomly pick one.
<chiselfuse> hmm, what codec should i pick for audio, or just copy it?
<BtbN> And you are giving it no settings whatsoever, and the defaults are typically poor
<BtbN> just copy the audio really
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<BtbN> its size is insignificant compared to the video
<BtbN> and "quality can only go down" applies to it as well
<chiselfuse> BtbN: not that i care in particular but how would `-c:a copy` make quality go down? it won't even decode it as far as i understand
<BtbN> hm? it literally copies it
<BtbN> unless something is broken, it will be identical
<chiselfuse> so quality can only go down also applies to when it copies it and is identical?
<BtbN> I don't follow.
<BtbN> You copy it especially to _not_ have quality go down.
<chiselfuse> < BtbN> its size is insignificant compared to the video < BtbN> and "quality can only go down" applies to it as well
<BtbN> Yes, hence you copy it
<chiselfuse> assuming you were talking about audio with "quality can only go down" applies to it too
<BtbN> Yes, audio encoders work the same way
<chiselfuse> okay so when i copy audio it won't affect quality, right?
<BtbN> of course not, it copies the input
<chiselfuse> okay i misunderstood earlier then
<chiselfuse> BtbN: i'm looking here https://trac.ffmpeg.org/wiki/Encode/AV1 and it looks to me like `-c {codec}` "codec" is really the name of the encoder and it implies which codec is by that too, right?
<BtbN> Yeah. you can put a codec name, but you won't know what encoder you get then, and thus can't configure it
<chiselfuse> ffmpeg -i systematic-rule-application.mp4 -c:a copy -c:v libaom-av1 out.mp4
<chiselfuse> how bad is this?
<BtbN> However bad the defaults of libaom are
<chiselfuse> heh why are the defaults not sane
<BtbN> they're just some default. They can't predict what you want them to do
<chiselfuse> is there a sheet somewhere with good "defaults" to override with?
<BtbN> not really. It's always highly dependent on the input and desired results
<BtbN> you linked the page that explains the parameters just a few minutes ago
<BtbN> constant quality mode is probably what you want
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<chiselfuse> do i limit the bitrate?
<BtbN> I don't see a reason why you would
<chiselfuse> ffmpeg -i systematic-rule-application.mp4 -c:a copy -c:v libaom-av1 -crf 30 out.mp4
<BtbN> you just need to find a crf value that fits your desired quality vs. file size tradeoff
<BtbN> 30 seems a bit high to me, but I have no idea about the crf scale on libaom
<chiselfuse> unless there's a way to calculate the crf, i guess i just have to guess it by trial and error, right?
<BtbN> Normally 10 and lower is considered visually lossless, but it's an arbitrary scale
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<BtbN> You might also want to pick a preset, but again, no clue about those on libaom/av1 encoders in general
<chiselfuse> dw I'll just blame you since you mentioned av1 to begin with :P
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<sonicrules1234> chiselfuse: I usually use ab-av1 for that
<sonicrules1234> It does a binary search with crf values and uses vmaf to tell how much quality they have
<sonicrules1234> The eta indicator library is broken right now, so don't worry if your eta skyrockets
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<TrillionEuroNote> Is anyone familiar with the timed_id3 metadata stream? I'm seeing it in an MKV and i'm wondering if i should worry about preserving it when copying to another file/container
<TrillionEuroNote> it's video data saved from an online video stream, so i was concerned it might contain data about what point in the video stream my capture started from
<TrillionEuroNote> But it does not appear that ffmpeg can copy it - ffmpeg reports that it only supports audio, video subtitles
<sonicrules1234> I think that's some proprietary data that comes in when recording commercial streams
<Lynne> it usually is either nothing or ad insertion timing
<TrillionEuroNote> Thanks
<furq> there's a recent-ish patch that adds support for copying it but idk what containers it'll work in
<furq> but you can probably just ignore it anyway
<TrillionEuroNote> So it's probably not what i thought it was then right? the only reason i'd care about it, would be if it marked the point in time that i began to capture the stream
<furq> i guess that would be up to whatever you used to capture it
<furq> it seems unlikely though
<TrillionEuroNote> hmm true
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<TrillionEuroNote> looks like the information i'm looking for is in what appears to be metadata, if i do -show_streams -print_format json, under the "start_time" key
<TrillionEuroNote> BBL
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<Flaesh> is it possible to take 2 rtmp streams (same res and framerate) and make one go out to a rtmp stream, but be able to change which one without stopping the stream? obs seems overkill for the job. especially cool if each input could have its own filters (e.g. maybe overlay an image and/or add a delay to one)
<Flaesh> if FFMPEG can't do this, but you know software that can, please refer me to that. must be free and work on linux
<BtbN> that's literally what OBS was designed for
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<Flaesh> OBS can do a lot more, i just want to be able to switch which stream goes to an output
<sonicrules1234> You also said you wanted different filters
<sonicrules1234> Like overlays or adding delays
<sonicrules1234> Literally what OBS does
<Flaesh> that was an optional extra
<Flaesh> if need be i can apply those with another instance of ffmpeg, just OBS is quite resource-heavy and running 3 instances of it is quite a task for my pc
<sonicrules1234> I don't think you need 3 instances of OBS for that
<Flaesh> the other 2 are doing other things
<Flaesh> what i want this for is to switch between which of the two goes to output
<sonicrules1234> Pretty sure you only need 1 instance for most things
<sonicrules1234> Aren't they called scenes in obs or something?
<sonicrules1234> You can even add an animation when switching between them
<Flaesh> yes, i know how OBS works
<Flaesh> the 2 instances are doing something different on different scene collections
<Flaesh> and i just want the simplest way of changing which one goes to output, which i dont believe OBS to be
<sonicrules1234> I don't think you could do that with ffmpeg alone, you'd need some sort of wrapper
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<Flaesh> a wrapper is fine, just OBS is quite resource-heavy and defo feels overkill
<BtbN> What you describe is quite exactly OBS scenes
<BtbN> OBS is not overly resource heavy. It just uses the GPU instead of the CPU
<BtbN> The majority of the load is the encoder anyway, and OBS and ffmpeg use the same ones
<Flaesh> OBS is quite a complicated piece of software, i feel very complicated just to be used to cut between 2 different rtmp streams
<Flaesh> i would much prefer minimize the number of OBS instances i use - it makes things more complicated for me to manage and will still have heavier resource usage
<BtbN> ffmpeg doing that would likely be more resource intensive
<BtbN> scene composition like that purely on CPU is not exactly light
<Flaesh> where would it be compositing? isnt it just switching between 2 streams?
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<BtbN> to do that reliably, it'd have to constantly decode both, and "render" them onto some output
<BtbN> I doubt you can live-switch without re-encoding
<BtbN> At least not without a lot of custom code that re-writes timestamps on the fly
<Flaesh> re-encoding is fine but surely that'dn't be compositing
<BtbN> something has to compose the image the encoder gets
<BtbN> If it just switches the entire stream, you get the timestamp-issue again
<BtbN> so you'd generate a blank base-stream, use that as clock source, and render the desired input on top of it
<BtbN> At least I can't think of a more elegant way to do the whole thing reliably with pure ffmpeg.c
<BtbN> And if one of the two streams goes down for even a bit, I don't think that's recoverable either
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<Flaesh> huh, interesting. i expected a simple solution compared to OBS (as OBS has a lot of features which are really quite unnecessary for the job) but i guess not. thanks for the help
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<BtbN> The whole thing also relies on there being some filter that even can switch between two inputs at all
<BtbN> I'm not sure if that exists
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<chilinux> Hello
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