[Ffmpeg-devel-irc] ffmpeg.log.20171123

[burek]

[00:33:55 CET] <TheRock> is it actually possible
[00:34:04 CET] <TheRock> to compress/change the bitrate to reduce the size
[00:34:08 CET] <TheRock> for an incomplete mp4 file?
[00:34:17 CET] <TheRock> Like, if i have only 10% of the entire video
[00:34:34 CET] <TheRock> that i start converting it to a lower bitrate to reduce the size
[00:34:39 CET] <TheRock> or must the file be completed first
[00:40:22 CET] <DHE> mp4 tends to be a pain for transcoding in general unless the file was originally muxed for it with fastmoov and stuff
[01:11:30 CET] <CCFL_Man> does -vcodec copy copy the video codec from the input to the output without any transcoding?
[01:11:51 CET] <CCFL_Man> err, video stream
[01:12:45 CET] <CCFL_Man> because i have a dvd vob that i merged. i just want to copy the video stream and one of the audio streams without any transcoding
[01:16:08 CET] <CCFL_Man> stream #0.0 is the video and stream #0.1 is the audio stream i need
[01:16:44 CET] <furq> CCFL_Man: yes
[01:16:50 CET] <furq> just -c copy will copy all selected streams
[01:18:53 CET] <CCFL_Man> furq: why does it say there in an encoder then while i run it?
[01:19:04 CET] <CCFL_Man> Lavf57
[01:27:40 CET] <CCFL_Man> the map option is kind of confusing
[01:33:19 CET] <CCFL_Man> Output #0, ty, to 'SiliconValley.ty':
[01:33:19 CET] <CCFL_Man>   Metadata:
[01:33:20 CET] <CCFL_Man>     encoder         : Lavf52.64.2
[01:42:59 CET] <CCFL_Man> i think i figured it out
[02:14:44 CET] <CCFL_Man> i converted from vob to the ty container for my series 1 directivo and i'm uploading it via mfs_ftp
[02:16:39 CET] <CCFL_Man> it writes at a rate of 150KiB/s, heh
[02:27:25 CET] <zash> What was the pixel format that worked well with current browsers?
[02:27:49 CET] <zash> yuv420p?
[02:28:48 CET] <DHE> yuv420p is the only thing that should be considered widely supported
[02:30:13 CET] <echelon> is there a way to rename the subtitle/audio track stream labels?
[02:32:43 CET] <echelon> does ffmpeg automatically copy all audio streams?
[02:32:56 CET] <echelon> from dvd->mkv for example
[02:37:23 CET] <pyBlob> I've got a "video-stream" that generates a new image every 10s, is there a simple way to call an external programm on every frame when using a format like jpeg/png?
[03:14:13 CET] <SortaCore> PyPlob: isn't that just MJPEG?
[03:15:04 CET] <SortaCore> ah he's gone
[08:56:10 CET] <esc-reality> Is it possible to lower only the video bitrate to i.e. 6000kbps for .mkv files in one command?
[09:04:47 CET] <Nacht> esc-reality: ffmpeg -i <input> -c:v libx264 -b:v 6M -c:a copy -f mkv output.mk4
[09:07:02 CET] <Nacht> the -b:v command stands for videobitrate. the -c:v libx264 selects the H264 codec
[09:15:37 CET] <esc-reality> Thank you!
[09:33:36 CET] <termos> I'm having the same issue as described here: https://stackoverflow.com/questions/45363566/ffmpeg-decoding-too-slow-avcodec-send-packet-avcodec-receive-frame is the new send/receive API slower than the deprecated avcodec_decode_video2?
[09:53:31 CET] <JEEB> termos: no, I haven't checked that ticket but it should be exactly the same. a lot of applications have switched to the new APIs and we'd have a lot of regressions if it was actually slower :)
[12:23:51 CET] <jam__> We wish to convert WAV files to AAC (in a m4a container) - I noticed we have to compile aac support in- do we need a licence for this codec?
