[Ffmpeg-devel-irc] ffmpeg.log.20170426

[burek]

[00:06:47 CEST] <james999> any idea what this means? i'm trying to capture the screen in windows
[00:06:49 CEST] <james999> [dshow @ 00000000004e7fe0] real-time buffer [UScreenCapture] [video input] too f
[00:06:49 CEST] <james999> ull or near too full (100% of size: 1000000000 [rtbufsize parameter])! frame dro
[00:06:49 CEST] <james999> pped!
[00:08:44 CEST] <faLUCE> is ffmpeg compatible with GPL 3.0 ?
[00:10:42 CEST] <JEEB> faLUCE: you can configure it as --enable-gpl --enable-version3 if you really need to (usually for apache2 stuff)
[00:18:00 CEST] <kepstin> faLUCE: a fairly large chunk of the ffmpeg code is either in LGPL2.1 or later, GPL2 or later, or GPL3 or later licenses, so as long as you don't enable anything incompatible with those, then yes.
[00:24:47 CEST] <faLUCE> thanks JEEB kepstin
[01:23:30 CEST] <|ance|ott> hi guys, I have a file in raw format, it is an UDP stream of a video call and I would like to play it.....I've used the tool videosnarf and it works when video call is using codecs (audio: pcmu, video: h264) but when i used (audio: opus, video: vp8) i can't play it
[01:23:43 CEST] <|ance|ott> appreciated any hint
[01:24:32 CEST] <|ance|ott> is there a way to decode it using ffmpeg?
[01:28:33 CEST] <furq> ffmpeg has builtin decoders for both of those so that should just work
[01:29:23 CEST] <furq> also libvpx-vp8 is both slower and worse quality than x264, so you should probably stick with h264 video
[01:29:31 CEST] <furq> assuming that thing uses x264 for video encoding which it probably does
[01:29:49 CEST] <|ance|ott> @furq thanks, I've tried and I got an error all the time
[01:33:55 CEST] <furq> are you going to tell us what the error is, or should we guess
[01:34:27 CEST] <|ance|ott> i'll just retrieving...wait pls
[01:34:46 CEST] <n4zarh> is there anyone who can tell me how to build ffmpeg for armeabi-v7a with android ndk? I can't find a way to set flags properly :(
[01:34:53 CEST] <n4zarh> and i am clueless
[01:36:04 CEST] <n4zarh> link to ./configure params for this build: https://pastebin.com/ZUSSdCNy
[01:36:09 CEST] <|ance|ott> so first of all, this is the command I am using it could be the wrong one
[01:36:36 CEST] <|ance|ott> ffmpeg -i opus_vp8.pcap -vcodec copy out.mp4
[01:36:43 CEST] <furq> oh
[01:36:51 CEST] <furq> replace out.mp4 with out.webm
[01:36:57 CEST] <furq> mp4 doesn't support vp8
[01:38:12 CEST] <|ance|ott> got it...this is the error anyway
[01:38:13 CEST] <|ance|ott> opus_vp8.pcap: Invalid data found when processing input
[01:38:27 CEST] <|ance|ott> even after your suggestion
[01:39:24 CEST] <furq> i take it .pcap means this is literally a wire capture
[01:39:28 CEST] <|ance|ott> this is the app which generate the that .pcap file
[01:39:29 CEST] <|ance|ott> https://github.com/sipwise/rtpengine
[01:39:49 CEST] <|ance|ott> and it says:
[01:39:50 CEST] <furq> i'd have to guess ffmpeg doesn't know how to read those codecs without a container
[01:39:51 CEST] <|ance|ott> When recording to pcap file in raw (default) format, there is no ethernet header. When set to eth, a fake ethernet header is added, making each package 14 bytes larger.
[01:40:11 CEST] <|ance|ott> and I am using the default option
[01:40:33 CEST] <|ance|ott> @furq "i'd have to guess ffmpeg doesn't know how to read those codecs without a container" it means i should pass an extra parameter?
[01:40:54 CEST] <furq> i have no idea how you'd go about reading that
[01:41:25 CEST] <|ance|ott> haha okay
[01:41:39 CEST] <|ance|ott> someone told me that Mplayer should do the magic
[01:41:43 CEST] <|ance|ott> what do you tbhink?
[01:41:50 CEST] <|ance|ott> even without any decodification
[01:42:02 CEST] <|ance|ott> just passing the file to the Mplayer but I've not tried
[01:42:29 CEST] <furq> if this is a dump of rtp then maybe -f rtp before -i
[01:42:39 CEST] <furq> idk if that does anything with file inputs though
[01:43:50 CEST] <|ance|ott> sweet
[01:43:56 CEST] <|ance|ott> i used that option and now
[01:43:58 CEST] <|ance|ott> Unsupported RTP version packet received
[01:44:03 CEST] <|ance|ott> Unable to receive RTP payload type 96 without an SDP file describing it
[01:44:15 CEST] <|ance|ott> and that's because those codecs are VRB
[02:00:20 CEST] <james999> ok success sort of
[02:00:42 CEST] <james999> i  finally got mpeg2video and mpegts to stream to my xbox with vlc app
[02:00:53 CEST] <james999> now only problem is it isn't syncing properly
[02:03:03 CEST] <james999> it's like, not synced up when I go to the other room to check
[02:03:11 CEST] <james999> I got rid of rtbufsize but idk do i need to control the frame rate?
[02:08:28 CEST] <james999> er wtf
[02:08:37 CEST] <james999> if I put -r 15 for input and -r 30 for output
[02:08:43 CEST] <james999> does it automatically duplicate frames to bring it up?
[02:13:28 CEST] <james999> bleh this is never gonna work
[02:13:44 CEST] <james999> now i have audio, i have video, but the audio plays from the beginning when I load the stream from the xbox
[02:13:48 CEST] <relaxed> why not use dlna to server the video?
[02:13:55 CEST] <james999> while the video is synced
[02:14:07 CEST] <james999> yes indeed relaxed, that's what I want to know
[02:14:13 CEST] <relaxed> https://sourceforge.net/projects/minidlna/
[02:14:50 CEST] <relaxed> I use that with my roku
[02:15:13 CEST] <|ance|ott> okay i think i am close with ffmepg, I am getting this error: Guessing on RTP content - if not received properly you need an SDP file describing it
[02:15:31 CEST] <|ance|ott> and I do have the SDP metadata, how do I pass the sdb data as parameter to ffmepg?
[02:15:32 CEST] <james999> to answer your question non sarcastically, I'm trying to do a screencast
[02:15:47 CEST] <james999> and that necessitates using ffmpeg cause I spent 2 hours on vlc already trying to get it to work
[02:16:45 CEST] <relaxed> screencasting to your xbox? I'm interested in your use case
[02:17:42 CEST] <james999> so far i have this command and miraculously I managed to get it to work with vlc app on xbox one
[02:17:55 CEST] <james999> ffmpeg   -f dshow   -r 15 -i video="UScreenCapture"  -vcodec mpeg2video  -f dshow -i audio="Stereo Mix (Realtek High Defini" -listen 1 -r 30 -f mpegts http://0.0.0.0:8082
[02:18:22 CEST] <james999> unfortunately when it starts playing the video is synced up but the audio is delayed to when i started the command
[02:19:18 CEST] <relaxed> try h264/aac in flv
[02:20:37 CEST] <james999> ok sec
[02:21:37 CEST] <relaxed> -async 1 might help with the audio sync
[02:22:04 CEST] <james999> it says unknown decoder libx264
[02:22:09 CEST] <james999> maybe if i put it after both inputs
[02:22:20 CEST] <klaxa> yes, use inputs first, then codecs :P
[02:22:21 CEST] <relaxed> I meant for the output
[02:22:41 CEST] <klaxa> argument order is important
[02:24:50 CEST] <james999> i dont' see aac anywhere when I type ffmpeg -encoders
[02:25:16 CEST] <james999> oh there it is nvm
[02:25:20 CEST] <klaxa> was about to say
[02:25:25 CEST] <klaxa> ffmpeg even has a native aac encoder
[02:27:14 CEST] <james999> ok that sort of worked relaxed
[02:27:19 CEST] <james999> but the audio/video are still unsynced
[02:28:35 CEST] <james999> i'm using Uscreencapture since the other dshow thing i got Medialooks didn't work
[02:28:52 CEST] <james999> i think the -r 15 and -r 30 is somehow doubling the frames maybe to bring it up to 30 fps output
[02:28:54 CEST] <james999> but i'm not sure
[02:29:02 CEST] <klaxa> i think it is
[02:29:06 CEST] <furq> yes it is
[02:29:07 CEST] <james999> let me take them out and see what happens
[02:29:24 CEST] <klaxa> i remember having sync issues like that with live recordings as well
[02:29:54 CEST] <klaxa> i don't think it's really fixable? i would assume it comes down to input being opened not at the exact same time
[02:31:09 CEST] <james999> i think taking the -r out just made it worse
[02:31:35 CEST] <relaxed> if it's just starting at the wrong time and not drifting, -itsoffset might help
[02:31:44 CEST] <james999> klaxa: the problem is so far with the -r options the video starts right when I load the stream but the audio starts at the beginnign when I initiated the ffmpeg command
[02:32:22 CEST] <james999> to be clear the xbox is in the other room, so I'm starting ffmpeg, starting a youtube vid on my pc. Then I walk to the living room and activate stream on xbox which takes about 15 sec
[02:33:01 CEST] <james999> i think somehow the sound is starting at the time index when ffmpeg starts but the video is starting synced up
[02:33:17 CEST] <james999> without the -r i think it might be the reverse
[02:38:15 CEST] <james999> does itsoffset just do a constant offset?
[02:38:31 CEST] <james999> i think i need a way to tell it to force the dshow inputs to be synchronized
[02:39:06 CEST] <relaxed> yes
[02:46:04 CEST] <james999> hmm I tried  -vf scale=240:-1  and that just made it green and distorted
[02:47:10 CEST] <james999> also kalaxa idk if th is is intional or not
[02:47:19 CEST] <james999> but the ffmpeg process quits when I close the stream on my phone or xbox
[02:53:29 CEST] <furq> yeah it'll do that
[02:53:39 CEST] <furq> ffmpeg isn't a media server
[02:57:31 CEST] <james999> i'm getting a ton of messages like this but i don't u nderstand what it means
[02:57:33 CEST] <james999> [dshow @ 0000000000618540] real-time buffer [UScreenCapture] [video input] too f
[02:57:33 CEST] <james999> ull or near too full (193% of size: 3041280 [rtbufsize parameter])! frame droppe
[02:57:33 CEST] <james999> d!
[02:57:33 CEST] <james999>     Last message repeated 6 times
[02:59:12 CEST] <james999> hey bombzen. i've made progress since we last spoke
[02:59:23 CEST] <thebombzen> wow hovering much
[02:59:27 CEST] <james999> now instead of a blank screen i have choppy video and desynced audio displaying on my xbox
[02:59:49 CEST] <thebombzen> Okay.
[03:00:09 CEST] <furq> james999: set -rtbufsize to something higher than 3041280
[03:03:03 CEST] <james999> furq: ok i'll try that next. i'm also not sure how to limit the size of the video and audio. i tried vfilter to resize but not sure if that worked
[03:03:13 CEST] <james999> i'm trying to stream over wifi if that helps
[03:03:35 CEST] <furq> if you're just streaming to vlc then rtmp is probably a good choice
[03:04:36 CEST] <furq> maybe not if you have serious latency requirements, but that's probably not going to happen over wifi anyway
[03:05:03 CEST] <james999> i just change the http:// to rtmp:// ?
[03:05:11 CEST] <james999> sorry i've been working on this all day, tired and a little confused lol
[03:05:28 CEST] <furq> you'd need some kind of rtmp server
[03:05:36 CEST] <furq> which will also solve the problem of the stream dropping when a client disconnects
[03:05:41 CEST] <furq> and also only being able to serve one client
[03:05:43 CEST] <james999> by the way I tried increasing the rtbufsize 3 times
[03:05:53 CEST] <james999> each time it fills up to that size and says the same error about frames dropped
[03:06:04 CEST] <furq> there are other protocols you could use but i actually know an rtmp server which is relatively easy to set up and doesn't suck
[03:06:09 CEST] <furq> which is more than i can say for rtsp or anything else
[03:07:01 CEST] <furq> https://github.com/arut/nginx-rtmp-module/
[03:07:09 CEST] <james999> hopefully vlc on xbox is able to decode rtmp:// because it couldn't do udp:// earlier and I had to switch to http:// to get it to work
[03:07:18 CEST] <furq> vlc on every other platform can do rtmp
[03:07:24 CEST] <furq> so if the xbox one can't then that'd be weird
[03:07:32 CEST] <james999> udp didn't work so idk
[03:07:41 CEST] <james999> http: is sort of working with desynced audio and video
[03:07:44 CEST] <furq> udp is pretty much a non starter over wifi anyway
[03:09:06 CEST] <james999> i don't see rtmp mentioned on the wiki for nginx
[03:09:15 CEST] <james999> does that mean ihave to install the module separately from taht link?
