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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<soreau> I'm working toward streaming rtsp to mp4 container using nodejs to play the stream in the browser. I think I have the 206 response mostly hammered out so the file buffers until I kill ffmpeg, but it doesn't start playing immediately. Instead, it buffers like the video is loading, then gives a very short video that is not streaming, all the while I can see the server sending and browser inspector recieving the file
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<soreau> so I'm staring at all these ffmpeg movflags and wondering which I should use.. so far, I've found "rtphint+frag_keyframe+empty_moov+frag_custom" to give a short file and without rtphint flag, it buffers until I kill ffmpeg and then plays the entire video
<soreau> basically going from rtsp to mp4 on stdout and piping that as the responses to client requests
<soreau> the data is getting where it needs to go, I just wish I could put some metadata up front, like a custom moov that tells it the video length
<soreau> so then hopefully it would start streaming right away and keep playing while 'downloading'
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<soreau> and I don't think faststart applies here since that's a second pass and this is real time streaming without transcoding anything
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<welder> Hi, I've got a question do I lose on quality when I'm only cutting a part of mp4 but not using codec copy option?
<welder> so something like: ffmpeg -ss 1:30 -t 10 -i in.mp4 out.mp4
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<soreau> finally figured it out
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<Nemo_pi> 2
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<another|> welder: yes
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<Curid> Is this actually a good idea for small scale streaming? https://medium.com/@vlad.pbr/in-browser-live-video-using-fragmented-mp4-3aedb600a07e
<JEEB> I think someone made a proof-of-concept server that would feed fragmented mp4 directly to browsers, and you could feed it media from f.ex. ffmpeg
<JEEB> it's nice in the sense that it works with "just a video tag", but it is going to be actually live live
<JEEB> not something you can easily CDN
<Curid> would browsers complain about 100ms fragments?
<JEEB> why would they?
<Curid> idk, ll-hls has a minimum part duration of 300ms
<JEEB> that's the whole separate connections thing (or same connection but multiple requests)
<JEEB> while here it's "just" an actual live stream that is continuously being pushed to you
<Curid> is the browser smart enough to discard old video before running out of ram?
<JEEB> you'd have to ask whomever it was who made a PoC for this sort of stuff
<JEEB> but yea, the general idea of just feeding fmp4 as a live, ongoing http request is something that I've come up with and someone made a python PoC with, and I recall there's at least some company out there utilizing that technique. and now you've linked a medium post which with short amount of scrolling seems to be about the same thing.
<Curid> do you think it can beat the latency of ll-hls on iphone?
<JEEB> it's an actual live stream without the whole segment stuff so I don't see any reason why not? similar to webrtc on the UDP front.
<JEEB> of course actually trying to be low low latency is a case of "lol enjoy figuring out which bits are causing latency and where you can fight it easiest"
<JEEB> also I have no idea what your expectation with LL-HLS would be :D
<Curid> I get sub 2 seconds with a custom LL-HLS player, but it doesn't work in iphone.. https://codeberg.org/Curid/basic_ll-hls.js
<JEEB> and then figure out how much of that is the protocol etc
<JEEB> as opposed to things that would not change with the encoding, capture, whatever
<Curid> the problem is that i don't have an iphone lol
<JEEB> uhh, bad wording but basically trying to say "how much of your latency would change if the only thing you switch is the way you push your encoded result"
<JEEB> since 2 seconds is a lot for anything saying it's supposedly low latency
<JEEB> but yea, welcome to the deep hole of latency optimization
<JanC> if you want low latency just use a non-HTTP dedicated streaming protocol :)
<Curid> webrtc requires a third party server and has limited hardware acceleration
<JEEB> I'd say just having a live live stream already minimizes various bits of the latency :D
<JEEB> instead of the client having to request segments
<JEEB> webrtc has the same formats available underneath, no? everything that's specified in RTP as that is the media protocol
<JEEB> so you get your H.264 and friends