[12:34:17 CET] <sfan5> the aac encoder within ffmpeg is released under LGPL (same as the most code in ffmpeg)
[12:34:25 CET] <sfan5> no idea about patents if that's what you want to know
[12:36:12 CET] <Fyr> guys, it seems like, when converting 1000 fps into 60 fps, FFMPEG drops unnecessary frames and writes only every 900 or 1000 frame.
[12:36:45 CET] <Fyr> is it possible to make FFMPEG avoid omitting frames?
[12:39:49 CET] <termos> JEEB: hm okey thanks, I'm doing some refactoring now. Trying to call avcodec_send_packet and avcodec_receive_frame from different threads. I'm getting some weird decoding artefacts though. Is that not suppose to be supported? I'm locking the codec context as well as registering a lock manager with av_lockmgr_register
[12:43:12 CET] <theperfectpunk> I'm running Windows 10 64bit, whenever i try to export video frames to png, ffmpeg crashes
[12:44:26 CET] <theperfectpunk> I tested it on a mac and it exported video frames fine, can anybody point what am i doing wrong on windows?
[12:44:39 CET] <theperfectpunk> *i am
[12:45:56 CET] <theperfectpunk> The source video is h264 avc1 high with yuv420p pixels
[12:47:06 CET] <theperfectpunk> 1080x1080, 23.98 fps, command used : ffmpeg -i out.mov -r 24/1 rocketframes\file%03d.png
[12:49:32 CET] <theperfectpunk> pastebin: https://pastebin.com/7YZyj3Q1
[13:18:28 CET] <wazzBate> hello all, I'm having a bit of a problem parsing output to a stream correctly. I have it succesfully saving the video to an file but it will only stream the first input video (unfiltered) instead of how its supposed to look with all three inputs.
[13:18:28 CET] <wazzBate> The output file appears correctly. https://pastebin.com/qwtuvRHy
[15:49:37 CET] <esc-reality> ffmpeg -i <input> -c:v libx264 -b:v 6M -c:a copy -f mkv output.mk4
[15:49:56 CET] <esc-reality> Is it possible to do this only for files with a bitrate above a certain value?
[15:49:58 CET] <esc-reality> Automated
[15:51:22 CET] <sfan5> bash script
[15:52:11 CET] <sfan5> ffprobe can output json which should allow you to parse the video stream bitrate easily, then just compare that to your limit
[15:52:23 CET] <esc-reality> Yeah I know, it would be sort-off if[bitrate > 6M] then ... fi
[15:52:35 CET] <esc-reality> Ah good one, will try
[15:52:41 CET] <esc-reality> thx
[16:23:22 CET] <esc-reality> Got it :)
[16:23:50 CET] <esc-reality> ffprobe -i inputfile.mkv -v quiet -print_format json -show_format -hide_banner | jq -r '.format.bit_rate'
[16:24:11 CET] <esc-reality> returns only the bitrate value
[17:40:49 CET] <froo> much appreciation to the ffmpeg team.
[18:40:55 CET] <Fyr> guys, FFMPEG failed to malloc 147 Gb of my RAM.
[18:41:13 CET] <Fyr> is there a way to avoid using such huge sizes?
[18:42:59 CET] <SortaCore> @.@
[18:43:13 CET] <SortaCore> that sounds like a glitch
[18:43:32 CET] <TheRock> you can purchase ram for you system
[18:43:51 CET] <zash> or tweak the overcommit ratio
[18:43:52 CET] <Fyr> TheRock, what if FFMPEG needs more?
[18:44:03 CET] <TheRock> you can have up to 800 gb on some systems
[18:44:09 CET] <TheRock> it will be enough
[18:44:29 CET] <Fyr> will it?
[18:44:34 CET] <TheRock> yeah
[18:44:57 CET] <Fyr> why are you so sure?
[18:45:06 CET] <TheRock> because now you only need 147
[19:53:09 CET] <SortaCore> TheRock is giving terrible advice :p
[19:57:04 CET] <JEEB>  16
[20:06:13 CET] <SortaCore> how can you even call malloc with 16GB?