[03:09:17 CEST] <furq> yes
[03:09:24 CEST] <james999> ok
[03:09:32 CEST] <furq> you might need to rebuild ffmpeg with that module
[03:09:41 CEST] <furq> idk if they sorted out the dynamic module loading yet
[03:09:43 CEST] <james999> i didn't build ffmpeg, I downloaded a binary
[03:09:55 CEST] <furq> you might need to build it then
[03:10:09 CEST] <james999> it says  --enable-librtmp on the command line when i run ffmpeg
[03:10:15 CEST] <furq> er
[03:10:18 CEST] <furq> rebuild nginx
[03:10:40 CEST] <james999> ah ok.
[03:10:58 CEST] <james999> right now my ffmpeg command is ffmpeg -rtbufsize 900000000 -f dshow  -i video="UScreenCapture" -f dshow -i audio="Stereo Mix (Realtek High Defini" -vcodec libx264  -acodec aac -listen 1 -f flv http://0.0.0.0:8082
[03:11:22 CEST] <furq> get rid of -listen 1 and replace the http output with your server's rtmp endpoint
[03:11:26 CEST] <furq> everything else is fine
[03:11:41 CEST] <james999> i added that b/c klaxa said there's a basic http server inside ffmpeg so i tried it out
[03:11:54 CEST] <furq> emphasis on "basic"
[03:11:59 CEST] <james999> so the wierd audio and choppy desyncing is being caused by not having an rtmp server?
[03:12:12 CEST] <furq> as you just found out, all the http server stuff will only serve one client and it'll drop out when no clients are connected
[03:12:14 CEST] <james999> it's odd because it streams fine to my phone with vlc but not to the vlc app on the xbox one
[03:12:21 CEST] <furq> all the ffmpeg server stuff in general will do that
[03:12:38 CEST] <furq> if the xbox is on wifi then that could just be down to bad signal
[03:12:55 CEST] <james999> well i'm still getting those frame drop messages and rtbufsize overruns even when ie ncode to a file
[03:13:04 CEST] <furq> shrug
[03:13:04 CEST] <james999> so i think there's still something amiss maybe?
[03:13:12 CEST] <klaxa> if you want to go down a deep rabbit hole and vlc on the xbox one supports mkv you could try using my mkv server
[03:13:13 CEST] <furq> i can't judge the dshow stuff, i've never used that
[03:13:26 CEST] <james999> well i'm using whatever i can get
[03:13:31 CEST] <james999> if there's another way besides dshow i'll take it
[03:13:44 CEST] <klaxa> it may be synced better but it adds a lot of buffering, also not sure how well it handles real-time input
[03:13:46 CEST] <james999> klaxa: there's such a thing as an mkv... server...? o_0
[03:13:50 CEST] <furq> you might just want to use OBS or something
[03:14:08 CEST] <klaxa> you asked me about http server in ffmpeg after i posted the github link :P
[03:14:17 CEST] <klaxa> https://github.com/klaxa/mkvserver_mk2
[03:14:24 CEST] <furq> i've never used ffmpeg for anything other than streaming files, but based on conversations here it's a bit flaky with live sources
[03:15:21 CEST] <james999> ok downloading open broadcasting studio now. although i'm not encouraged by the help page
[03:15:38 CEST] <james999> "This guide assumes you already have a basic knowledge of streaming services and streaming terminology,"
[03:16:15 CEST] <furq> it's aimed at people who stream themselves pretending to get scared at video games to youtube, so i'm sure you'll be fine
[03:16:31 CEST] <james999> oh yeah klaxa i was on this page before lol
[03:16:46 CEST] <james999> furq: kk
[03:16:54 CEST] <james999> originally i tried dlna/upnp servers like plex and serviio
[03:17:06 CEST] <klaxa> obs sounds like a good idea though tbh
[03:17:06 CEST] <james999> but they wouldn't take the streaming link from the above ffmpeg command to the xbox correctly
[03:17:21 CEST] <furq> you'll still need some kind of streaming server with OBS
[03:18:21 CEST] <furq> and i have no idea what kind of latency you'll get, if that's important
[03:18:25 CEST] <james999> ok i got nginx-1.13 downloaded
[03:18:28 CEST] <furq> i suspect you'll have difficulty getting it under a second or so
[03:18:31 CEST] <james999> but idk let me see if i can get that rtmp plugin thing
[03:21:08 CEST] <james999> it's saying to go into the nginx source dir and do ./configure --add-module=path/to/rtmp module
[03:21:28 CEST] <james999> since i'm on windows idk do i need mingw and cygwin or something? there's no build instructions for windows
[03:24:52 CEST] <james999> oh nvm i found a link to a pre-compiled binary. *whew*
[03:38:26 CEST] <james999> ok i have nginx with rtmp support
[03:38:36 CEST] <james999> i guess is hould disable my apache server before i turn it on
[03:52:32 CEST] <james999> i downloaded a version with rtmp pre installed allegedly
[03:52:39 CEST] <james999> but i keep getting rtmp unknown error
[04:14:54 CEST] <klaxa> james999, i just tested and my server can handle live data quite well apparently
[04:15:10 CEST] <klaxa> so if you want to go down that rabbit hole i'm glad to be of help
[04:16:36 CEST] <james999> haha cool
[04:16:47 CEST] <james999> i'm currently in the rabbit whole however of compiling nginx on windows
[04:17:06 CEST] <james999> so as soon as i'm out of that i'll be ready.
[04:17:07 CEST] <klaxa> oh wait right windows...
[04:17:15 CEST] <james999> so when you say mkv server
[04:17:19 CEST] <klaxa> never tested to build on windows
[04:17:24 CEST] <klaxa> as i have no windows :P
[04:17:27 CEST] <james999> it's serving mkv file chunks to the vlc ?
[04:17:34 CEST] <james999> ah ok lol
[04:17:42 CEST] <james999> well i can get my linux in virtualbox loaded up
[04:17:51 CEST] <james999> but idk maybe there's already an rtmp in that thing
[04:18:05 CEST] <james999> does your source compile on debian?
[04:18:22 CEST] <klaxa> it reads a file from disk or stdin, chops into chunks in memory and glues them together for each client independently
[04:18:41 CEST] <klaxa> i got a bugreport from someone running debian so i'd guess so
[04:18:57 CEST] <klaxa> but you have to build ffmpeg from source since it needs patches not yet in git-master
[04:19:22 CEST] <james999> well i'd have to download all teh devel stuff for that maybe
[04:19:28 CEST] <klaxa> to the client it's just an mkv
[04:19:31 CEST] <james999> why does ffmpeg need patching?
[04:19:40 CEST] <james999> to the client it's an mkv?
[04:19:42 CEST] <klaxa> because made a mistake 2 years ago :P
[04:19:49 CEST] <james999> so what i go to vlc on my phone and search for a file?
[04:19:52 CEST] <klaxa> no
[04:20:02 CEST] <klaxa> you just open http://your-ip:8080
[04:20:03 CEST] <james999> a URL?
[04:20:07 CEST] <klaxa> yes
[04:20:09 CEST] <james999> oh ok
[04:20:24 CEST] <james999> sorry i am pretty noob so i don't know anything except basic bash commands lol
[04:21:13 CEST] <klaxa> we've all been there before :P
[04:21:31 CEST] <james999> in simple terms if you have just a URL like http://ip
[04:21:36 CEST] <YokoBR> hi folks
[04:21:37 CEST] <james999> how does vlc know to play what's there
[04:21:45 CEST] <james999> or rather what does it expect to find there?
[04:21:48 CEST] <YokoBR> anybody using fluent-ffmpeg?
[04:22:31 CEST] <james999> i saw things like flv, mpegts, and avi as possible -f parameters when doing my testing earlier
[04:22:43 CEST] <james999> i guess those are technically video formats
[04:24:30 CEST] <klaxa> james999: when vlc (or any other player) open the http connection it gets send data, it analyzes that data and determines that it is matroska
[04:24:36 CEST] <klaxa> *opens
[04:25:09 CEST] <james999> sounds simple. so why doesn't anything work with -f fmt in ffmpeg?
[04:27:22 CEST] <klaxa> what
[04:29:13 CEST] <james999> well like
[04:29:20 CEST] <james999> you give it ip addr like http://blah blah
[04:29:27 CEST] <james999> it sees it's avi or mkv or something
[04:29:36 CEST] <james999> so why doesn't it play anything it find there
[04:30:03 CEST] <james999> or it does that and i misunderstand?
[04:30:05 CEST] <klaxa> i still don't get what you are asking
[04:30:20 CEST] <james999> i thought the idea of your server is, now it can find mkv and play it where it couldn't before
[04:31:44 CEST] <klaxa> the idea was to detach reading what you want to stream from writing it to clients
[04:32:09 CEST] <james999> ok
[04:34:28 CEST] <james999> i'm getting ready to run compile command in a min
[04:49:13 CEST] <james999> i can 't believe that compile command worked
[04:53:23 CEST] <klaxa> what did you compile?
[04:56:26 CEST] <james999> nginx with the rtmp module on windows
[04:56:39 CEST] <james999> the guide i was reading for linux gave a link and said it was precompiled for windows already
[04:56:44 CEST] <james999> but i guess that's not true anymore or something
[04:56:53 CEST] <james999> so i'm trying to compile it with mingw on win7
[04:58:25 CEST] <klaxa> alright
[05:00:39 CEST] <james999> fortunately i already had perl and msys installed
[05:00:45 CEST] <james999> and VS
[05:00:51 CEST] <james999> or else this would take a lot longer
[05:18:47 CEST] <james999> hmm ran into trouble with nginx version mismatch
[05:19:08 CEST] <james999> who helped u on the project klaxa? you said you wrote it with others suggestions
[05:20:10 CEST] <klaxa> nicolas george mentored me during gsoc 2015 and ubitux reported a bug when the internal api changed and i started working on it after that
[05:20:19 CEST] <klaxa> i wrote all the code myself
[05:20:33 CEST] <klaxa> they have more of a passive role in it
[05:36:44 CEST] <james999> ah ok
[05:36:53 CEST] <james999> yeah you said something about Google summer of code i think?
[05:53:20 CEST] <klaxa> yes, that was the project where i implemented the http serving capabilities
[05:53:45 CEST] <klaxa> with lots of help and advice from nicolas george and others
[05:59:45 CEST] <james999> ok this is last try on my end
[06:00:02 CEST] <james999> i found a reddit post where guy says literally download x file from y page and it has rtmp nginx in it
[06:00:07 CEST] <james999> if it doesn't then oh well
[06:00:38 CEST] <james999> klaxa you said your project compiles on debian?
[06:00:53 CEST] <james999> that should be good then. i think.
[06:01:17 CEST] <klaxa> yes, keep in mind you need to patch ffmpeg itself and compile it from source
[06:01:32 CEST] <james999> well it can't be harder than what i've been doing just now
[06:01:49 CEST] <james999> adding VC folders to my windows path, changing file names in openssl folder
[06:02:09 CEST] <james999> i even had to rename perl.exe in my mingw install to perl_UNUSED.exe so that it would use the right perl i installed!
[06:21:12 CEST] <mattwj2002> hi guys
[06:21:52 CEST] <james999> weird
[06:22:03 CEST] <mattwj2002> will a quadro 600 support h265 encoding via the graphics card?
[06:22:10 CEST] <james999> i got the nginx rtmp server working, and started ffmpeg streaming to it and i can load it on my phone but video not coming through
[06:29:12 CEST] <james999> i can't even stream a video file wtf
[06:29:24 CEST] <james999> ffmpeg -f mp4 -i "I:\\yt.mp4" -f flv rtmp://127.0.0.1:1935/live/live
[06:30:37 CEST] <james999> now vlc just sits there when i try to load that on my phone
[06:30:57 CEST] <james999> oh wait it worked now... wtf
[06:31:36 CEST] <james999> super laggy and choppy, but it worked
[06:46:12 CEST] <mattwj2002> guys I think it is going to work
[06:46:13 CEST] <mattwj2002> :D
[06:46:19 CEST] <mattwj2002> I'll keep you informed
[06:46:19 CEST] <mattwj2002> :)
[06:55:04 CEST] <james999> is there any reason why rtmp://localhost:1935/live/live would not work?
