[Ffmpeg-devel-irc] ffmpeg.log.20160623

[burek]

[00:31:28 CEST] <rmbeer> hello again
[00:33:11 CEST] <rmbeer> ignore my text, this is only my error.
[00:33:39 CEST] <rmbeer> thank and bye
[00:48:15 CEST] <Psi-Jack> Is there a way to use ffmpeg to convert audio from one format to another, displaying the progress of the file, but NOT the metadata that fills the screen every time?
[00:48:44 CEST] <Psi-Jack> I know about -loglevel quiet, but that suppresses the progress too.
[01:10:42 CEST] <c_14> Psi-Jack: -loglevel quiet -stats
[01:26:05 CEST] <Psi-Jack> Hmmm, sheash, how did I not see that one? ;)
[01:26:09 CEST] <Psi-Jack> Thanks, c_14.
[03:22:26 CEST] <ochoa> hey, does ffmpeg have the option to use the same filename as the imput file?
[03:22:45 CEST] <c_14> no
[03:22:53 CEST] <ochoa> really?
[03:23:05 CEST] <kepstin> ochoa: as output? no.
[03:23:31 CEST] <kepstin> it doesn't do any of the 'temporary output file then rename overtop' thing, so it would end up deleting/overwriting its input as it goes.
[03:23:42 CEST] <ochoa> so i can't ffmpeg -i file1.mkv $FILENAME.mp4?
[03:23:43 CEST] <kepstin> iirc
[03:24:19 CEST] <kepstin> ... that's a different filename :) If the filename's in a variable, you can shell script it...
[03:24:33 CEST] <ochoa> that's what i maen
[03:25:02 CEST] <kepstin> but nothing builtin to ffmpeg, you'd use your shell to do it.
[03:25:57 CEST] <ochoa> yeah, now i don't know how to grab a filename from shell
[03:27:11 CEST] <furq> ffmpeg -i "$1" "${1%.*}".mp4
[03:28:09 CEST] <ochoa> ah, the wildcards!
[03:29:07 CEST] <ochoa> i think that % is to ignore everything after the point? furq
[03:29:34 CEST] <furq> https://www.gnu.org/software/bash/manual/html_node/Shell-Parameter-Expansion.html
[03:29:49 CEST] <furq> most if not all of that applies to posix shell
[03:38:37 CEST] <ochoa> thanks furq i'm reading it right now
[07:45:26 CEST] <Bray90820> What does this error mean
[07:45:26 CEST] <Bray90820> Past duration 0.871452 too large
[08:51:10 CEST] <jookiyaya> b6e7" timestamp="1398231221"><legacyquote>[4/22/2014 10:33:41 PM] Mark: </legacyquote>Is mike high at night and getting his chat game on while the wife and kids are sleeping?<legacyquote>
[08:51:11 CEST] <jookiyaya> <<< </legacyquote></quote>
[09:02:17 CEST] <sceadwian> Can anyone tell me how to compile ffmpeg with .aax support?
[10:39:16 CEST] <alsecz> hi! how can i mix rawvideo & dshow formats?
[10:39:47 CEST] <alsecz> i need video from sequence of images & audio from dshow device on windows
[10:57:40 CEST] <m3k> Hello
[12:06:11 CEST] <touki> Hello, could someone indicates me a ressource for configuring FFMPEG so it captures specific window under Linux?
[12:07:41 CEST] <spirou> oh, I didn't know it was possible to do screencapture with ffmpeg
[12:07:57 CEST] <touki> it is!
[12:08:05 CEST] <spirou> I suppose this is the short short guide: https://trac.ffmpeg.org/wiki/Capture/Desktop
[13:51:53 CEST] <touki> Has anyone screened a single window on Linux with ffmpeg?
[14:19:34 CEST] <nick0> Hi, I'm trying to decode an audio file, filter it, resample the filtered output and output the final PCM. Problem is, the output has the entire song, properly resampled, but it also has a lot of extra frames (something like 1:40 minutes, which sounds like the song sped up and played backwards) but I can't figure out why.. Here's my code: https://gi
[14:19:34 CEST] <nick0> st.github.com/anonymous/dd9fa94c7fdf96df47690265c781279a
[15:27:39 CEST] <nifwji2__> I realised that irc is ruining my life.
[15:27:51 CEST] <nifwji2__> and constantly complaining about irc on irc just gets me banned.
[15:28:03 CEST] <nifwji2__> and that makes me hate irc more.
[15:28:05 CEST] <nifwji2__> so screw you guys
[15:28:54 CEST] <__jack__> huh !
[15:38:44 CEST] <Danimal> I haz question: is there a way to turn off interpolation when converting frame rate? if I do this `ffmpeg -i vsync_1080p_60FPS_4s.mp4 -r 30 output.mp4` the output seems to 'vibrate' like the frames are coming out of order in the output or something. if it simply took every other frame and played them half as fast I wouldn't think it'd do that?
[15:48:35 CEST] <spirou> when a tv channel converts a anime from the original framerate to ours, wouldn't it be possible to have an algorithm that don't make panning look horrible?
