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
<iconoclasthero>
https://bpa.st/QFQPY i was comparing the ffmpeg that i just compiled with the one from the ubuntu oracular repo and there are a bunch of options enabled in the latter that aren;t in the former. I'm wondering if that's because the repo version has a lot of redundant detaults enabled or ...?
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<ePirat>
iconoclasthero, its because distro packaging needs to be explicit about the enabled dependencies
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<ePirat>
for a lot of dependencies, ffmpeg's configure will auto-detect them but for distro packaging you do not want to accidentally miss a depepdency that you actually expected to be enabled. So enabling them explicitly makes sure configure will error if they are not found.
<iconoclasthero>
if not, i'll look through the ubuntu list and go from there.
<iconoclasthero>
on 176 I have a list element commented out with `#--enable-vulkan disabled` which isn't really a standard bash thing but it's the only way to comment something out of a list of items where the lines are termianted with ` \`
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<soreau>
When I try to stream ffmpeg with an existing nginx instance with this command `ffmpeg -re -i file.mp4 -c:v libx264 -c:a aac -f flv rtmp://127.0.0.1/feed/stream` I get this output:
<soreau>
It seems it's trying to open the stream link as a file
<soreau>
is there a way to tell it that it's a stream and not a file?
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<kmikita>
"-c:v libxvid -qscale:v 2 -maxrate 4854000 -bufsize 3145728" I have exceed maxrate when using libxvid (native mpeg4 well done whith qscale+maxrate). Can I control maxrate with xvid?
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<kmikita>
"qmin" and "qmax" ignored by libxvid?
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<kmikita>
find solution "-c:v libxvid -qmin 2 -qmax 5 -b:v 3500k -maxrate 4854000 -bufsize 3145728" lets to hold qmin quantizer in the bitrate limits
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<vlt>
Hello. Is there a better way to get the output of `-filter:a astats=metadata=1:reset=1,ametadata=print:key=lavfi.astats.Overall.RMS_level` than piping everything (including stream layout info) into a file and trying to grep every "[Parsed_ametadata_1 ...] frame:..." string?
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<vlt>
The re expression, in case anyone is interested :D r"(\[Parsed_ametadata_1 @ .+\]) frame:(?P<frame>\d+) +pts:(?P<pts>\d+) +pts_time:(?P<pts_time>[\d.]+)\s*\1 lavfi.astats.Overall.RMS_level=(?P<rms>-?(?:[\d.]+|inf))"
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<soreau>
I think everyone here has asked a question at some point in history, and they are still idling for an answer to this day
<iconoclasthero>
soreau, where is stream coming from?
<furq>
vlt: ametadata=file=-
<soreau>
iconoclasthero: in my command, it's file.mp4
<soreau>
this is not a publish command in the config, I am trying to run this manually from terminal
<jprjr>
change /feed/stream to /live/something-or-other
<soreau>
oh fuck you
<soreau>
how does that work?????????
<soreau>
god damn it
<jprjr>
In your nginx config you named the application "live"
<soreau>
well well
<JEEB>
FFmpeg sucks, right? :P
<jprjr>
ffmpeg parses the rtmp url as rtmp://(host)/(application)/(stream)
<soreau>
heh
<soreau>
jprjr: thank you!!!
<soreau>
TIL...
<jprjr>
Or well lemme double-check my rtmp terminology here
<jprjr>
Where I put (stream) a lot of apps refer to as a stream key
<jprjr>
ok yeah I think RTMP, or at least, the nginx rtmp module refers to that last part of the URL as the stream so yeah, /(app)/(stream)
<iconoclasthero>
wait, so the solution was to change the suffix of the stream address?
<soreau>
well it runs to stream to rtmp but I cannot play the stream sadly
<soreau>
iconoclasthero: no
<soreau>
iconoclasthero: the identifier
<iconoclasthero>
oh, i never got the nginx rtmp module working and give up on it.
<iconoclasthero>
so i'm curious how this ends up.
<soreau>
probably because of this problem :P
<soreau>
I think now I just need to curb the rate or so
<jprjr>
What's more accurately iconoclasthero the fix was to make sure the application part of the RTMP url matched the application defined in the nginx config
<iconoclasthero>
possibly. i was running audio only and satisficed on mediamtx until the patch for libzmq came through this weekend.