[06:58:43 CEST] <mattwj2002> hi james999
[06:58:45 CEST] <mattwj2002> how are you?
[06:58:52 CEST] <mattwj2002> what are you trying to do?
[06:59:30 CEST] <mattwj2002> are you trying to broadcast from one computer to another computer or device?
[07:00:21 CEST] <mattwj2002> james are you there?
[07:00:28 CEST] <mattwj2002> I might have the answer if you are
[07:00:53 CEST] <mattwj2002> replace localhost or 127.0.0.1 with your nic ip address
[07:00:57 CEST] <mattwj2002> let me know if that works
[07:01:13 CEST] <mattwj2002> for example if your ethernet card is set to 192.168.1.1
[07:01:15 CEST] <mattwj2002> do
[07:01:26 CEST] <mattwj2002> rtmp://192.168.1.1/live/live
[07:01:27 CEST] <mattwj2002> :)
[07:01:39 CEST] <mattwj2002> try that and let me know if it helps james999
[07:03:16 CEST] <james999> oh yeah sorry
[07:03:24 CEST] <james999> i'm just fiddling with my phone and several windows at samet ime
[07:03:46 CEST] <james999> yeah i'm trying localhost, 127.0, and 192.168.1.99
[07:03:52 CEST] <james999> none are working seemingly even on same PC
[07:03:59 CEST] <james999> but one worked for a second. idk what the hell is going on
[07:04:26 CEST] <james999> now i'm getting audio only from 127.0.0.1
[07:04:35 CEST] <mattwj2002> james999: is that suppose to be the source or destination address?
[07:04:46 CEST] <mattwj2002> or is it suppose to be a multicast address?
[07:05:03 CEST] <james999> here's my ffmpeg command. idk what multicast and unicast mean
[07:05:04 CEST] <james999> I:\>ffmpeg -f mp4 -i "I:\\yt.mp4" -vcodec libx264 -acodec aac -f flv rtmp://127.
[07:05:04 CEST] <james999> 0.0.1:1935/live/live
[07:05:11 CEST] <mattwj2002> james999: okay
[07:05:19 CEST] <mattwj2002> are you looking at an example somewhere?
[07:05:27 CEST] <james999> like a dozen at this point lol
[07:05:32 CEST] <mattwj2002> okay
[07:05:32 CEST] <james999> not at the moment
[07:05:38 CEST] <mattwj2002> are you on a windows computer I assume?
[07:05:46 CEST] <james999> this all started with me trying to stream from a windows pc to my xbox one
[07:05:55 CEST] <mattwj2002> okay
[07:06:03 CEST] <mattwj2002> try shutting off your windows firewall
[07:06:06 CEST] <james999> i managed to get vlc installed on the xbox and an nginx server with rtmp on this pc after someone here suggested that
[07:06:11 CEST] <james999> did that
[07:06:17 CEST] <mattwj2002> it might be able to go out but can't come back
[07:06:39 CEST] <mattwj2002> here is an idea
[07:06:51 CEST] <james999> this is over wifi by the way. i'm not sure but mb other people on the network are congesting it
[07:07:03 CEST] <mattwj2002> if you do ffplay -i "I:\\yt.mp4"
[07:07:08 CEST] <mattwj2002> does the file play?
[07:07:31 CEST] <james999> not sure i have ffplay on windows but i'll check
[07:07:41 CEST] <james999> i restarted the command and it played for a second in vlc player then froze
[07:08:48 CEST] <mattwj2002> oh okay
[07:08:55 CEST] <james999> yes to your question
[07:08:59 CEST] <james999> ffplay is able to play the file fine
[07:09:03 CEST] <mattwj2002> okay
[07:09:13 CEST] <mattwj2002> plays without stopping then?
[07:09:18 CEST] <james999> yeah
[07:09:28 CEST] <mattwj2002> hmmmm
[07:09:30 CEST] <james999> but it seems when I try to stream it rtmp and load it in vlc it feezes after a second
[07:09:41 CEST] <mattwj2002> 1080p?
[07:09:55 CEST] <mattwj2002> how big is the file in terms of size and resolution?
[07:10:56 CEST] <james999> it's 47MB, about 5 minutes, 1280x720
[07:11:16 CEST] <james999> i downloaded it from youtube for testing purposes
[07:12:11 CEST] <mattwj2002> oh that is nothing
[07:12:29 CEST] <james999> vlc says it's h264 and aac audio
[07:12:32 CEST] <mattwj2002> I was thinking it might be 1080p or worse yet 4k in h265
[07:12:32 CEST] <james999> yeah that's the idea
[07:12:33 CEST] <mattwj2002> :)
[07:12:38 CEST] <james999> lol no
[07:12:44 CEST] <mattwj2002> I thought you were overloading your system
[07:12:45 CEST] <mattwj2002> haha
[07:12:54 CEST] <james999> i couldn't tell you the diff between h263, h264, and h265 either
[07:13:15 CEST] <mattwj2002> a pentium 3 might be able to play that with the right graphics card
[07:13:16 CEST] <mattwj2002> :P
[07:13:24 CEST] <james999> haha
[07:13:38 CEST] <james999> i was also getting a lot of buffer full/frame drop messages earlier when doing my live capture from desktop
[07:13:46 CEST] <james999> but they've gone now that i'm just streaming a file
[07:14:06 CEST] <james999> kinda pisses me off because i looked for an hour for an rtmp server
[07:14:19 CEST] <james999> i kid you n ot, i downloaded 3 different windows builds of nginx
[07:14:22 CEST] <james999> no rtmp support
[07:14:35 CEST] <james999> then this fucking reddit post says "download version 1.7.11 and it has rtmp in it"
[07:14:50 CEST] <james999> and i'm thinking, why does that have it but 1.7.whatever i downloaded doesn't? o_o
[07:16:26 CEST] <james999> so idk i can stream fine with udp or http from ffmpeg so i suppose maybe the rtmp nginx i downloaded is bad?
[07:16:43 CEST] <mattwj2002> james999: got ya
[07:16:56 CEST] <mattwj2002> give me a couple minutes
[07:17:03 CEST] <mattwj2002> I'll help research with you
[07:17:04 CEST] <mattwj2002> brb
[07:17:09 CEST] <james999> np i'll be around
[07:20:05 CEST] <mattwj2002> so the goal is this.....
[07:20:18 CEST] <mattwj2002> stream a video from one computer to another commputer
[07:20:22 CEST] <mattwj2002> is that correct?
[07:20:33 CEST] <mattwj2002> it isn't one computer to a group of them is it?
[07:21:27 CEST] <james999> oh hey i was testing the xbox
[07:21:33 CEST] <mattwj2002> oh okay
[07:21:37 CEST] <james999> no it's to stream from my win pc to an xbox
[07:21:39 CEST] <mattwj2002> okay from pc to xbox
[07:21:42 CEST] <mattwj2002> got it
[07:21:43 CEST] <james999> i'm just using my phone and local vlc player for testing
[07:21:53 CEST] <mattwj2002> what program is receiving it on the xbox?
[07:21:54 CEST] <james999> i have some upnp servers on my pc that I tried
[07:22:01 CEST] <james999> vlc player
[07:22:17 CEST] <james999> the default media player with xbox can receive upnp/dlna streams
[07:22:37 CEST] <james999> but the upnp things i dl wouldn't accept the streaming http or udp or even rtmp urls from ffmpeg
[07:22:54 CEST] <james999> so someone here suggested doing it with rtmp and that's why i went down this rabbit hole with nginx and rtmp
[07:23:13 CEST] <mattwj2002> got ya
[07:23:21 CEST] <james999> i can send an http stream to the xbox but it was choppy earlier
[07:23:34 CEST] <james999> i think it was libx264 and aac again with no bitrate or parameters
[07:23:40 CEST] <james999> but it was super laggy
[07:23:53 CEST] <james999> that might be due to the wifi or the codec or idk what
[07:24:05 CEST] <mattwj2002> https://rwdy15.wordpress.com/2015/02/12/streaming-with-ffmpeg-and-receiving-with-vlc/
[07:24:10 CEST] <james999> when i say "super laggy" i mean like one frame every 20 seconds laggy
[07:24:11 CEST] <mattwj2002> does that work?
[07:24:27 CEST] <mattwj2002> https://rwdy15.wordpress.com/2015/02/12/streaming-with-ffmpeg-and-receiving-with-vlc/
[07:24:29 CEST] <mattwj2002> oops
[07:24:30 CEST] <mattwj2002> :)
[07:24:36 CEST] <mattwj2002> ffmpeg -re -i input_file.ts -c copy -f mpegts udp://192.168.2.10:1234
[07:24:50 CEST] <mattwj2002> odviously change the ip addres
[07:25:08 CEST] <mattwj2002> that should be the destination address aka the XBOX
[07:25:56 CEST] <mattwj2002> then using vlc on your xbox tell it to receive from udb://{ip addresss of the computer):1234
[07:26:02 CEST] <mattwj2002> see if that works
[07:26:21 CEST] <mattwj2002> *udp sorry
[07:26:40 CEST] <mattwj2002> how new is your computer?
[07:26:48 CEST] <mattwj2002> do know the specs on it?
[07:26:51 CEST] <james999> an i5
[07:27:28 CEST] <mattwj2002> new generation of an i5 should be able to handle that
[07:27:29 CEST] <mattwj2002> hmmm
[07:27:59 CEST] <mattwj2002> instead of wifi
[07:28:19 CEST] <mattwj2002> is an Ethernet connection an option
[07:28:42 CEST] <mattwj2002> ethernet is always better connection that wifi
[07:28:43 CEST] <mattwj2002> :)
[07:28:47 CEST] <james999> well it's late here and i don't want to start moving monitors and shit around while ppl trying to sleep
[07:28:52 CEST] <james999> but i guess i could try it later maybe
[07:28:58 CEST] <mattwj2002> fair enough
[07:28:59 CEST] <mattwj2002> :)
[07:29:02 CEST] <james999> but practically it's not really there's no ehternet port in the living room where the xbox is
[07:29:11 CEST] <james999> which is partly why i try doing this in the first place. XD
[07:29:19 CEST] <mattwj2002> lol
[07:29:20 CEST] <mattwj2002> got ya
[07:29:40 CEST] <mattwj2002> what standard of wifi do you?
[07:29:57 CEST] <mattwj2002> 802.11a , 802.11b, 802.11g, 802.11n, or 802.11ac ?
[07:30:05 CEST] <james999> i don't know. i tried logging into my netgear52 router earlier and setting the speed to "up to 150Mbps" but i wasn't sure if that did an ything
[07:30:07 CEST] <james999> so i set it back to 54
[07:30:16 CEST] <mattwj2002> got ya
[07:30:36 CEST] <james999> by the way idk which command on that webpage you were interested in
[07:30:43 CEST] <james999> but the ffmpeg one is going reaaally slow ly
[07:30:58 CEST] <james999> speed=0.0625x it says
[07:31:10 CEST] <mattwj2002> james999: I was thinking this one
[07:31:11 CEST] <mattwj2002> ffmpeg -re -i input_file.ts -c copy -f mpegts udp://192.168.2.10:1234
[07:31:17 CEST] <mattwj2002> it is near the top
[07:31:29 CEST] <james999> the ip is my phone or my pc's ip?
[07:31:56 CEST] <mattwj2002> which device are you streaming to and which one are you streaming from?
[07:32:02 CEST] <james999> i'm on my pc
[07:32:05 CEST] <james999> using my phone for testing
[07:32:08 CEST] <james999> xbox is in the next room
[07:32:18 CEST] <mattwj2002> okay
[07:32:39 CEST] <mattwj2002> try using this command
[07:33:16 CEST] <mattwj2002> ffmpeg -re -i input_file.ts -c copy -f mpegts udp://destationIPadd.....where are you pushing the data too
[07:34:27 CEST] <james999> tried this ffmpeg -i I:\\yt.mp4 -c copy -f mpegts udp://192.168.1.15:1234
[07:34:36 CEST] <james999> but my phone at 192.168.1.15 couldn't play it no matter which url i tried.
[07:34:38 CEST] <mattwj2002> basically in this example - input_file.ts is the input file
[07:34:51 CEST] <mattwj2002> udp://ipaddress is the destination
[07:34:54 CEST] <james999> udp://@:1234, udp://@192.168.1.15:1234, a few others i tried
[07:34:56 CEST] <mattwj2002> yes
[07:35:36 CEST] <james999> wait it's coming through sort of
[07:35:39 CEST] <james999> er
[07:36:16 CEST] <james999> plays like a second of sound then stopped
[07:36:35 CEST] <james999> i think something might be wrong with my wifi
[07:36:43 CEST] <james999> i pinged the p hone and got 80ms of delay
[07:36:53 CEST] <james999> but only 2 ms to the xbox in the living room.