[20:06:27 CET] <SortaCore> doesn't it max out at 4GB?
[20:07:54 CET] <SortaCore> (how do you call it with 147GB I mean)
[20:08:13 CET] <kepstin> 64bit machines can handle that fine
[20:08:28 CET] <SortaCore> yea, but isn't the function definition a 32-bit unsigned int
[20:08:56 CET] <kepstin> no, it takes a size_t, which has variable length depending on machine architecture
[20:09:20 CET] <SortaCore> size_t varies huh
[20:18:01 CET] <doublya> I'm trying to use the -ec flag for error concealment for hevc decode.  I'm not seeing any effect and I've tried passing all three parms per the documentation
[20:22:40 CET] <sfan5> isn't EC enabled by default?
[20:23:09 CET] <kepstin> it is - but from a quick look at the code, it looks like the hevc encoder doesn't call into the generic error concealment code, so the option will have no effect
[20:28:51 CET] <JEEB> yea it has no ec
[20:31:23 CET] <doublya> Thanks guys! Which src file would be the place to look into implementing a call to error_resilience.c
[20:32:24 CET] <kepstin> I'd assume you'd start somewhere in hevcdec.c ... :)
[20:35:03 CET] <doublya> Thanks I'll take a look
[20:45:48 CET] <ccooc> why this command ffmpeg -i rtsp://x.x.x.x -i 2.264  -preset ultrafast -filter_complex "[0]crop=20:20:20:20[a];[a][1] overlay=0:0" http://127.0.0.1/feed.ffm  takes me 250% cpu ??? that crazy
[20:50:38 CET] <BtbN> video encoding needs CPU, indeed
[20:50:52 CET] <BtbN> And you should not be using ffserver, it's dead
[20:51:54 CET] <ccooc> so do you have another server for streaming?
[20:52:02 CET] <ccooc> free and working with ffmpeg ?
[20:52:51 CET] <ccooc> BtbN: i not found any another server , and what is problem with ffserver? preformance ?
[20:53:17 CET] <BtbN> Nobody maintains it, for years. It's only dragged along by now, nobody knows how to properly fix it.
[20:53:31 CET] <BtbN> Using it is a terrible idea. It's dead software
[20:54:22 CET] <ccooc> BtbN: so do you have recommencded on another server that can easily work with ffmpeg?
[20:54:33 CET] <BtbN> most people use nginx for rtmp
[20:54:45 CET] <ccooc> rtmp is rtsp ?
[20:55:02 CET] <BtbN> rtmp is rtmp
[20:55:17 CET] <ccooc> cani stream with that rtsp via udp/tcp ?
[20:55:27 CET] <BtbN> no, you stream rtmp with that
[20:56:36 CET] <ccooc> so how can i stream rtsp with ffmpeg?
[20:59:19 CET] <aiena> Can https://launchpad.net/~mc3man/+archive/ubuntu/trusty-media be used to install ffmpeg on any ubuntu version e.g. 17.10?
[20:59:41 CET] <BtbN> Why would you need a ppa for that?
[21:00:10 CET] <aiena> BtbN: on suse ffmpeg is partial I dont know if ubuntu also disables codecs etc.
[21:00:37 CET] <aiena> so the ffmpleg from the distro on suse was broken and I was notusre if I would discover the same thing on buntu
[21:01:56 CET] <aiena> is the ffmpeg the full ffmpeg on ubuntu versions?
[21:04:17 CET] <SonicTheHedgehog> aiena: https://packages.ubuntu.com/search?keywords=ffmpeg  Have a look
[21:04:32 CET] <ccooc> BtbN: so do you know another server to stream with ffmpeg (rtsp )?
[21:05:04 CET] <aiena> SonicTheHedgehog: not sure how to identify if it is the full ffmpeg on suseI discovered the distro version is practically useless by trial and error and building ffmpeg itself.