<iconoclasthero>
ah.
<jprjr>
ie, if you write "application puppies {" in nginx, your RTMP url's path needs to begin /puppies
<iconoclasthero>
so in my case i might want to use application mpd
<soreau>
ok now I have it working
<soreau>
just needed to s/h264/flv/
<soreau>
iconoclasthero: so if you want to try again, now is the time :P
<jprjr>
Typically I make the application match whatever the conceptual application is
<soreau>
jprjr: thank you for pointing that out, I thought it was standard nginx config directive
<soreau>
had no idea it meant 'you have to stream here'
<soreau>
and thanks to everyone that pointed to the problem, even though it wasn't ffmpeg
<jprjr>
like "ingest" for incoming streams, then you could do say, "720p" for transcodes to 720p
<jprjr>
np, I wrote a whole app around ffmpeg and the nginx-rtmp-module a few years ago
<soreau>
I figure you guys probably get a lot of 'oh I have problem xyz and ffmpeg does not work, so it must be ffmpeg's fault'
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<soreau>
jprjr: nice
<iconoclasthero>
soreau, just to be clear what you're doing, you're trying to stream a file via rtmp to nginx's rtmp module so you can access it from outside via your RP as opposed to opening another port?
<soreau>
no
<soreau>
that was just the bait
<soreau>
now I will tell you the switch
<JEEB>
anyways, one thing seems to be a thing in general with RTMP
* iconoclasthero
waitng with bated breath
<JEEB>
the servers never give any info to the client
<soreau>
I am using wf-recorder to stream (wlroots) desktop to rtmp
<JEEB>
it's kinda OK with servers you host and where you can stare at relevant logs
<JEEB>
(if the server even logs disconnects etc)
<JEEB>
but lol 3rd party services
<iconoclasthero>
well, in my case i'd be using some sort of stream from mpd.
<iconoclasthero>
probably start with the named pipe at /tmp/mpd.fifo
<jprjr>
If you have your MPD configured to support pipe output (should be default on Unix/Linux) I tend to use that over fifo
<jprjr>
Main benefit is (last I checked) MPD will wait for your app to process the audio when you use a pipe, you can get blips in the fifo if you get out of sync
<iconoclasthero>
i'm not 100% on my terminology here, i thought that using output type "fifo" path "/tmp/mpd.fifo" is called a named pipe
<iconoclasthero>
while output type "pipe" was an unnamed pipe.
<jprjr>
So I like to configure a pipe output and have it run a script that execs into ffmpeg. When I need to change something, edit the script, toggle the output
<jprjr>
Let me double-check my MPD terminology here, you're right that a fifo is technically a named pipe
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<iconoclasthero>
b/c ffmpeg libzmq is working with the fifo but not with a direct [unnamed] pipe where mpd calls command: "ffmpeg -hide_banner -f s16le -ar 48000 -ac 2 -i - -f mpegts zmq:tcp://127.0.0.1:5555 >>
<iconoclasthero>
/var/log/mpd/ffmpeg.log 2>&1"
<jprjr>
ok yeah s, in MPD, a fifo output is a named pipe
<jprjr>
But there's also MPD's "pipe" output which is really a subprocess
<iconoclasthero>
right, and i've got that working with rtsp with mpd calling ffmpeg internally and then it goes to mediamtx and then i pick that up from outside.
<jprjr>
I do something similar - MPD calls ffmpeg, but I just do mpegts over udp to mediamtx
<iconoclasthero>
that was actually leads into one of my next questions: what formats can mpegts use?
<jprjr>
for audio - nearly all the AAC variants (unsure about xHE-AAC, I think that would require LATM instead of ADTS muxing), AC3/EAC3, MP3, Opus, tons of others
<jprjr>
But for the purposes of going into MediaMTX and presumably out as HLS - AAC, AC3/EAC3, MP3, Opus
<jprjr>
The mediamtx readme lists the codecs it accepts with udp+mpegts
<vlt>
furq: Thanks!
<Kaybi__2>
Damn there is a lot of action right now =D
<Kaybi__2>
I've had some conversations with intrac tonight and we didn't manage yet to find a solution to my issue, would someone want to try to help me as well? O:3
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<iconoclasthero>
so briefly, i had tried libzmq because someone here said they used it but when i compiled ffmpeg with it, it ended up broken, patch came through sunday or monday and i've been using that w/o mediamtx since.