[07:37:10 CEST] <james999> but i don't think i got the udp:// to work with the xbox either
[07:37:53 CEST] <james999> haha  that's kinda weird
[07:38:08 CEST] <james999> it plays a second of audio and downloads about 1/8 of the picture from the top
[07:38:19 CEST] <james999> then fills in the rest of the picture with noise
[07:39:49 CEST] <james999> i can't stream to a vlc instance on the same pc though with udp either
[07:39:51 CEST] <james999> is that normal?
[07:40:34 CEST] <james999> oh wait now it worked sort of
[07:40:45 CEST] <mattwj2002> hmmm
[07:41:10 CEST] <mattwj2002> port generally are look to a special part of an application I believe
[07:41:26 CEST] <james999> changed the IP to this pc's ip on the lan and it sort of works.
[07:41:41 CEST] <mattwj2002> sort of?
[07:42:23 CEST] <james999> ffmpeg keeps terminating after a few sec
[07:42:28 CEST] <mattwj2002> hmmm
[07:42:44 CEST] <james999> and then once just now with vlc trying to play udp://@:1234 it worked when I restarted ffmpeg command
[07:42:56 CEST] <mattwj2002> try to find a small file that plays fine locally
[07:43:24 CEST] <james999> this file plays locally
[07:43:32 CEST] <james999> when i open it with vlc yt.mp4 it plays
[07:43:35 CEST] <james999> or what do you mean
[07:43:35 CEST] <mattwj2002> without any glitches
[07:44:02 CEST] <mattwj2002> well you are encoding it right?
[07:44:25 CEST] <james999> i don't know, is -c copy considered "encoding"?
[07:44:26 CEST] <james999> ffmpeg -i I:\\yt.mp4 -c copy -f mpegts udp://192.168.1.101:1234
[07:44:35 CEST] <mattwj2002> maybe you computer can't recode the file on the fly because it can't do it in realtime
[07:45:11 CEST] <mattwj2002> yeah it is transcoding
[07:45:23 CEST] <mattwj2002> from yt.mp4 to mpegts
[07:45:55 CEST] <james999> well i have an avi file i encoded earlier with ffmpeg
[07:46:05 CEST] <james999> it's h264 video/mp3 audio in an avi container
[07:46:07 CEST] <james999> will that do?
[07:46:21 CEST] <james999> or want me to transcode this file i have now to something else?
[07:47:17 CEST] <james999> i remember seeing mpeg2video before as a format
[07:53:42 CEST] <james999> hmm i made an avi container with mpeg2video/mp2 audio and tried streaming it with -f mpegts output over udp
[07:53:48 CEST] <james999> failed with an error though
[07:56:07 CEST] <james999> well i'm just about out of ideas
[07:56:37 CEST] <james999> i've tried everything i can think of to get this to work. the most i got was 3 seconds of lag-free video to my xbox lol
[08:13:49 CEST] <mattwj2002> bummer james999
[08:14:25 CEST] <mattwj2002> james999: what about getting rid of f mpegts all together?
[08:17:10 CEST] <teratorn> james999: -c copy is transmuxing not transcoding
[08:18:10 CEST] <teratorn> imho - some people might still think of it as encoding... I don't
[08:48:28 CEST] <james999> lol nice one teratorn
[08:49:35 CEST] <james999> to answer mattwj2002's question, ffmpeg crashes if I omit that with error [NULL @ 0000000002e7f7c0] Unable to find a suitable output format for 'udp://192
[08:49:36 CEST] <james999> .168.1.101:1234'
[08:50:34 CEST] <james999> If I in turn specify -f avi it terminates normally with messages
[08:50:47 CEST] <james999> [avi @ 000000000035c520] Timestamps are unset in a packet for stream 0. This is
[08:50:48 CEST] <james999> deprecated and will stop working in the future. Fix your code to set the timesta
[08:50:48 CEST] <james999> mps properly
[08:51:16 CEST] <james999> that's with the mpeg2video/mp2 audio avi file
[09:55:50 CEST] <termos> is there a specific order I need to do filters in when deinterlacing + fps filter + scaling?
[09:56:09 CEST] <termos> I do it in that order right now but I'm getting weird artefacts on non-interlaced input
[10:14:23 CEST] <teratorn> termos: does it work if you don't apply a deinterlacing filter on non-interlaced video?
[10:15:47 CEST] <termos> yes but then the output looks interlaced
[10:16:05 CEST] <termos> oh non-interlaced, yes then it works well
[10:18:56 CEST] <termos> I guess my problem is that I don't know if the input is interlaced or not, and I don't want to run yadif if it's not interlaced
[10:19:24 CEST] <termos> deint=interlaced only looks if a frame is tagged as interlaced, but there could be cases where this tagging is not present
[10:22:40 CEST] <durandal_170> try idet filter before deinterlacing
[10:33:37 CEST] <termos> I see thanks, it will forward the result to the next filter? I'll give it a try
[12:05:18 CEST] <diverdude> hi, I am trying to learn how to read and write a video file using ffmpeg. I want to do it in a C++ program. Any examples of how to do that?
[13:16:35 CEST] <termos> diverdude: I would check out the examples here https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples
[13:25:48 CEST] <tiagogomes> Hi, I am trying to determine the framerate of a mp4 file using ffprobe. I am getting this values: 22.92 fps, 60 tbr
[13:26:20 CEST] <tiagogomes> Between the 'fps' and 'tbr' fields, which one is more correct?
[13:35:22 CEST] <BtbN> I think tbr is the inverse timebase
[13:38:40 CEST] <diverdude> termos: ok... so i a have begun my demoprogram : http://paste.ubuntu.com/24459708/   i try to compile it and i get linker errors (also shown in the same paste) I am trying to build on macosX. I dont understand because all library paths should be correct. What am I missing?
[13:42:47 CEST] <termos> should link against bz2 as well, but I would rather use pkg-config or even cmake to compile the program at least if you plan on making it a larger project
[13:58:17 CEST] <pihpah> I am getting Negative values are not acceptable. on this filter options -vf 'pad=ih*16/9:ih:(ow-iw)/2:(oh-ih)/2,scale=426:240'
[13:58:21 CEST] <pihpah> Any idea?
[13:58:39 CEST] <pihpah> Wihtout padding works fine
[14:34:27 CEST] <kepstin> pihpah: if that ends up making the video less wide than it was originally, you could see that issue I guess?
[14:34:38 CEST] <kepstin> in which case, you need to be using the crop filter rather than pad.
[14:39:19 CEST] <alfal> Hello guys! I have a question regarding ffmpeg/ffserver. Is there some way of perodically add a jpg to a stream once every second?
[14:43:43 CEST] <alfal> Anyone? I'm very thankful for any advice!
[15:09:13 CEST] <pihpah> kepstin: got it
[15:23:17 CEST] <diverdude> termos: are you still here?
[15:23:49 CEST] <diverdude> termos: right now i am just trying to get things working
[15:24:14 CEST] <diverdude> termos: later i will think about a proper cross compilation
[15:24:53 CEST] <diverdude> termos: but do you have any idea what I am missing?
[15:25:57 CEST] <diverdude> http://paste.ubuntu.com/24459708/
[15:26:26 CEST] <termos> I would try to add -lbz2
[15:28:46 CEST] <diverdude> termos: still the same errors :(
[15:31:45 CEST] <diverdude> termos: did you try to compile an ffmpeg program on mac before?
[15:33:50 CEST] <termos> yes, try to run the makefile in the examples directory. If that works use the same compiler arguments
[15:34:53 CEST] <faLUCE> Hello. I just published this C++ library, based on ffmpeg. It's for h264/aac http-live streaming (Linux)   https://github.com/paolo-pr/laav . If anyone wants to test it, he is welcome
[15:36:47 CEST] <diverdude> termos: hmmm i dont have example dir in my ffmpeg download
[15:37:06 CEST] <termos> doc/examples/
[15:42:55 CEST] <faLUCE> Hello. I just published this C++ library, based on ffmpeg. It's for h264/aac http-live streaming (Linux)   https://github.com/paolo-pr/laav . If anyone wants to test it, he is welcome
[15:47:09 CEST] <diverdude> ah, sorry....its in doc/examples
[15:49:05 CEST] <erick3k> Hi, can someone help me on why this failed https://0bin.net/paste/go0Mc1dsiHy2Qjza#bEJsWxNGCJIOJXw+3qLyeBld5KuY54EV46kAP9vQdOH
[16:13:45 CEST] <Al3x4nd3r> Hi everybody, i am trying to segment a ts stream, this stream is generated with vlc (through list of files) when vlc changes from a file to another he add a new stream in the ts stream. So the initial stream has maped 0:0 for audio and 0:1 for video, when vlc changes the file he also change the stream of 0:2 for audio and 0:3 for video and so on. When occurs changes in the stream ffmpeg shows that a new stream was added and hangs 
[16:14:27 CEST] <Al3x4nd3r> Anyone knows how to workaround this?
[16:28:06 CEST] <erick3k> nop
[16:28:10 CEST] <erick3k> noboddy here
[16:28:29 CEST] <erick3k> 414 online but it seems like they all bots
[17:10:15 CEST] <petecou__> Morning all, I have a client that has a script that needs to run but they are discovering their videos are badly encoded. It starts getting interframes without keyframes. Is there a way to preencode the videos or flag them so they are rencoded with a keyframe? Here's the error and the script they are running.
[17:10:16 CEST] <petecou__> https://pastebin.com/JPvpX8dt
[17:14:35 CEST] <cryptodechange> Trying to get rid of black bars from an encode
[17:15:07 CEST] <cryptodechange> cropdetect tells me to use crop=1920:800:0:140, but when I use that in my command, I get an error
[17:15:17 CEST] <cryptodechange> [Parsed_crop_0 @ 0x21d6240] Invalid too big or non positive size for width '1920' or height '800'
[17:16:53 CEST] <james999> doesn't cropping in ffmpeg work by specifying how much to crop from each side?
[17:17:09 CEST] <james999> i thought I read that in m travails yeserday with the ffmpeg documentation
[17:17:21 CEST] <cryptodechange> According to the docs, it's width:height:xaxis:yaxis
[17:19:31 CEST] <kepstin> cryptodechange: I'm guessing that you're probably either scaling the video in one command but not the other, or using different inputs...
[17:20:52 CEST] <cryptodechange> https://pastebin.com/H6fHYAzE
[17:21:31 CEST] <cryptodechange> I added -t 1 to make sure it wasn't trying to apply on other tracks, but I understand that's what -vf vs. -f does
[17:22:17 CEST] <cryptodechange> ffmpeg -i input.mkv -vf "cropdetect=24:16:0" -f null -y /dev/null
[17:22:35 CEST] <cryptodechange> That outputs [Parsed_cropdetect_0 @ 0x16f0f20] x1:0 x2:1919 y1:137 y2:942 w:1920 h:800 x:0 y:140 pts:7027479 t:7027.479000 crop=1920:800:0:140
[17:38:37 CEST] <kepstin> cryptodechange: and the *COMPLETE* console output, please?
[17:38:46 CEST] <kepstin> (of the failing command)
[17:56:55 CEST] <james999> hmm
[17:57:03 CEST] <james999> streaming udp to my phone with ffmpeg I get green blockiness
[17:57:25 CEST] <james999> i know video can be streamed b/c I did it earlier with windows7 dlna sharing
[17:57:29 CEST] <james999> any tips?
[17:59:01 CEST] <james999> here's a pastebin: https://pastebin.com/SPu4D9r7
[17:59:21 CEST] <cryptodechange> @kepstin https://pastebin.com/Xyuspktn
[17:59:21 CEST] <cryptodechange> ty
[17:59:31 CEST] <james999> the first part was truncated due to space limitatinos but I can include it if necessary
[18:00:06 CEST] <diverdude> can i use ffmpeg in a commercial product?
[18:02:19 CEST] <kepstin> cryptodechange: ok, so your issue is that the crop filter is being applied to the stream 0:11, which is a  640x360 jpeg. Either use the -map options to exclude that stream, or use qualifiers on the -filter option (like -filter:v:0 ...) to only filter the one stream.
[18:02:45 CEST] <james999> here's an updated paste with the beginning and end of ffmpeg output included and middle redacted: https://pastebin.com/SrcpDs6J
[18:04:27 CEST] <james999> i think maybe i need to duplicate frames, or resize video or something. idk.
[18:04:31 CEST] <kepstin> diverdude: the basic answer is "yes, as long as you follow all the terms of whichever licenses apply to the ffmpeg build you're using (usually LGPL-2.1 or GPL-2 depending on build options)"
[18:07:52 CEST] <diverdude> kepstin: if i enable h264 its LGPL right?