[21:06:12 CET] <aiena> SonicTheHedgehog: I have webm audio do I need some extra packages to convert it to mp3 or the ffmpeg base buntu package is enough
[21:06:52 CET] <aiena> ffmpeg seems to be converting it with the base package but very slowly the file is 122 MP and has about 2 ours worth of audio
[21:06:55 CET] <aiena> hours
[21:07:06 CET] <kepstin> the ffmpeg shipped in ubuntu has a reasonably complete set of codecs (most stuff that's gpl compatible), but it's kind of an old version.
[21:07:10 CET] <aiena> *122 MB
[21:07:16 CET] <SonicTheHedgehog> aiena: Well, on xenial (16.04LTS) for example, this is what's pulled from Ubuntu's repos, https://packages.ubuntu.com/xenial/ffmpeg   Note the "Download Source Package" on the right-hand side too
[21:07:28 CET] <aiena> SonicTheHedgehog: I am on 17.10
[21:07:29 CET] <SonicTheHedgehog> Yeah, it often isn't the latest stable
[21:07:43 CET] <SonicTheHedgehog> aiena: It was an example, you can look at your version. :P
[21:07:51 CET] <aiena> I know
[21:07:56 CET] <kepstin> looks like 17.10 has 3.2, that's not that bad.
[21:08:01 CET] <SonicTheHedgehog> Yeah
[21:08:02 CET] <aiena> 3.3.4-2
[21:08:14 CET] <aiena> is the version
[21:08:19 CET] <kepstin> ah, couldn't remember the codenames
[21:08:32 CET] <aiena> for me version numbers are easier
[21:08:36 CET] <SonicTheHedgehog> More than lack of features, I generally worry more about if it's the latest stable; if I need the latest stable, I generally either build from source or use a distro that keeps up, like Arch
[21:08:46 CET] <iranen> guys unless you dont need old portable devices dont use mp3 anymore. opus is best lossy audio codec atm
[21:09:09 CET] <aiena> iranen: I think the webm container has the audio stream in opus itself
[21:09:17 CET] <SonicTheHedgehog> ^ This is true
[21:09:19 CET] <aiena> I dont know if android 5 can play opus natively
[21:09:21 CET] <kepstin> pretty much the only reason to replace the ffmpeg from a recent (non-lts) ubuntu is if there's a specific codec you want that's no included, like fdk-aac or something.
[21:09:25 CET] <SonicTheHedgehog> I did run some ffprobe's on some webm downloads lately
[21:09:31 CET] <SonicTheHedgehog> And I do recall opus showing up
[21:10:14 CET] <aiena> if android 4 can play it I wonder if ffmpeg can just extract stream 0 in opus format directly
[21:10:21 CET] <aiena> *err android
[21:10:23 CET] <aiena> 5
[21:10:32 CET] <kepstin> android 5? pretty sure there's no opus decoder there, but third party apps often include opus
[21:10:58 CET] <aiena> so mp3 for now I dont want to add yet another app on this phone
[21:10:59 CET] <kepstin> of course, if you only have android 5, you should probably grab a newer device so you can get security updates :(
[21:11:02 CET] <iranen> Im using vlc and/or foobar to play opus on my phone
[21:11:20 CET] <aiena> vlc is perfect just dont have space to handle yet another app
[21:11:27 CET] <SonicTheHedgehog> I wish VLC had gapless playback
[21:11:39 CET] <aiena> ideally wav is the best but it hogs spaaaaaaaaace
[21:11:40 CET] <SonicTheHedgehog> A dev in the VideoLan channel told me it doesn't and likely won't in the near future ._.
[21:11:46 CET] <SonicTheHedgehog> aiena: what.
[21:11:52 CET] <SonicTheHedgehog> aiena: Why not FLAC
[21:12:00 CET] <aiena> actuall FLAC is better
[21:12:00 CET] <SonicTheHedgehog> That takes up less space, and it's lossless compression.
[21:12:06 CET] <kepstin> if you really want lossless, you should at least compress it...
[21:12:13 CET] <SonicTheHedgehog> ^
[21:12:48 CET] <aiena> but WAV is guaranteed to play on almost anything it is soooo ancient.