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<iconoclasthero>
i've got autossh opening a tunnel and then just catching the zmq from localhost
<iconoclasthero>
on some level I wanted to try to push as much traffic through nginx as possible, but even with the theoretical ability to use the tcp stream module, it seems like a lot of hastle to save an open port.
<iconoclasthero>
(which, using an autossh tunnel does anyway)
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<iconoclasthero>
but for the purposes of this discussion, if the mpeg-ts container can hold opus then I'm happy. i just noticed this morning that mpd was saying...
<Kaybi__2>
The segment generated is not the same that the upper command generated (duration is 6s vs expected 3s) as if my keyframe was not being used..
<Kaybi__2>
When I analyze packets through ffprobe on these two different segments I can see that the good one only presents one heading keyframe while the bad segment contains many keyframes
<Kaybi__2>
So the question is: How can I create a single segment using the exact same keyframes that ffmpeg is using when generating the whole playlist without transcoding?
<Kaybi__2>
For information: On a h264 video + aac audio, those commands seems to work pretty nicely
<jprjr>
iconoclasthero: so in your example ffmpeg cli you didn't specify an output codec though I'm assuming you know that
<jprjr>
iconoclasthero: so that bitrate + codec combo is likely just a default for mpeg-ts. But yeah you can do Opus. I haven't tried udp+mpegts+opus+mediamtx specifically but the docs indicate you can do that
<jprjr>
I'll be honest I'm not sure what zeromq does with ffmpeg, never used it
<iconoclasthero>
@jpjr by any chance do you have any idea why mpd/ffmpeg can't make the unnamed pipe output to stdin work with zmq? i can get ffmpeg to use cat /tmp/mpd.fifo | ffmpeg stdin from the named fifo pipe, i can just use the named pipe as -i /tmp/mpd.fifo
<iconoclasthero>
but i cannot get it to work where mpd calls ffmpg directly in the type "pipe" with ffmpeg in command
<iconoclasthero>
ok
<iconoclasthero>
sorry, i was typing with your responses off screen.
<jprjr>
All good
<iconoclasthero>
ffmpeg errors out with "Address family not supported by protocol (src/ip_resolver.cpp:542) / Aborted"
<iconoclasthero>
oh, i don't know if this is relevant, but ffmpeg says this: `Input #0, s16le, from 'fd:':`
<iconoclasthero>
if fd is file descriptor it occured to me it might need a number, eg., fd1
<jprjr>
You could try 'pipe:0' instead of '-' for your input in ffmpeg but I'm pretty confident it knows '-' means stdin
<iconoclasthero>
and i have no idea why ip_resolver would be in the mix since i'm using zmq:tcp://127.0.0.1:5555
<iconoclasthero>
yeah, same thing.. and i see the same stream/bitrate info so it's getting the stream just fine i think. it's sending it to 127.0.0.1:5555 that appears to be the issue.
<jprjr>
Question, are you using your ffmpeg command line directly in your mpd.conf file, or are you calling a script that then calls ffmpeg?
<iconoclasthero>
does src/ip_resolver.cpp:542 mean that the error is from the line 542 of ip_resolver.cpp in the source code?
<iconoclasthero>
the former.
<iconoclasthero>
wait, it works when I call everything from the cli (i've not actually written this into a script or alias yet)
<iconoclasthero>
but the rtsp outputs i have work when i call them from mpd directly and that's simpler so i was hoping i could do the same with zmq
<jprjr>
I wonder if MPD is doing something odd with parsing the CLI, right
<jprjr>
that error message about ip_resolver is coming from libzmq
<iconoclasthero>
i looked into that briefly by looking at the ps aux and tried to escape the address, etc., and it didn't help
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<jprjr>
yeah, I always just make mpd pipe outputs call like /home/me/bin/whatever, and then update said script
<jprjr>
That way I rule out anything with how MPD parses things, calls processes, etc
<iconoclasthero>
i suppose i can change my enable alias to an enable script and when i call the zmq thing it can just start ffmpeg in a detached screen
<jprjr>
Question is MPD running as the same user as you or is it running as a daemon under systemd?