[18:08:24 CEST] <kepstin> diverdude: if you enable h264 encoding via x264, it's GPL-2.0
[18:09:13 CEST] <kepstin> (unless you have a commercial x264 license, I suppose; in that case you probably want a lawyer to help figure out how that interacts with ffmpeg)
[18:09:47 CEST] <diverdude> kepstin: aha yeah ok. No i do not have a commercial of h264
[18:12:16 CEST] <faLUCE> why people are so scared of GPL?
[18:12:35 CEST] <utack> is there a way to get SMPTE 2084 input converted to non-hdr vp9, with correct colors? https://trac.ffmpeg.org/ticket/6132
[18:12:47 CEST] <utack> with no additional arguments the color conversion goes poorly, colors are very wrong
[18:12:48 CEST] <diverdude> kepstin: so lets say i am a developer and i have a client. I build a piece of software based on ffmpeg incl. h264. The client is the only one who will ever use this software. He will use it internally to optimize his production line - he is creating physical products. So could i use ffmpeg in this scenario without breaking any license terms?
[18:13:10 CEST] <kepstin> diverdude: hire a lawyer.
[18:14:01 CEST] <diverdude> kepstin: mmm ok...but how to find a lawyer who knows about these things
[18:14:07 CEST] <kepstin> i suspect it depends on the details of work for hire laws in your jurisdiction, etc. No clear answer, and I can't give legal advice.
[18:14:20 CEST] <JEEB> diverdude: FFmpeg only cares if you break FFmpeg's licrnse that you picked
[18:14:51 CEST] <JEEB> lgpl or id you built with enable-gpl gpl
[18:14:54 CEST] <diverdude> JEEB: but lets say i picked GPL2, would i be breaking the license in my scenario?
[18:15:01 CEST] <kepstin> diverdude: it would probably be perfectly fine if the software that you build and give to your client is also GPL
[18:15:20 CEST] <kepstin> (which of course lets them distribute it and modify it if they want to)
[18:15:30 CEST] <diverdude> kepstin: right yeah
[18:15:32 CEST] <furq> i don't think you can violate the gpl if it's for a product which is only for internal use
[18:15:49 CEST] <furq> the main concern is if you distribute binaries that contain/link to gpl code without distributing the source
[18:16:04 CEST] <JEEB> diverdude: the ones that get your software then get it under gpl (and any other license you might have with them)
[18:16:25 CEST] <furq> also yeah if you're in the EU you don't need to care about h.264 license fees
[18:16:30 CEST] <JEEB> and if they then distribute the binaries those must also be given out as gpl
[18:16:48 CEST] <kepstin> the technicality here is about whether this "building a piece of software for a client" counts as distribution, and that's what you'd need a (local) lawyer to get a legal opinion on.
[18:16:57 CEST] <diverdude> JEEB: ok...but if they dont distribute anything...they are only using it internally in their labs
[18:17:21 CEST] <furq> then that's probably fine
[18:17:30 CEST] <JEEB> yea, but just in case they do
[18:17:33 CEST] <kepstin> diverdude: it's perfectly ok to not distribute GPL software, the terms only come in if they do distribute it
[18:17:34 CEST] <james999> oh hey furq
[18:17:56 CEST] <james999> i have green blocks on udp trying to wirelessly stream to my phone... and lowering size and bitrate don't help. :(
[18:17:59 CEST] <furq> you definitely need to be aware of everyone's responsibilities with gpl software
[18:18:14 CEST] <furq> james999: i've never done any udp streaming
[18:18:15 CEST] <cryptodechange> ty kepstin!
[18:18:20 CEST] <diverdude> allright i see...thanks....this also confirms my understanding of gpl
[18:18:22 CEST] <furq> i'd have to suspect wifi is to blame
[18:18:32 CEST] <JEEB> basically gpl means that if you give the binaries the source has to also be available as gpl
[18:18:46 CEST] <kepstin> diverdude: so give them the GPL software and source code, and make sure that they know what their responsibilities are if they decide distribute it.
[18:18:47 CEST] <JEEB> be it you or that other party
[18:19:00 CEST] <JEEB> yup
[18:19:00 CEST] <furq> i mean even if you're technically "distributing" binaries within the company, it's not a violation unless someone reports it
[18:19:12 CEST] <diverdude> kepstin: yeah - exactly
[18:19:24 CEST] <james999> furq: all the examples on this page at trac.ffmpeg are about udp streaming but not how to fix green blocks. :(
[18:19:27 CEST] <BtbN> There's ffmpeg in my TV, and I'm not aware of any way to modify the software on it.
[18:19:52 CEST] <JEEB> yea but my tv manuf at least gives the lgpl sources
[18:20:07 CEST] <JEEB> which is what lgplv2 requires
[18:20:26 CEST] <JEEB> v3 adds the tivo stuff which is why people tend to stay away from v3
[18:22:53 CEST] <cryptodechange> strange issue now kepstin
[18:23:02 CEST] <cryptodechange> I don't get any errors but it hangs on this output:
[18:23:03 CEST] <cryptodechange> frame=   24 fps=1.3 q=0.0 q=0.0 size=       0kB time=00:00:00.98 bitrate=   0.0kbits/s speed=0.0532x
[18:23:23 CEST] <cryptodechange> speed and fps decreasing
[18:24:14 CEST] <diverdude> is any part of ffmpeg running under LGPL?
[18:24:31 CEST] <furq> depends what you link it against
[18:24:43 CEST] <furq> it's lgpl2 by default but if you link against gpl libraries then the whole thing becomes gpl
[18:25:05 CEST] <furq> likewise with gpl3 libs and non-free libs
[18:31:05 CEST] <diverdude> furq: ah ok i see....
[18:34:23 CEST] <utack> nvm, i got zscale to correctly convert colorspace
[18:34:56 CEST] <james999> specifying packet size over udp with pkt_size made the video not green and blocky. i think
[18:35:15 CEST] <james999> i see a 3 year old bug here about whether pkt_size in ffmpeg is a fixed size or a max?
[18:37:25 CEST] <rictan> what is the difference between muxing and remuxing -- (reference to l-smash open source) -- trying to step through what is happening in the code
[18:37:42 CEST] <rictan> with relation to h.264 to .mp4
[18:40:32 CEST] <furq> rictan: i generally just use muxer for everything
[18:40:48 CEST] <furq> muxer can remux mp4, which is what you'd expect remuxer to do, and also remuxer seems to have fewer options
[18:40:54 CEST] <furq> so i'm not entirely sure what remuxer is for
[18:40:59 CEST] <JEEB> oh
[18:41:05 CEST] <furq> this guy probably knows
[18:41:05 CEST] <rictan> thanks for your answer furq!
[18:41:10 CEST] <JEEB> i'll have to v
[18:41:26 CEST] <JEEB> check cause I didn't expect muxer to do remux
[18:41:39 CEST] <JEEB> i know remuxer is supposed to work on higher level
[18:42:08 CEST] <JEEB> as in not touch actual streams and/or the isobmff structure too much
[18:42:31 CEST] <furq> actually nvm i remembered wrong
[18:42:36 CEST] <furq> muxer can remux, but remuxer has more options
[18:42:46 CEST] <klaxa> udp without any application level sanity checks over wifi will most likely result in lost packets
[18:42:52 CEST] <klaxa> james999: ^
[18:43:00 CEST] <furq> or it has at least one option that i want that muxer doesn't have
[18:43:01 CEST] <klaxa> that would probably explain the green blocks
[18:44:35 CEST] <rictan> cool thanks furq
[18:44:59 CEST] <james999> back now i think
[18:44:59 CEST] <rictan> with regards to l-smash, l-smash muxing does not do re-encoding or compression correcT?
[18:45:03 CEST] <rictan> correct*
[18:45:15 CEST] <james999> yes klaxa i was thinking maybe doubling the framerate would help to alleviate that
[18:45:28 CEST] <klaxa> i... doubt it
[18:45:35 CEST] <james999> i sort of have it working now by spccifying the pkt_size to ffmpeg whereas i just got green blocks before
[18:45:38 CEST] <james999> so i'm making progress
[18:46:26 CEST] <furq> rictan: right
[18:46:31 CEST] <Kiicki> I'm an extremely noob but can I ask what software I can use to be able to convert to "HE-AAC"? I'm on Windows
[18:46:46 CEST] <Kiicki> audio I mean
[18:47:07 CEST] <furq> Kiicki: if you mean with ffmpeg then fdk-aac
[18:47:23 CEST] <furq> if you want something standalone then nero is probably easiest
[18:47:28 CEST] <furq> or fhg
[18:48:04 CEST] <Kiicki> fdk-aac would work? Doesn't handbrake also use it? Because HE-AAC is not an option there when it comes to Windows
[18:48:09 CEST] <Kiicki> Mac does have it though
[18:48:18 CEST] <furq> fdk is gpl incompatible so you can't distribute ffmpeg with fdk
[18:48:23 CEST] <furq> you'd have to build it yourself
[18:48:36 CEST] <furq> the osx handbrake is probably using apple aac
[18:49:02 CEST] <BtbN> Or decided not to care, the fdk license is debatable
[18:49:25 CEST] <james999> i still don't get what these rtbufsize parameters are referring to
[18:49:25 CEST] <james999> [dshow @ 00000000004d6fe0] real-time buffer [UScreenCapture] [video input] too f
[18:49:25 CEST] <james999> ull or near too full (96% of size: 3041280 [rtbufsize parameter])! frame dropped
[18:49:25 CEST] <james999> !
[18:50:13 CEST] <Kiicki> furq that sounds like a lot of hassle. Even licensing comes to play? Should I bother
[18:50:14 CEST] <Kiicki> ?*
[18:50:23 CEST] <furq> what are you trying to do
[18:50:30 CEST] <furq> he-aac is only really useful for streaming
[18:51:08 CEST] <Kiicki> Yes, I'm trying to compress some files. I have heard HE-AAC 128kbs 5:1 in action and it actually sounds decent
[18:51:14 CEST] <Kiicki> so I'm trying to re-create that
[18:51:49 CEST] <furq> if you can use opus then that's probably better
[18:51:49 CEST] <Kiicki> or at least play with it a little bit
[18:52:11 CEST] <furq> there are standalone nero and fhg-aac encoders for windows but idk where you'd get them
[18:52:13 CEST] <klaxa> james999: it's an internal buffer to keep realtime data (according to the docs)
[18:52:20 CEST] <furq> fhg was only ever distributed as part of winamp
[18:52:32 CEST] <furq> there's some cli wrapper for it but i've never had cause to use it
[18:54:21 CEST] <james999> klaxa: well i lost sound again. it was finally playing over udp but delayed by like 5 sec. but full video + audio
[18:54:56 CEST] <james999> i mean i can stream video over this network with minimal buffering. i know that
[18:55:04 CEST] <james999> so it's just a matter of figuring out the right udp settings i think
[18:55:16 CEST] <Mista_D> changing a fourcc vtag on MP4 from "AVC1" to "H264", says its incompatible codec id '28'... Any advice?  https://pastebin.ca/3803353
[18:55:21 CEST] <james999> or maybe using a dshow filter option with UScreencapture
[18:55:26 CEST] <james999> maybe it's slow or something
[18:56:14 CEST] <james999> what would cause constant messages about rtbufsize overflowing from the UScreenCapture and Stereo mix audio though?
[19:01:46 CEST] <klaxa> james999: too much data for a buffer too small, you can increase with the -rtbufsize parameter
[19:05:28 CEST] <james999> idk i think the problem is handling of the screen capture dshow
[19:05:47 CEST] <james999> cuz I tried increasing rtbufsize 3 times and every time it reported 110% overload type messages
[19:06:10 CEST] <james999> let me see if i can set rtbufsize to the max size
[19:07:48 CEST] <james999> what does Past duration 0.890617 too large mean?
[19:07:54 CEST] <james999> that message keeps appaering over and over too now
[19:16:39 CEST] <james999> why do i get an error trying to use gdigrab?
[19:16:51 CEST] <james999> "MPEG-1/2 does not support 5/1 fps"
[19:17:46 CEST] <cryptodechange> I think my command is near ready, but I get a YUV warning
[19:17:51 CEST] <cryptodechange> tells me to use -pix_fmt yuv420p
[19:17:54 CEST] <james999> i get that when I type ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg from the ffmpeg webpage example
[19:19:31 CEST] <cryptodechange> My source tells me this: Chroma subsampling: 4:2:0
[19:19:36 CEST] <cryptodechange> So I should use yuv420p?
[19:20:35 CEST] <cryptodechange> Is there an option to pass to pix_fmt so it keeps it the same as the source?