[21:12:49 CET] <kepstin> but there's no way you could tell the difference given the dac and headphones on an average phone.
[21:13:05 CET] <aiena> ok
[21:13:25 CET] <iranen> even youtube uses vp9 and opus codec (webm)
[21:13:44 CET] <SonicTheHedgehog> Yeah, a usual 320kbps MP3 and a FLAC are virtually indistinguishable on most equipment for most people
[21:13:48 CET] <aiena> yeah webm was made and vpx were made by google
[21:14:04 CET] <aiena> SonicTheHedgehog: I prefer variable encoding
[21:14:14 CET] <aiena> i think if you keep a quality of 2
[21:14:27 CET] <aiena> as per lame it is hard to make out any difference
[21:14:44 CET] <aiena> cant remember it was some 128 - something kbps
[21:14:53 CET] <aiena> for variable bit rate encoding
[21:14:55 CET] <kepstin> yeah, lame -V2 is gonna be pretty great for most folks, I think it averages around 160-192kbps?
[21:15:20 CET] <aiena> not sure how you tell ffmpeg which v you want to use
[21:15:23 CET] <kepstin> could be off there, it's been a while since i've encoded mp3 :)
[21:15:51 CET] <aiena> suppose I want to convert opus to mp3 with VBR and quality 2
[21:15:52 CET] <kepstin> in ffmpeg, use the "-c:a libmp3lame -q 2"
[21:16:04 CET] <SonicTheHedgehog> iirc on maybe older LAME versions, alt preset standard was one of those settings
[21:16:10 CET] <SonicTheHedgehog> That gave a good output
[21:16:15 CET] <SonicTheHedgehog> I can't recall what it is on newer lame versions
[21:16:24 CET] <aiena> kepstin: that is before -i right
[21:16:39 CET] <kepstin> SonicTheHedgehog: yeah, on modern lame versions the alt preset system became the standard system, and now you just specify a quality level with -V
[21:16:47 CET] <SonicTheHedgehog> right, I see, kepstin
[21:17:01 CET] <kepstin> aiena: those are output options, so they go before the output file (after all input files)
[21:17:22 CET] <SonicTheHedgehog> aiena: isn't opus already lossy compressed? I'm not a fan of making an already lossy compressed file go through another lossy compression
[21:18:05 CET] <aiena> SonicTheHedgehog: yes it is but if I do not have an opus decoder and do not want to install an opus decoder I do not have any other choice
[21:18:41 CET] <SonicTheHedgehog> The most I can tell you then is to maybe convert to FLAC, that'll retain the quality (even if lossy) but won't degrade it further
[21:19:03 CET] <aiena> Ok could test if flac plays
[21:19:21 CET] <kepstin> of course, if this is for a phone, you're not gonna fit very much music in flac without a huge sd card :)
[21:19:27 CET] <SonicTheHedgehog> Yeah that too
[21:19:32 CET] <SonicTheHedgehog> And unfortunately FLAC's tend to be big
[21:19:42 CET] <aiena> SonicTheHedgehog: so the output file sie will be larger than the input in flac?
[21:19:56 CET] <aiena> size
[21:20:06 CET] <aiena> for opus -> flac?
[21:20:16 CET] <SonicTheHedgehog> Well, there is a setting to make it compress it the most, which will take more time for your computer to crunch and possibly slightly more CPU to decode later when being played back
[21:20:29 CET] <kepstin> aiena: depends on the input, but typically flac will be much larger yes
[21:20:30 CET] <SonicTheHedgehog> But that setting will make the smallest size possible with FLAC with that specific encoder
[21:20:48 CET] <SonicTheHedgehog> But yes, usually a FLAC tends to be bigger than a high-quality lossy compressed equivalent
[21:21:17 CET] <aiena> wow input is 122 mp and with default options the FLAC is 1GB what the heck
[21:21:22 CET] <aiena> 10 times larger
[21:21:24 CET] <SonicTheHedgehog> wait what.
[21:21:25 CET] <aiena> *122 mb
[21:21:44 CET] <SonicTheHedgehog> Don't use default, try to use best compression
[21:21:52 CET] <aiena> how do I do that?