<Curid> i read somewhere that the implementation is immature and hwaccel is limited
<JEEB> dunno, I would not see how decoding X format is any different whether it's received from mp4, via the media decoding APIs that are utilized for JS based playback or webrtc
<furq> i think some h264 profiles don't work everywhere over webrtc
<furq> last i checked chrome wouldn't decode high profile over webrtc
<furq> but that was years ago
<JEEB> given that they bundle FFmpeg for swdec and also have hwdec wrappers that's a golfclap.jpg
<JEEB> unless they specifically hard-code openh264 for webrtc for some god-forsaken reason :)
<JEEB> in which case it has now for a year or two - if not even more - supported high profile
<furq> it's amazing how much of a mess this all still is years later
<furq> i think the only thing that's changed is that a bunch more competing specs exist now
<furq> who would have thought it
<Curid> this is the new thing https://quic.video/blog
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<furq> whoever designed this website is a true expert at making me want to close the tab
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<Unit640> In a desperate attempt to save storage space on my overflowing disks, I've written a script which is supposed to go through any new PNG image files entering my system and re-encode them with the maximum amount of compression. It's good that I didn't blindly trust this, but rather have the script compare how big the file is before and after, and only go through with it if it's actually smaller. Shockingly, in almost every case, the ffmpeg command causes
<Unit640> the PNG image to become MUCH LARGER than before, sometimes 4 MB extra! Am I doing something fundamentally wrong? Example commmand I use: `ffmpeg -loglevel quiet -y -i '/path/to/in.png' -compression_level 100 '/path/to/out.png'` What tool could they possibly be using which encodes so much more efficiently than ffmpeg?
<sonicrules1234> I suggest using jxl
<Unit640> (When asking ChatGPT about this, it told me to use pngquant, which turned out to be *lossy*.)
<Unit640> Looking into jxl...
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<furq> add -pred mixed if you want to use png
<furq> but really just use lossless webp or jxl or something
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<Unit640> sonicrules1234: Everything related to "jxl" mentions JPEG but none mention "PNG"...
<Unit640> Wait... jxl is an image format?
<furq> yes
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<Unit640> I thought it was a converter.
<sonicrules1234> Yeah
<Unit640> Never heard of jxl until now.
<furq> it supports lossless rgb
<sonicrules1234> It's a format
<Unit640> "add -pred mixed if you want to use png" <-- What do you mean by that?
<furq> it's not very well supported yet
<Unit640> If at all possible, I'd like to stick with PNG.
<Unit640> But I'm not married to the format.
<Curid> would encoding an already lossy encoded png image losslessly increase the size?
<furq> what is a lossy encoded png image
<Curid> nvm
<BtbN> depending on what the png image is, jpeg might be less efficient than png
<BtbN> like, one big solid color or something extreme like that
<furq> Unit640: ffmpeg -h encoder=png
<furq> but to actually make png small you will probably want to use a dedicated png optimiser
<furq> which will take forever to encode and you'll end up with bigger images than if you'd used webp or jxl
<furq> the only benefit of png is universal support
<furq> which is not that big a benefit when it's lossless and you can convert it as you need to
<furq> it is pretty annoying that -compression_level for png doesn't set -pred
<furq> i'm not sure what it even does
<furq> -pred none usually gives disappointing results
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<sonicrules1234> Unit640: What OS are you on?
<furq> apparently it just passes compression_level to zlib deflateInit
<furq> so that's not much use
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<tmatth> i'm trying to cut a video with output seeking (https://trac.ffmpeg.org/wiki/Seeking#Output), so instead of `-ss start_time -i video.mp4` and `-ss start_time -i captions.vtt`, i'm dropping both of those ss options and having a `-ss start_time` to the right of those
<tmatth> unfortunately this seems to only be cutting the video and the subtitles are not lining up where i'd expect
<tmatth> am i missing something?
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<Unit640> sonicrules1234: Linux.
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<Unit640> furq: I'm confused about the two different things you've told me about, but will definitely try them out tomorrow (late now).
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