[19:22:46 CEST] <james999> cryptodechange: I don't know, but you can look at the pix fmt of what you have and compare it to the output of fmpeg -pix_fmts
[19:22:53 CEST] <james999> which will display available pix fmts
[19:24:54 CEST] <james999> i'm not sure if rgb/yuv is encoded in video but
[19:25:15 CEST] <james999> when i type say ffmpeg -i file.mp4 output.flv it displays some info about the input file
[19:25:20 CEST] <james999> int his case it says  Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709)
[19:25:38 CEST] <james999> so i presume the pixel format is yuv420p of each frame in this video
[19:33:38 CEST] <kepstin> cryptodechange: by default ffmpeg will try to preserve pixel format if possible. You forgot to paste the actual output, so I can't say for sure, but I suspect you're doing libx264 encoding?
[19:34:28 CEST] <kepstin> cryptodechange: so in that case, it will print a warning like that if the pixel format being used isn't yuv420p, because formats other than yuv420p have limited player compatibility
[19:53:23 CEST] <cryptodechange> kepstin, it outputted the following:
[19:53:25 CEST] <cryptodechange> No pixel format specified, yuvj444p for H.264 encoding chosen.
[19:53:25 CEST] <cryptodechange> Use -pix_fmt yuv420p for compatibility with outdated media players.
[19:53:40 CEST] <cryptodechange> looks like it defaults to yuvj444p?
[19:54:07 CEST] <kepstin> cryptodechange: it defaults to picking the best value supported by the codec based on what the input is.
[19:54:31 CEST] <kepstin> cryptodechange: so to say why it picked that, I need to know what the input its...
[19:54:50 CEST] <cryptodechange> Oh, I think it's for that jpeg again
[19:55:39 CEST] <cryptodechange> When it outputs the streams, #0:0 is yuv240 (libx264), #0:1 is yuvj444p(image/jpeg)
[19:59:19 CEST] <james999> if i use ffmpeg to display a 1920x1080 on a tv which doesn't support that res
[19:59:23 CEST] <james999> will it look blurry?
[20:01:42 CEST] <james999> any idea how i can make ffmpeg display a jpeg for 20 seconds, then a different one for 20 seconds, then a different one up to 5 or 6 jpegs?
[20:02:44 CEST] <ChocolateArmpits> james999, for the images, use image2 format, it has the required functionality
[20:03:24 CEST] <ChocolateArmpits> james999, next, what do you mean by "doesn't support that res"? It's more about the input format rather than ffmpeg preparing an image
[20:03:48 CEST] <james999> i want to maybe use -loop or something to slideshow a set of images onto a tv
[20:04:00 CEST] <james999> like mb eadch one display for 30 sec plus the file name
[20:04:19 CEST] <james999> i want to display pictures scaled to above its res but i guess that wouldn't accomplish anything idk
[20:04:31 CEST] <james999> my idea was to find out the rest of this tv set by displaying pictures
[20:05:26 CEST] <james999> if i do something like ffmpeg -loop 1 file1.jpg file2.jpg -f flv udp://ip-to-tv will that work?
[20:05:45 CEST] <james999> like show one pic then the other one back and forth?
[20:08:58 CEST] <james999> hmm no that doesn't work
[20:10:38 CEST] <ChocolateArmpits> james999, image2 format specifies an image sequence to play https://www.ffmpeg.org/ffmpeg-formats.html#image2-1
[20:11:52 CEST] <ChocolateArmpits> myself, I would probably have a slideshow playing fullscreen and have a screen grabber running and sending that to your destination, seems simpler. There are image viewing applications that can do this with file names displayed too
[20:12:07 CEST] <james999> oh ok iw as trying to look wt -r and -vf
[20:12:59 CEST] <ChocolateArmpits> well vf is for video filters
[20:13:21 CEST] <james999> i'm confused, i don't have to specify -f image2?
[20:13:26 CEST] <james999> the examples don't say that
[20:13:42 CEST] <ChocolateArmpits> it's probably because of the extension
[20:13:50 CEST] <ChocolateArmpits> but I add it anyways to be 100% clear
[20:14:00 CEST] <james999> [image2 @ 00000000005ba640] Pattern type 'glob' was selected but globbing is not
[20:14:01 CEST] <james999>  supported by this libavformat build
[20:14:13 CEST] <james999> i thought i had the latest ffmpeg build for windows.
[20:14:30 CEST] <james999> i guess it doesn't do that on win w/o file system support
[20:14:30 CEST] <ChocolateArmpits> >This is only selectable if libavformat was compiled with globbing support
[20:14:49 CEST] <ChocolateArmpits> So not just libaformat version is required
[20:15:01 CEST] <ChocolateArmpits> globbing has to be enabled there too
[20:16:43 CEST] <james999> i tried this but it didn't work: I:\>ffmpeg -f image2 -framerate 10 -i sicao_green_tunnel-wallpaper-1280x720.jpg
[20:16:43 CEST] <james999> -i sicao_green_tunnel-wallpaper-3554x1999.jpg -f mpegts udp://127.0.0.1:1234
[20:19:34 CEST] <james999> image2 is not mentioned anywhere in ffmpeg -decoders
[20:19:37 CEST] <james999> is that where it would be?
[20:19:42 CEST] <JEEB> no
[20:19:44 CEST] <JEEB> it's a demuxer
[20:20:13 CEST] <JEEB> decoders are what you set with -c , demuxers are -f . (and encoders|muxers after -i)
[20:20:14 CEST] <james999> ok it is l isted
[20:20:51 CEST] <james999> so if image2 is available why doesn't my command above work?
[20:21:23 CEST] <JEEB> because image2 has different features depending on how it was built as noted by ChocolateArmpits
[20:21:36 CEST] <JEEB> if the OS doesn't support glob() (hint: windows doesn't)
[20:21:54 CEST] <JEEB> although not sure of the second example without globbing
[20:21:58 CEST] <JEEB> (´4@)
[20:22:22 CEST] <JEEB> also get used to actually posting into pastebin or anything similar the command & terminal output, and then linking that
[20:22:32 CEST] <JEEB> otherwise it's pretty hard to grasp where anything is going wrong :P
[20:24:46 CEST] <james999> alright well i at least figured out how to display one image: ffmpeg -loop 1 -i sicao_green_tunnel-wallpaper-1280x720.jpg -t 15 -f mpegts udp://ip-to-thing
[20:27:49 CEST] <JEEB> I will say that by the sound of it you probably don't want to use ffmpeg.c but rather the APIs directly
[20:28:06 CEST] <JEEB> that would let your application completely control the logic of how to show images etc
[20:28:28 CEST] <james999> JEEB: well that would require going back to nmake and cl and gcc on windows
[20:28:28 CEST] <JEEB> generate/override PTS and so forth
[20:28:33 CEST] <james999> and that's very scary
[20:29:39 CEST] <JEEB> I don't think MS's nmake would work, GNU make or something similar is required methinks
[20:29:59 CEST] <furq> please don't say "methinks"
[20:30:03 CEST] <furq> thanks
[20:30:17 CEST] <JEEB> and if you want to use MSVC then I don't think GCC is required
[20:30:21 CEST] <furq> james999: if you have a linux vm then it's easier to cross compile
[20:30:33 CEST] <JEEB> since --toolchain msvc should IIRC already switch to cl/link for everything
[20:32:16 CEST] <james999> now i'm really confused
[20:32:31 CEST] <james999> i can view the image -> udp stream locally on the PC, but when I go to vlc on the xbox it doesn't display
[20:33:03 CEST] <james999> it's same as above command but i added -vcodec libx264
[20:33:25 CEST] <JEEB> have you verified that you can use UDP with xbone vlc
[20:33:38 CEST] <james999> yes
[20:34:06 CEST] <james999> after much pain and torment. :)
[20:34:27 CEST] <james999> JEEB: well I was following some obscure instructions to compile nginx on windows with MSYS and rtmp module
[20:34:37 CEST] <JEEB> lol
[20:34:38 CEST] <james999> and I had to rename the gnu perl.exe to perl_UNUSED.exe so it would use windows perl
[20:34:47 CEST] <furq> yeah this is why i cross compile everything
[20:34:52 CEST] <furq> mingw/msys are a fucking mess
[20:34:57 CEST] <james999> and it also used nmake and cl.exe from windows but it was from inside the MSYS shell
[20:35:16 CEST] <james999> ok maybe i should learn to cross compile on linux then it sounds better lol
[20:35:21 CEST] <JEEB> it's not the FFmpeg docs on the wiki etc are perfect but IIRC the MSVC compilation docs for WinRT (as they're mostly shared with normal MSVC builds)
[20:35:24 CEST] <JEEB> are pretty OK
[20:35:34 CEST] <JEEB> and yes, if you're OK with mingw-w64 then you can just cross-compile
[20:35:42 CEST] <ChocolateArmpits> MSYS2 + MINGW is the easiest way to compile ffmpeg imo
[20:36:24 CEST] <james999> so you would prefer compiling ffmpeg on linux for windows or MSYS for windows or visual c++ for windows ChocolateArmpits?
[20:36:38 CEST] <BtbN> you still need msys or cygwin to build with msvc
[20:36:45 CEST] <ChocolateArmpits> msys+mingw
[20:36:56 CEST] <ChocolateArmpits> You're already on Windows, why migrate
[20:36:57 CEST] <BtbN> but it's not really any difference in complexity
[20:37:08 CEST] <BtbN> just a slightly changed configure line
[20:37:29 CEST] <furq> the complexity is getting msys in a fit state to build anything
[20:37:39 CEST] <furq> which i've traditionally had very bad luck with
[20:37:43 CEST] <JEEB> msys2 at least has simplified that
[20:37:55 CEST] <JEEB> since the packages aren't effing ancient any more
[20:38:16 CEST] <furq> msys2 seemed to have the potential to be better but i've managed to fuck up two msys2 installs by running pacman
[20:38:27 CEST] <furq> so i'm not wasting any more of my time on that
[20:38:29 CEST] <JEEB> sure, nothing's perfect
[20:38:44 CEST] <JEEB> also if you already have a *nix env
[20:38:47 CEST] <furq> yeah i do
[20:38:54 CEST] <JEEB> it's actually quicker to build there
[20:39:04 CEST] <JEEB> due to shell configure taking ages
[20:39:06 CEST] <furq> and yeah that too
[20:39:34 CEST] <furq> i can build 20 deps and ffmpeg from scratch before configure is done under msys
[20:47:19 CEST] <james999> ok i think i'm getting somewhere
[20:47:36 CEST] <james999> the udp video streaming to the xbox and phone didn't work with vlc until i put pkt_size=1316 on the ffmpeg command line
[20:47:48 CEST] <james999> and it ALSO made it so the image jpeg I was trying to do over mpegts succeeded
[20:47:57 CEST] <james999> i think my network doesn't allow large packets over wireless or somethign?
[20:49:33 CEST] <kepstin> the larger the packet, the more likely it is to fail to send over wireless
[20:51:06 CEST] <james999> well after 2 hours of fiddling with things i have now learned that fact
[20:51:09 CEST] <james999> so... yay
[20:51:12 CEST] <geforce> hey... I have a question... I want to decode DTS-HD MA to LPCM (wav)... Can ffmpeg that completely loseless ?
[20:55:14 CEST] <kepstin> geforce: I think ffmpeg can't handle the lossless parts of the master audio extensions right now, but I'd have to double check. my knowledge of that's kind of old.
[20:56:02 CEST] <geforce> Hm... I'm searching for a solution to do that... Any suggestions ?
[20:58:07 CEST] <klaxa> james999, i told you, like 2 hours ago ;) ><klaxa> udp without any application level sanity checks over wifi will most likely result in lost packets
[21:07:26 CEST] <geforce> Anyone here to help me to decode DTS-HD MA to LPCM (Wav) loseless ?
[21:08:12 CEST] <kepstin> huh, it might actually work if you're using ffmpeg 3.1 or later
[21:11:05 CEST] <geforce> Okay so how I can verify that it is really loseless ?
[21:11:23 CEST] <JEEB> only by having the source it was encoded from
[21:11:55 CEST] <JEEB> unless there's some checksum for the source audio in that stream
[21:12:01 CEST] <geforce> Yes I have the source I'm personally encode the DTS-HD MA track to LPCM (WAV)...
[21:12:20 CEST] <geforce> okay how does it work with this "checksum" ? How I can verify ?
[21:12:25 CEST] <JEEB> if you encoded the DTS-HD MA track yourself then it should be relatively simply :P
[21:12:30 CEST] <JEEB> *simple
[21:12:40 CEST] <JEEB> geforce: the checksum thing was an "if there is one"
[21:12:47 CEST] <JEEB> if there's not then nope
[21:13:10 CEST] <JEEB> if you don't have the raw source audio then you cannot possibly verify that the decode was lossless
[21:13:15 CEST] <geforce> Hm, so not every track has this Checksum ?