[21:22:00 CET] <SonicTheHedgehog> You were ffmpeg for it, right?
[21:22:03 CET] <ccooc> witch server is good and working easily with ffmpeg to stream rtsp and mjpeg?
[21:22:10 CET] <kepstin> 10 times larger is typical, best compression provides only marginal improvements
[21:22:17 CET] <iranen> average song in FLAC is like 50mb and opus 5mb
[21:23:16 CET] <aiena> and how do I do that?
[21:24:23 CET] <ccooc> why di i get on this command ffmpeg -i 1.264 -i 2.264 -filter_complex "[0][1] overlay=0:0" http://x.x.x.x    broken pipe??????
[21:24:29 CET] <SonicTheHedgehog> aiena: ffmpeg -i input -c:a flac -compression_level 12 output.flac
[21:24:53 CET] <SonicTheHedgehog> I don't know how much it'll reduce it by, but it'll be something
[21:24:58 CET] <aiena> SonicTheHedgehog: also do you know how to extract only the opus stream from webm but not important I dont see any benefit VLC can easily play the opus from the webm container.
[21:26:06 CET] <SonicTheHedgehog> aiena: For that, ffmpeg -i videofile.webm -vn -acodec copy audiofile.opus    may do it
[21:27:01 CET] <aiena> good to see blue here too.
[21:27:10 CET] <aiena> ANyway thanks sonic good night.
[21:27:18 CET] <SonicTheHedgehog> yw
[21:27:33 CET] <aiena> and thank you kepstin too
[21:27:46 CET] <ccooc> SonicTheHedgehog: can you help me please?
[21:28:16 CET] <kepstin> ccooc: broken pipe there just means that the remote side of the network connection hung up on you.
[21:28:44 CET] <aiena> ccooc: https://en.wikipedia.org/wiki/List_of_streaming_media_systems may help you in your research
[21:28:48 CET] <ccooc> kepstin: is not correct becuase if i do   :
[21:28:49 CET] <ccooc> fmpeg -i 1.264 -i 2.264 -filter_complex "[0][1] overlay=0:0"  -f h264 pipe:1 | ffmpeg -i pipe:0 http://x.x.x.x
[21:28:56 CET] <ccooc> all is right , the server is ok
[21:29:03 CET] <ccooc> but i don't want to use with 2 pipes
[21:32:51 CET] <furq> ccooc: what's wrong with -f h264 http://...
[21:43:06 CET] <kepstin> furq: ccooc complains about 'broken pipe' errors with that, so I guess the server doesn't like something about ffmpeg's output without the extra pipe and re-encode - too bursty, maybe?
[21:43:18 CET] <kepstin> hard to tell without some server error logs :)
[21:43:32 CET] <furq> 20:24:22 ( ccooc) why di i get on this command ffmpeg -i 1.264 -i 2.264 -filter_complex "[0][1] overlay=0:0" http://x.x.x.x    broken pipe??????
[21:43:41 CET] <furq> there's no -f h264 there so i guess it's defaulting to the wrong codec
[21:43:49 CET] <furq> although idk what it would be defaulting to
[21:44:07 CET] <kepstin> hmm, there's a solution to that question
[21:51:23 CET] <ccooc> ok i will look on this
[21:52:27 CET] <ccooc> but friend, i saw that ffserver is drop from project and not Maintained , so how do you stream rtsp with ffmpeg ?
[21:53:08 CET] <ccooc> i know that there are lot of good server , but i not sure if they working well with ffmpeg
[22:55:02 CET] <durandal_1707> ccooc: via nginx
[23:10:05 CET] <TheRock> just tried to compile with vs 2017 (before VS 2015, worked). I'm getting this error:
[23:10:24 CET] <TheRock> cmdutils.o : error LNK2019: unreferenced Symbol "_SetDllDirectory" in function "_init_dynload".
[23:10:29 CET] <TheRock> Is it to anyone known?
[00:00:00 CET] --- Fri Nov 24 2017