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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<echelon>
hi
<echelon>
i keep getting "not enough frames to estimate rate; consider increasing probesize"
<echelon>
it's supposed to just grab input from x11grab that's supposed to be 1 fps
<echelon>
i set -framerate 1, but i still get that message
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<echelon>
anyone have experience doing low latency MJPEG?
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<echelon>
why can't i find a single example use case of mjpeg output with ffmpeg
<CounterPillow>
-c:v mjpeg
<echelon>
CounterPillow: these are my flags.. -f x11grab -video_size 1000x1000 -draw_mouse 0 -i :1 -map 0:v:0 -c:v mjpeg -preset ultrafast -framerate 2 -r 2 -tune animation -f mjpeg main.mjpeg
<echelon>
it hasn't generated a single file, but ffmpeg is running
<CounterPillow>
do you really want to use raw mjpeg as the container? Seems like a footgun
<furq>
get rid of all of the options between -map 0:v:0 and -f mjpeg and move -framerate 2 before -i
<echelon>
including '-c:v mjpeg'?
<CounterPillow>
that can be inferred from -f mjpeg in this case
<echelon>
i made a mistake by setting the wrong path in the container, but i see the file now
<echelon>
it keeps growing in size and when i play it in vlc it speeds by really fast
<CounterPillow>
because it's a raw mjpeg stream, it contains no information as to the timing of the frames
<echelon>
is there away to make it not grow in size so that whenever a person opens the stream they only see the most recent frames?
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<echelon>
do i need to use ffserver?
<CounterPillow>
ffserver is dead
<echelon>
so what replaced it
<echelon>
and how do i "wrap" the video so that it remains at a certain size?
<CounterPillow>
Nothing, there are plenty of media streaming servers out there like mediamtx. The thing you're trying to do, feeding an mjpeg stream to presumably a browser that just thinks it's a regular jpeg as a livestream, is not really something to be done anymore in this day and age.
<echelon>
video.js seems to support it
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<CounterPillow>
if you're already requiring clientside JS to play back the stream then using a real streaming protocol like HLS or WebRTC seems like a strictly superior option
<echelon>
the problem with hls is latency, mjpeg seems ideal for near zero latency
<CounterPillow>
webrtc
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<echelon>
does ffmpeg support it?
<CounterPillow>
ffmpeg supports acting as a producer for a multimedia server (like mediamtx) that implements it
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<echelon>
does mediamtx support multiple streams or does each stream require its own process?
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<CounterPillow>
it supports multiple
<echelon>
> Serve multiple streams at once in separate paths
<echelon>
interesting, i wonder how i can leverage that over http
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<CounterPillow>
actually not sure if named pipes work, that just seems to do fd numbers
<echelon>
ok
<CounterPillow>
but writing to a pipe and then serving that to unsuspecting applications is generally not how livestreaming works
<echelon>
yeah :/
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<furq>
that is sort of how it works
<furq>
just not for browsers
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<furq>
how much latency is too much latency here
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<echelon>
furq: anything within a minute is good
<echelon>
i've tried hls, but with every tweak it's been between 3-10 latency
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<BtbN>
"within a minute" should be trivial, even with HLS
<BtbN>
just don't use 30 second segments or something
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<intrac>
hi, I'm using ffmpeg to stream to Twitch over rtmp. is there a way to make a stream end more gracefully, as just hitting 'q' on ffmpeg causes the stream on Twitch to pause, but still appears to show as live
<intrac>
I'm not sure if this is an ffmpeg issue or Twitch, as I've seen it on other streams. but I think possibly when other streams drop out, rather than end properly
<intrac>
is there some extra data that should be sent to let Twitch know the stream has finished?
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<intrac>
I'm also getting random ffmpeg errors: "av_interleaved_write_frame(): End of file", "error writing trailer of rtmp://live.twitch ..."
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<intrac>
my current test is to loop a short video and audio file from ffmpeg with -stream_loop
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<intrac>
"It is currently impossible not to send deleteStream, I suspect it is trivial not to send it but nobody so far felt the need or saw the usefulness."
<intrac>
so it looks like Ffmpeg sends that as default. maybe this is just Twitch not properly refreshing the web page
<intrac>
I think I've seen this with other people's streams too
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<intrac>
possibly the av_interleaved_write_frame(): End of file issue is caused by Ffmpeg being compiled with librtmp: https://stackoverflow.com/a/63729262
<intrac>
if Ffmpeg has been compiled with librtmp, is there any way to make it use its own version of rtmp instead of librtmp?
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<BtbN>
If you are still using librtmp, there's your issue right there.
<BtbN>
And no, that's a compile-time decision.
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<intrac>
BtbN: thanks for confirming
<BtbN>
We should probably remove the librtmp code
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<intrac>
BtbN: so that would force all builds to use Ffmpeg's own rtmp implementation?
<BtbN>
well, yes
<BtbN>
librtmp is just dead
<intrac>
ok. just wasn't sure how it fits together :)
<intrac>
is the rtmp code in ffmpeg a historic fork of librtmp, or entirely different?
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<JEEB>
intrac: it's separate
<JEEB>
and to my knowledge it works, the only major difference is that options are not passed as part of the URL, but instead as AVOptions which can be set on the command line
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<JEEB>
`ffmpeg -h protocol=rtmp` shows the options
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<intrac>
JEEB: ah ok. I'll try setting a few things like the rtmp buffer size and see if that helps at all
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<bpmedley>
echelon: Do you only need video in the browser?
<echelon>
yes
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<bpmedley>
I didn't see it mentioned. You can use WebCodecs to decode H.264 NALU sent to the browser, I believe. That coupled with sending the decoded frames to a canvas via a WebWorker might work.
<echelon>
is there a write up of how to do that somewhere?
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<bpmedley>
echelon: I can give you the javascript to decode the nalu, if you want. You'd need to find or code a ffmpeg + media server solution to send the nalu, I would think.
<echelon>
oh :/
<BtbN>
You aren't seriously suggesting to implement a decoder in JavaScript?
<echelon>
lol
<bpmedley>
It's like 30 lines of code, at most.
<BtbN>
For a video decoder?
<bpmedley>
Want to see it?
<echelon>
i'll have to hire somebody
<BtbN>
It's either not a video decoder or nonsense
<BtbN>
ah, WebCodec is just a decoder itself
<bpmedley>
Yes
<BtbN>
But why would you use that? Its advertised as "for video editors that need ultra low level access"
<bpmedley>
Decodes a annexb avc1.42C02A stream.
<BtbN>
for just playing a video, that seems horrible and will ruin most if not all hardware acceleration
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<bpmedley>
I've only tinkered with WebCodecs. I don't know many of the pros/cons of using WebCodecs as a viewer.
<BtbN>
The whole point of that API seems to be to export the raw decoded frame to JavaScript as an array of pixels
<BtbN>
That's automatically slow and resource intensive
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<CounterPillow>
I don't see why you'd suggest this in any capacity when mediamtx will literally make a webrtc js+html page for you
<bpmedley>
I've never used mediamtx
<CounterPillow>
then I don't know why you'd suggest this in any capacity when webrtc is literally the low-latency web streaming standard and does not need writing your own custom inefficient player with webcodecs