[21:13:32 CEST] <kepstin> not every lossless audio codec does.
[21:13:34 CEST] <JEEB> I don't even know if DTS corp added such stuff :P
[21:14:02 CEST] <JEEB> anyways, if you did not encode that audio track there is no way to know if the decode was lossless
[21:14:11 CEST] <kepstin> it's all reverse-engineered, yeah, so hard to tell
[21:14:13 CEST] <JEEB> or well, rather than lossless I'd say "correct"
[21:14:18 CEST] <JEEB> kepstin: well even if it wasn't
[21:14:26 CEST] <JEEB> if you just have someone's decoder for the spec
[21:14:31 CEST] <JEEB> it can just as well have a bug
[21:14:38 CEST] <JEEB> you cannot possibly verify it
[21:15:35 CEST] <JEEB> geforce: INPUT->ENCODED BIT STREAM->DECODED OUTPUT , you require the first two ones to verify things
[21:15:40 CEST] <utack> is it possible that zscale converts some colorspaces incorrectly? or is it more likely that hte source has some kind of problem
[21:16:00 CEST] <JEEB> utack: the zscale filter can also have a bug - or anything else like that
[21:16:25 CEST] <JEEB> everything is possible so you start scratching things off
[21:16:28 CEST] <utack> problem is i have no refernece, but colors do seem pretty wrong to me in the result. i will try to grab a sample and file a bug
[21:17:49 CEST] <geforce> JEEB: so for understanding... I'm the one who encodes the DTS-HD MA Track to LPCM (Wav) and want to verify that the LPCM Track is loseless to the DTS-HD MA Track... So can you tell me for me as inexperienced user to do this ?
[21:18:22 CEST] <JEEB> geforce: encoding is taking raw input and coding it into something
[21:18:41 CEST] <JEEB> decoding is taking coded input and decoding it into raw output
[21:18:42 CEST] <kepstin> geforce: if you don't have the original source that was used to make the DTS-HD MA track, then the best you can do is to try decoding it with a different "reference" decoder and comparing the result
[21:19:02 CEST] <JEEB> kepstin: and with DTS the decoders can do random crapola in addition to actual decoding so bleep that
[21:19:11 CEST] <kepstin> (and I don't know where you'd find that other decoder)
[21:19:13 CEST] <JEEB> at this point I would just trust the dca decoder
[21:19:25 CEST] <JEEB> because the libdcadec guy is handling it
[21:19:50 CEST] <geforce> Okay the libdcadec is implemented in FFMPEG ?...
[21:20:13 CEST] <JEEB> libdcadec became what is now the dca decoder in FFmpeg :P
[21:20:23 CEST] <JEEB> and the libdcadec maintainer became the FFmpeg dca decoder maintainer :P
[21:20:32 CEST] <geforce> So I use ./ffmpeg -i dts-hd-ma-input.dts lpcm-output.wav ???
[21:20:41 CEST] <JEEB> yes
[21:20:44 CEST] <kepstin> yeah, it looks like ffmpeg's been able to decode dts-hd ma correctly since version 3.0 (which added support for external libdcaenc), and 3.1 (where the internal dca decoder is based on libdcaenc)
[21:21:14 CEST] <durandal_1707> libdcadec
[21:21:22 CEST] <kepstin> er, no, that's wrong - ver 3.0 is the first with the full internal encoder :)
[21:21:57 CEST] <geforce> Okay so not libdcadec in version 3.0 ?
[21:22:26 CEST] <kepstin> support for external libdcadec was added in ... 2.7 and removed in 3.1 (because it was redundant)
[21:22:39 CEST] <kepstin> geforce: just use ffmpeg 3.0 or later, and it should be fine.
[21:22:54 CEST] <geforce> Ok but ffmpeg does encode all tracks as 16bit, but i've some 24bit tracks...
[21:23:22 CEST] <kepstin> geforce: unless you specify otherwise, ffmpeg converts to s16le pcm in wav.
[21:23:52 CEST] <JEEB> also FFmpeg has no 24bit audio format so you will get 8 extra bits in your 24bit samples
[21:23:55 CEST] <JEEB> but that is still lossless
[21:24:10 CEST] <JEEB> (as the actual data doesn't change)
[21:24:27 CEST] <kepstin> geforce: so if you want higher bit depth, you have to specify otherwise (e.g. by using "-c:a pcm_s24le")
[21:24:56 CEST] <kepstin> ffmpeg stores samples in 32bit ints internally, but it supports 24bit audio, and can write 24bit pcm in wav, i'm pretty sure...
[21:25:03 CEST] <JEEB> ok
[21:25:24 CEST] <JEEB> then that should be OK as long as FFmpeg doesn't try to do anything fancy with that 24bit in 32bit decoded output :)
[21:26:12 CEST] <geforce> Okay I will try that. but what does the "-c:a" mean ? c for codec and a for audio ?
[21:26:38 CEST] <JEEB> yes
[21:27:35 CEST] <geforce> Okay, another silly question ... I want to use a built for windows... Its recommended to use Version "20170425-b4330a0" or Version 3.2.4 ?
[21:33:12 CEST] <kepstin> geforce: shouldn't make a difference for this use case. The git snapshot ("20170425-b4330a0") will be newer, but might contain experimental/less tested stuff& but it's usually fairly stable.
[21:33:34 CEST] <JEEB> geforce: use fate.ffmpeg.org as a stick
[21:33:40 CEST] <JEEB> that tests the current master
[21:33:58 CEST] <JEEB> so you can check if the FATE tests pass on various operating systems and architectures
[21:34:31 CEST] <geforce> Great, thank you
[21:35:36 CEST] <cryptodechange> Any advice on the best method for building ffmpeg with x264 10-bit along side regular ffmpeg? (linux)
[21:35:51 CEST] <cryptodechange> Currently have my build in /usr/local
[21:36:14 CEST] <JEEB> having separate ffmpeg binaries is IMHO the best way to do that
[21:36:25 CEST] <JEEB> as I don't think you can only switch the libx264 library
[21:36:34 CEST] <furq> doesn't LD_LIBRARY_PATH work
[21:36:51 CEST] <furq> or LD_PRELOAD
[21:36:52 CEST] <JEEB> well it would cause it to load but does that actually work :P
[21:36:56 CEST] <furq> idk i've never tried it
[21:37:00 CEST] <furq> but i've seen other people say it works
[21:37:07 CEST] <JEEB> I thought libavcodec definitions would also have to change
[21:37:08 CEST] <JEEB> etc
[21:37:11 CEST] <cryptodechange> so compile x264 with 10-bit into /tmp, and use that when building ffmpeg
[21:37:15 CEST] <JEEB> but sure (´4@)
[21:37:22 CEST] <cryptodechange> But the question is, how should I use --prefix when compiling?
[21:37:38 CEST] <JEEB> yes
[21:37:42 CEST] <JEEB> prefix is where you install
[21:37:43 CEST] <cryptodechange> I could just do --prefix=/opt
[21:37:50 CEST] <furq> what, with x264?
[21:37:55 CEST] <cryptodechange> ffmpeg
[21:38:06 CEST] <JEEB> so binaries go into prefix/bin and libraries into prefix/lib
[21:38:07 CEST] <furq> oh right you mean for separate binaries
[21:38:23 CEST] <cryptodechange> Can the 10bit version of libx264 be trashed once ffmpeg is done?
[21:38:42 CEST] <furq> only if you do a full static build
[21:38:51 CEST] <cryptodechange> --enable-static
[21:38:51 CEST] <furq> --extra-ldflags=-static
[21:39:03 CEST] <furq> enable-static just builds static libav*.so
[21:39:13 CEST] <furq> it'll still link against dynamic libs for external deps
[21:39:14 CEST] <JEEB> or if you give it to x264 configure, static libx264 :P
[21:39:18 CEST] <furq> that works too
[21:39:21 CEST] <geforce> JEEB: I've now try to transcode a dts-hd ma with 24bit to wav with 24bit and than a message appears after working "Filesize 6944219238 invalid for wav, output file will be broken" Can I ignore it ?
[21:39:49 CEST] <JEEB> geforce: if the thing you are going to open it with supports >4 gigabyte WAVs :P
[21:39:53 CEST] <furq> but if you're building two separate libs you probably want them to be fully static so you can stick them both in /usr/local
[21:39:57 CEST] <furq> two separate binaries, rather
[21:40:14 CEST] <furq> otherwise things will get messy
[21:40:16 CEST] <geforce> OK i try
[21:41:01 CEST] <cryptodechange> furq, how can I build 2 ffmpegs in /usr/local without them replacing each other?
[21:41:10 CEST] <cryptodechange> I was just going to do the following
[21:42:12 CEST] <cryptodechange> x264: ./configure --prefix=/opt/10bit --enable-static --enable-shared --bit-depth=10
[21:43:05 CEST] <matej_k> avcodec_flush_buffers doesnt seems work for svq3, it still returns old frame in next avcodec_decode_video2, is that expected?
[21:44:15 CEST] <cryptodechange> ffmpeg: ./configure --prefix=/opt/10bit --enable-gpl --enable-shared --enable-libx264
[21:44:35 CEST] <cryptodechange> So when I want to encode 10bit, I just run /opt/10bit/bin/ffmpeg (i think)
[21:51:27 CEST] <thebombzen> cryptodechange: I have 8bit libx264 installed from my package manager, but I built 10bit libx264 myself in /usr/local/lib
[21:51:44 CEST] <thebombzen> I have my system configured so it'll link to /usr/local/lib first
[21:51:59 CEST] <thebombzen> so when I use ffmpeg, it loads /usr/local/lib/libx264.so which contains the 10bit x264 code
[21:52:21 CEST] <thebombzen> however, if I run LD_PRELOAD=/usr/lib/libx264.so ffmpeg
[21:52:25 CEST] <thebombzen> it'll link to the 8bit libx264
[21:52:36 CEST] <furq> right
[21:52:37 CEST] <furq> that's easier
[21:52:46 CEST] <furq> you should do that, but for completeness' sake
[21:53:02 CEST] <cryptodechange> Using debian, everything is out of date
[21:53:07 CEST] <furq> i meant build your 10-bit ffmpeg with --enable-static --extra-ldflags=-static
[21:53:12 CEST] <furq> then you just get one static binary
[21:53:22 CEST] <furq> then call that /usr/local/ffmpeg-10bit or something
[21:53:29 CEST] <furq> er, /usr/local/bin/ffmpeg-10bit
[21:53:30 CEST] <thebombzen> basically, if you dynamically link libx264, you can swap in either the 8bit or the 10bit version at runtime
[21:53:37 CEST] <thebombzen> using LD_PRELOAD
[21:53:49 CEST] <cryptodechange> On one ffmpeg install?
[21:53:52 CEST] <furq> yes
[21:54:03 CEST] <cryptodechange> That certainly sounds easier
[21:54:03 CEST] <thebombzen> >static ffmpeg
[21:54:10 CEST] <thebombzen> ew
[21:54:15 CEST] <thebombzen> I generally prefer to leave it on 10bit unless I explicitly need 8bit though
[21:54:29 CEST] <furq> LD_LIBRARY_PATH or LD_PRELOAD will both load those dirs/libs first
[21:54:42 CEST] <cryptodechange> I was planning on 10bit my anime library, because, uh, I have no idea
[21:54:51 CEST] <cryptodechange> It seems that's the trend
[21:54:53 CEST] <cryptodechange> haha
[21:55:05 CEST] <thebombzen> eh
[21:55:13 CEST] <furq> if you're not also downscaling or something then don't bother
[21:55:16 CEST] <thebombzen> but then you don't have dynamic linking of libav*.so
[21:55:44 CEST] <furq> thebombzen: if that's addressed at me then i wasn't talking to you
[21:55:54 CEST] <furq> i have no opinion about static vs dynamic ffmpeg
[21:56:03 CEST] <furq> other than static is generally less hassle if you're building it yourself
[21:56:21 CEST] <cryptodechange> Well, no bluray source uses 10bit, so is it literally saving on file size?
[21:56:30 CEST] <furq> yes
[21:56:55 CEST] <cryptodechange> Color banding, etc doesn't really get addressed, as if its there, it was probably in the source too
[21:57:08 CEST] <furq> but it probably won't if you're just going from 8-bit x264 to 10-bit x264 at the same resolution and similar settings
[21:57:10 CEST] <cryptodechange> And I primarily use Plex, so it'd transcode in most cases
[21:57:41 CEST] <furq> obviously if you're resizing then it could potentially look better
[21:58:23 CEST] <furq> emphasis on potentially
[21:58:42 CEST] <cryptodechange> So going back to the compiling, whether it's two x264 or a static ffmpeg
[21:58:57 CEST] <cryptodechange> How would I compile it along side the same prefix, but append 10bit to the bin?