<iconoclasthero>
the latter
<jprjr>
Hm so, if your systemd unit is restricting any syscalls or anything that may have something to do with it?
<iconoclasthero>
do you think that it if that was the case rtsp would still work?
<jprjr>
Unsure
<jprjr>
Since it works from your local CLI which is really a pretty different environment, that makes me wonder if either there's a restriction from systemd, or maybe libzmq relies in some environment variable being set, things like that
<iconoclasthero>
that's what ps aux shows when it tries to start it from the pipe output, mpd says broken pipe and ffmpeg says the thing about the address family which is apparently coming from zmq
<iconoclasthero>
i keep scrolling down the user list hoping to see "lunalovegood" who told me about zmq in the first place, lol
<jprjr>
In order to make things match more closely - I would try running MPD as yourself in the CLI, same way you're running your zmq ffmpeg command
<jprjr>
If it works there - then you know it's likely some environment issue
<iconoclasthero>
that is excellent troubleshooting advice.
<iconoclasthero>
i need to go back and figure out how that's done since when pipewire came through i flat out refused to switch from a daemon.
<jprjr>
Looking at the libzmq source, I do see calls to getenv, looks like it needs to create some kind of temp path at some point in some ip-related code but, I don't know if that's really the issue or not. Just that there seem to be cases where it looks at environment variables
<iconoclasthero>
well, it's helpful to know WHERE the problem is... if it's not mpd and it's not ffmpeg then i can stop barking up those trees
<iconoclasthero>
but yeah, i'll stop the systemd service and run it as $USER
<iconoclasthero>
i might wait til this cover of knockin on heavens door is over though, lol
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<jprjr>
Also re: pushing traffic through nginx and reducing ports, I've had more success using haproxy for port multiplexing, if that's what you're talking about
<jprjr>
To do things like, ssh, ssh-over-tls, https, irc all on port 443
<iconoclasthero>
huh.
<iconoclasthero>
what irc do you use?
<iconoclasthero>
please say quasselcore/quasselclient!
<jprjr>
I run a bouncer - soju
<jprjr>
then have gamja as a web interface, which is what I'm using now actually
<iconoclasthero>
"haproxy" was a word i ran across recently but haven't investigated
<iconoclasthero>
i stopped using a > decade-old router and had to open all the ports on a new router so it was time to rationalize what was open for several obvious reasons.
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<jprjr>
Yeah, the issue with nginx is a port can either be an HTTP listener, or a stream listener, not both. You could, I think (maybe?) define a port 443 stream listener then proxy to https on some other port, or a unix pipe, etc
<jprjr>
but I don't think nginx supports using the PROXY protocol with proxied tcp connections (it only supports it for listeners) so your backend services may not have a way to determine the client IP
<jprjr>
So I do haproxy in front of nginx, have haproxy use the PROXY protocol when communicating with nginx, configure nginx to get the IP from that so various logs etc show the client IP
<jprjr>
Then in haproxy you can do content inspection, or route based on the SNI of the TLS connection, things like that
<iconoclasthero>
i get to keep everything already on nginx there ?!!? that's fantastic if so.
<jprjr>
Oh yeah
<iconoclasthero>
so the theoretical lift (in my mind) to get everything moved from nginx to haproxy just got a hell of a lot lighter.
<jprjr>
When you define the listener in nginx you just add the proxy_protocol directive, like "listen 8443 proxy_protocol;", and use the realip module to specify you'll allow the realip from 127.0.0.1
<jprjr>
* "listen 127.0.0.1:8443 proxy_protocol" - probably just have it listen on localhost
<iconoclasthero>
this is in nginx.conf or the sites_avalable confs?
<jprjr>
then in haproxy there's a flag named like, "send-proxy" that will enable the proxy protocol
<jprjr>
I think you need it defined on very "listen" directive
<jprjr>
Can't recall if that's one of those settings that will get inherited by duplicate listen directives or not
<iconoclasthero>
so ... erver {
<iconoclasthero>
listen 80;
<iconoclasthero>
in my sites-enabled
<jprjr>
I think so? That's how it is in my config technically but also, I don't actually define listeners like that
<jprjr>
My config is more like: server { include snippets/listen_https.conf; }
<jprjr>
That way if I do have to change something I just change that included snippet file and reload
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<jprjr>
instead of updating all my sites-available conf files
<iconoclasthero>
i think that if i have to go through and change everything in sites-enabled then i should probably do that as well... i mean i'm making the change anyway, right?