[21:59:02 CEST] <thebombzen> furq: on a linux system you generally will have libav* loaded already
[21:59:03 CEST] <cryptodechange> e.g. ffmpeg and ffmpeg10bit
[21:59:36 CEST] <furq> if it's a full static build then all you need is the binary, so just don't run make install
[21:59:44 CEST] <furq> and move the binary to where you want it
[21:59:49 CEST] <furq> but yeah don't do that if LD_PRELOAD works
[22:00:12 CEST] <cryptodechange> LD_PRELOAD=/usr/lib/libx264.so
[22:00:21 CEST] <thebombzen> 10bit encoding has better filesize than 8bit because the predictors are more accurate, even if scaled from 8bit. but that's about it
[22:00:34 CEST] <furq> alias ffmpeg-10bit=LD_PRELOAD=/path/to/10bit/libx264.so ffmpeg
[22:00:44 CEST] <cryptodechange> If I compile 10bit first, I could do 'mv libx264.so libx26410bit.so'?
[22:00:49 CEST] <thebombzen> your display can only draw 8bit anyway so i'll decrease it down to 8 when you view it, even with a 10bit source.
[22:01:09 CEST] <furq> isn't there something to do with hardcoded dithering as well
[22:01:17 CEST] <furq> which is why it's particularly popular for animes
[22:01:26 CEST] <thebombzen> hardcoded dithering?
[22:01:31 CEST] <furq> at 8-bit
[22:01:37 CEST] <thebombzen> I hadn't heard anything about that
[22:02:01 CEST] <thebombzen> I don't know why it would given that the original source is 8bit depth
[22:03:28 CEST] <geforce> JEEB: It works with the Bose Lifestyle 650 and popcorn Hour a400.
[22:03:30 CEST] <cryptodechange> So I could just move the 10-bit /usr/lib/libx264.so to /usr/lib/10-bit/libx264.so?
[22:03:34 CEST] <cryptodechange> Or would that break things?
[22:04:13 CEST] <geforce> JEEB: But a last question in ffmpeg it shows the dts source s32p (24bit) and the destination wav file s32 (24 bit) what does the "p" mean ?
[22:04:44 CEST] <furq> cryptodechange: any of these things will work fine
[22:06:22 CEST] <cryptodechange> furq: crf has different ranges for 10-bit right? so a 8-bit crf=16 would be higher/lower
[22:06:32 CEST] <thebombzen> I don't think so
[22:11:56 CEST] <JEEB> cryptodechange: the range is different but generally the range is added to the negative side so the same CRF should give at least on some level similar results on 8,9,10bit
[22:12:14 CEST] <JEEB> (crf zero won't be lossless with >8bit because of that)
[22:13:05 CEST] <furq> in my experience 20 is about the same with both
[22:13:27 CEST] <furq> also bear in mind you lose a lot of compat with 10-bit
[22:18:24 CEST] <sinanksu> Hi
[22:21:18 CEST] <thebombzen> furq: not as much as you'd think
[22:21:35 CEST] <thebombzen> you lose compat with many old pieces of hardware. but with software players and most websites it works fine
[22:23:12 CEST] <klaxa> anime nerds will like you everyone else will struggle to notice the difference even
[22:23:32 CEST] <JEEB> well it does compress better, lately one of the streaming guys even noticed it
[22:23:41 CEST] <JEEB> faster preset + 10bit was better than slower preset + 8bit
[22:23:52 CEST] <JEEB> the negative side of course is that hw decoders barf with 10bit AVC
[22:24:00 CEST] <JEEB> so you can't really use it even if it is better
[22:24:03 CEST] <JEEB> (´4@)
[22:24:19 CEST] <JEEB> at least as long as you are aiming for mobile devices or STBs etc without proper SW decoders
[22:24:27 CEST] <JEEB> (or the perf to run that SW decoder)
[22:30:07 CEST] <geforce> Anyone who can explan the difference between s32p (24bit) and s32 (24bit) audio ?
[22:31:18 CEST] <BtbN> one is planar, the other isn't
[22:36:31 CEST] <cryptodechange> talking of anime, there are so many conflicting pieces of advice on what -x264-params to use
[22:36:44 CEST] <cryptodechange> particularly with aq, psy and deblocking
[22:36:58 CEST] <geforce> okay but what is planar ?... I dont understand the word ...
[22:37:28 CEST] <furq> cryptodechange: -tune animation
[22:37:54 CEST] <cryptodechange> which is deblock=1,1:psy-rd=0.4,0:aq-strength=0.6
[22:38:25 CEST] <furq> probably
[22:38:31 CEST] <furq> there's no reason to know what it does
[22:38:44 CEST] <cryptodechange> but because of lines/edges some comments claim deblock=-1,-1 is better, and aq-strength is better higher/lower
[22:39:21 CEST] <sfan5> furq: are you srs about -tune animation
[22:39:42 CEST] <geforce> BtbN: is there any quality difference with s32p and s32 ?
[22:40:13 CEST] <BtbN> it's an uncompressed format, how should there be any quality loss?
[22:41:04 CEST] <sfan5> couldn't find a better source but
[22:41:13 CEST] <sfan5> https://encodingwissen.de/codecs/x264/referenz/ says -tune animation is for classical cartoons
[22:41:16 CEST] <sfan5> which anime is certainly not
[22:41:56 CEST] <kepstin> sfan5: that's probably either a translation or research error
[22:42:34 CEST] <klaxa> isn't it more about 2D vs 3D content?
[22:42:45 CEST] <sfan5> hm https://app.zencoder.com/docs/api/encoding/h264/tuning says something else
[22:42:48 CEST] <kepstin> the "animation" tune in ffmpeg adds extra reference frames, which helps a lot with animations loops (used more in anime than in trad. animation even...) and does some deblocking tweaks to make flat shading look better
[22:43:27 CEST] <furq> cryptodechange: who do you trust more, the x264 authors or some guy commenting about animes on the internet
[22:43:48 CEST] <furq> whose username is "actually,"
[22:48:21 CEST] <sfan5> [Coalgirls] seem to use something close to tune=animation with a different psy-rd (1.00:0.00)
[22:50:53 CEST] <sfan5> [DameDesuYo] seem to use something close to tune=animation with different deblock (1:-2:-2)
[22:51:19 CEST] <cryptodechange> HorribleSubs uses default tune=animation
[22:51:44 CEST] <sfan5> i would'nt trust the people at crunchyroll to make good encodes
[22:53:12 CEST] <klaxa> horriblesubs doesn't do shit
[22:53:15 CEST] <klaxa> they rip of CR
[22:53:46 CEST] <klaxa> *off
[22:53:47 CEST] <sfan5> [FFF] also uses something that resembles -tune animation
[22:53:51 CEST] <sfan5> looks like I was wrong
[22:54:29 CEST] <cryptodechange> KamiFS (bluray mux) uses deblock=-1,-1, psy_rd=0.80:0.20 and some weird combo for aq
[22:56:57 CEST] <kepstin> also, keep in mind that one of the main devs on x264, DarkShikari, is an anime fan. so.
[22:57:23 CEST] <cryptodechange> Though for older anime sources, where its on the fuzzy side and grainy
[22:57:35 CEST] <sfan5> surely he'd rename the preset to "anime" or did the other devs not want that :P
[22:57:48 CEST] <cryptodechange> I assume I would wanna mix up the settings, so it's in between grain and animation?
[22:58:07 CEST] <sfan5> why
[22:58:16 CEST] <cryptodechange> For older, less clean sources
[22:58:41 CEST] <sfan5> usually you look at your source instead of trying to find "one size fits all" settings
[22:58:42 CEST] <cryptodechange> e.g. DVD of anime from the 90s
[22:59:05 CEST] <klaxa> sfan5: there is (was?) a touhou preset :P
[22:59:18 CEST] <kepstin> klaxa: still is. it's undocumented tho, an easter egg.
[22:59:26 CEST] <sfan5> huh
[22:59:27 CEST] <klaxa> ah, that's why i can't find it
[22:59:33 CEST] <cryptodechange> also, I'm using the following command to check for deinterlacing
[22:59:41 CEST] <cryptodechange> talking about old content
[22:59:44 CEST] <cryptodechange> ffmpeg -i input.mkv -filter:v idet -frames:v 1000 -an -f rawvideo -y /dev/null
[23:00:07 CEST] <cryptodechange> where i compare the numbers to see if it's deinterlaced or not
[23:00:13 CEST] <cryptodechange> Is there a better method?
[23:00:16 CEST] <kepstin> the amount of filtering you need to clean up a 90s anime dvd makes preserving grain kind of useless. Now, if you're encoding a modern BD rescan from film source of a 90s anime, then you might want to preserve grain :)
[23:00:20 CEST] <sfan5> interlaced anime exists?
[23:00:36 CEST] <kepstin> telecined, stored in interlaced video, yes
[23:00:43 CEST] <cryptodechange> I have a DBGT dvd set which is apparently
[23:01:10 CEST] <sfan5> oh right dvds are still a thing
[23:01:30 CEST] <kepstin> actual interlaced anime is non-existent, but a lot of early 2000s stuff had video editing done on top of the telecined animation, adding interlaced pans/fades/effects.
[23:01:41 CEST] <kepstin> so... yeah, it's a mess
[23:01:56 CEST] <cryptodechange> the moar i kno
[23:02:02 CEST] <durandal_1707> just do rm
[23:02:22 CEST] <kepstin> (the opening video on the serial experiments lain blu-rays has some really poorly done progressive upscale of interlaced video in a few places, it's rather annoying)
[23:02:32 CEST] <sfan5> trying to fix the mistakes of the people who made your source material is a world of pain
[23:03:09 CEST] <cryptodechange> Should -vf yadif suffice?
[23:04:07 CEST] <cryptodechange> running the idet filter on some source gives out a high number of TFF and BFF frames, and low progressive
[23:04:43 CEST] <cryptodechange> https://gist.github.com/aktau/6660848
[23:08:06 CEST] <ChocolateArmpits> cryptodechange, first you should check if it's actually interlaced or telecined, or maybe just field shifted then apply appropriate filter for that task
[23:08:30 CEST] <ChocolateArmpits> I wouldn't touch a telecined source with a typical deinterlacer
[23:08:57 CEST] <furq> the better method of seeing if something is interlaced is to look at it
[23:09:08 CEST] <furq> with your eyes
[23:29:45 CEST] <Michaela_> how can i convert mono8 (packed) raw to yu420p?
[23:35:04 CEST] <Mista_D> changing a fourcc vtag from "AVC1" to "H264".  Get "incompatible codec id 28", I thought AVC and H264 are freely interchangable...  Any advice please?  https://pastebin.ca/3803353
[23:38:36 CEST] <Michaela_> how can i convert mono8 (packed) raw to yu420p? can i use YUV400A?
[23:39:31 CEST] <sfan5> Mista_D: the AVC1 tag does not seem to be used in the source of ffmpeg anywhere
[23:40:29 CEST] <Mista_D> sfan5: possibly in libx264, but FFmpeg shodul allow to modify it when -c copy'ing ?
[23:40:42 CEST] <Mista_D> *should
[23:41:14 CEST] <sfan5> it should but seems like it doesn't allow tags it doesn't know
[23:41:24 CEST] <sfan5> try adding it to ./libavformat/riff.c maybe
[23:53:28 CEST] <kepstin> Mista_D: what tood do you have which is making files with this "AVC1" fourcc? is this in avi/riff or some other container?
[23:53:34 CEST] <kepstin> what tool*
[23:55:59 CEST] <Mista_D> kepstin: FFmpeg encoded. Its here : ffprobe VID.mp4 -show_streams -select_streams v 2>&1|grep codec_tag_string
[23:56:54 CEST] <kepstin> Mista_D: so why are you changing anything? ffmpeg should default to doing the correct thing...
[23:57:06 CEST] <holgersson> klaxa: With '-c:v libx264 -preset veryslow -crf 25' I was able to shrink the filesize from 16GB to 2.4GB plus some 100MB for Audio. It was slow however (33h acording to time). Thanks again to you and the others!
[23:57:55 CEST] <Mista_D> kepstin: my player hardware wants to see fourcc set to H264, so I'm looking to change it without re-encoding...
[23:58:36 CEST] <klaxa> glad it worked out as you hoped :)
[00:00:00 CEST] --- Thu Apr 27 2017