<jprjr>
So I have it defined on every listen directive since that's included multiple times, so I'm not 100% sure if you *have* to define it on every listen directive or not
<jprjr>
because I know some flags like, ipv6only - you only have to define on 1 listen directive and it gets inherited
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<jprjr>
Anyway we've gotten pretty far from ffmpeg lol
<soreau>
I am getting this error when attempting to build my applications against latest ffmpeg:
<soreau>
error: 'AVFilterLink' {aka 'struct AVFilterLink'} has no member named 'frame_rate'
<soreau>
has this changed recently? the docs seem to say it should still exist
<iconoclasthero>
yes, we have and i appreciate the info.
<soreau>
and by latest, I mean git ma*
<soreau>
master
<iconoclasthero>
my ffmpeg is out of date again. :(
<jprjr>
looking at latest release it's there, not sure about latest git
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<soreau>
jprjr: where is it defined?
<soreau>
oh "Definition at line 794 of file avfilter.h."
* soreau
looks
<jprjr>
Looks like it's not in latest git
<soreau>
ah
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<soreau>
So what's the solution, forget/nix it, or some complex replacement?
* soreau
git log's
<soreau>
-p
<jprjr>
Guess it depends on what you're doing with that field
<soreau>
lavfi: move AVFilterLink.frame_rate to FilterLink.. a23d565ea7
<jprjr>
I think that implies they made it an internal field
<soreau>
so just nix it
<jprjr>
Since FilterLink isn't prefixed it's probably just kept private
<soreau>
because things seem to work fine without it
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<soreau>
what might be the next release date with this change? (if that's not asking too much ;)
<jprjr>
I've never used AVFilterLink myself, at least not directly. I use avfilter_link() to link filter contexts together
<soreau>
I see
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<Kaybi__2>
Well -ss and -to are fine to limit the number of generated segments, now I'll have to find a way to keep timestamps from the original file and the segment numbering (since it starts with 0 on each new command)
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<another|>
possibly `-initial_offset` ?
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<jprjr>
iconoclasthero: interesting, maybe libzmq is using AF_NETLINK for some kind of IPC thing?
<iconoclasthero>
got me, but you were right that it was systemd. that unit file came directly from the compiled git source of mpd 0.23
<iconoclasthero>
so that's something Kellerman et al. put in there for some reason. replacing it with these and it works:
<iconoclasthero>
IPAddressAllow=localhost
<iconoclasthero>
IPAddressDeny=any
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<iconoclasthero>
jprjr ^
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<jprjr>
Yeah, I'm guessing that's all MPD itself really needs for 99% of use-cases
<jprjr>
Not sure how many other people are spawning ffmpeg, which then pulls in libzmq, right
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<iconoclasthero>
well, jptjr, i think it would be great if mpd could be compiled with zmq directly and cut out ffmpeg...and since it'll never happen, i might as well include rtsp and webrtc in this hypothetical mpd that will never exist.
<iconoclasthero>
but the autossh tunnel and zmq have been remarkably stable even as I've moved my laptop back and forth from 2 lans.
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<lairv>
I have an issue with this command: 'ffmpeg -i bg.mp4 -c:v libvpx-vp9 -i overlay.webm -filter_complex "[0:v][1:v]overlay=shortest=1[out]" -map "[out]" merged.mp4'
<lairv>
it combines a background video with a foreground video, and the results has the length of the shortest video
<lairv>
however in my tests, the output has actually one less frame than the shortest video
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<travisghansen>
I have a command to stream that I want to add an ability to simulcast to using tee...before even trying multiple outputs I am trying a single and it fails...what am I doing wrong: -f rtsp rtsp://127.0.0.1:8554/streamystreamy vs -f tee '[f=rtsp]rtsp://127.0.0.1:8554/streamystreamy'
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<another|>
>Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-select output streams. So all streams intended for output must be specified using "-map". See the examples below.
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<travisghansen>
ah, thank you kindly
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<travisghansen>
I'm an idiot with mapping (and ffmpeg in general) but that should push me over the hill for this setup, thanks