[15:53:49 CEST] <spirou> like regenerate the moving background, from the existing frames and then move it in the same speed during the panning instead of doing this strange mixing of some frames giving the jerky motion
[16:00:07 CEST] <klaxa> spirou: if the backgrounds don't move for a frame, whoever encoded the video was incompetent
[16:00:26 CEST] <spirou> yeah I guess they are :D
[16:00:41 CEST] <klaxa> otherwise, you want some motion interpolation, there is for example SVP afaik there is also a gsoc project regarding a motion interpolation filter
[16:01:28 CEST] <spirou> when I stepped through the film file here it looks like now and then there is a frame that is a mix of two or something, creating the jerky panning
[16:02:34 CEST] <spirou> since it is anime I suppose it is 30fps that they made into 25fps
[16:02:40 CEST] <klaxa> yeah i don't think that's the animation studio or distributor's mistake though
[16:02:53 CEST] <klaxa> it's 24 frames that they make into 60 frames
[16:02:56 CEST] <klaxa> to make it 24 again
[16:03:02 CEST] <klaxa> (actually 23.976)
[16:03:18 CEST] <spirou> ah yeah it could 24fps that is made into 25 fps perhaps?
[16:03:40 CEST] <spirou> since tv is 25fps (and this film is downloaded from the tv channels pages)
[16:03:45 CEST] <klaxa> maybe, but maybe there is anime that is animated in 25 fps
[16:04:04 CEST] <klaxa> tv is 60 interlaced frames though
[16:04:08 CEST] <klaxa> usually
[16:04:29 CEST] <spirou> in usa and japan I suppose they use 60fps. we in europe do 25fps
[16:05:06 CEST] <ocZio> is it possible to limit the file size by specifying a bitrate ?
[16:05:20 CEST] <ocZio> I have 4 images and want the output to be less than 280kb
[16:05:30 CEST] <ocZio> (making video out of 4 frames)
[16:05:34 CEST] <spirou> a modern tv can do any framrate of course, but the tv-channel can't send it other than 25fps or really old tv's would go bezerk :D
[16:05:38 CEST] <klaxa> well it's still not really 25 fps, but 50 interlaced frames
[16:06:13 CEST] <klaxa> ocZio: if it's that short you can try playing around with -b:v
[16:06:41 CEST] <klaxa> like: ffmpeg -i *.jpg -c:v libx264 -b:v 60k output.mkv or somethign
[16:06:56 CEST] <ocZio> what is -b:v standing for ? bitrate video ?
[16:07:03 CEST] <ocZio> just want to read the docs about the options
[16:07:05 CEST] <klaxa> yes
[16:07:28 CEST] <spirou> ah yeah true. tv standard is 50Hz interlazed. but the downloaded film of theirs is "Frame rate : 25.000 fps"
[16:08:12 CEST] <klaxa> it's probably just deinterlaced then
[16:08:17 CEST] <klaxa> and they still messed it up :D
[16:08:41 CEST] <spirou> ah
[16:09:49 CEST] <spirou> I guess it is too late to magically fix it up to be non-jerky now though. I probably should download the original film from torrents somewhere and mix it with our sound
[16:12:53 CEST] <spirou> ocZio: wouldn't a 4 images long film be quite short? :D
[16:13:06 CEST] <ocZio> spirou its 2 seconds
[16:13:11 CEST] <ocZio> 2 frames per second
[16:13:16 CEST] <ocZio> its like making an animated gif
[16:13:21 CEST] <spirou> ah
[16:13:32 CEST] <ocZio> I just want to make sure the size is around 250kb
[16:13:42 CEST] <ocZio> currently it is 110
[16:14:00 CEST] <spirou> is animated gifs still around btw? I thought they sort of died with the patents stuff....
[16:15:13 CEST] <spirou> but then I see ".gifv" files now that is horrible gzipped webpages with javscript and css and crap that then links to a .mp4 file on the web
[16:16:09 CEST] <klaxa> don't get me started on .gifv
[16:16:14 CEST] <klaxa> .gifv is not a standard
[16:16:16 CEST] <klaxa> it is not a format
[16:16:20 CEST] <klaxa> it's imgur javascript bullshit
[16:16:29 CEST] <ocZio> :)
[16:16:32 CEST] <ocZio> I am just doing mp4
[16:16:32 CEST] <klaxa> it's just an alias for "this file is available as webm, mp4 and gif"
[16:16:46 CEST] <klaxa> and you get whatever works worst in your browser
[16:17:19 CEST] <klaxa> they could have just jumped on the right train like everyone else and replace gifs with webms
[16:17:30 CEST] <klaxa> because gif is being abused for videos
[16:22:58 CEST] <spirou> ah
[16:27:01 CEST] <spirou> (yeah if they of some strange reason would want to have animated gif pictures then they could just add an <img> inside the <video></video> and the webbrowsers would find it themself, no need for javascrupts for that)
[16:29:56 CEST] <P4Titan> Hello all. I am looking at the transcode_aac.c example of ffmpeg, but when I try to compile my own code, AVStream complains that it does not have a codecpar field
[16:30:59 CEST] <P4Titan> Could that be in the configure options
[16:31:06 CEST] <P4Titan> that I didn't enable something
[17:02:31 CEST] <ArboreSign> https://www.youtube.com/watch?v=uOjH41Sjb7M Julia Probst ist eine der sichtbarsten Aktivistinnen für Gehörlose. Im Gespräch mit Philip Banse erklärt @einaugenschmaus, wie sie sich eine Gesellschaft vorstellt, in der sich Menschen, die schlecht oder gar nicht hören können, zuhause sind.
[17:05:16 CEST] <mynick> i have a question
[17:05:47 CEST] <mynick> can i add metadata without processing the file?
[17:06:11 CEST] <furq> you can add metadata without reencoding but not without creating a new file
[17:06:15 CEST] <furq> with ffmpeg, anyway
[17:06:25 CEST] <mynick> that, without reencoding
[17:06:30 CEST] <furq> -c copy
[17:06:34 CEST] <mynick> to make it quickly
[17:07:00 CEST] <mynick> ffmpeg -i song.mp3 -metadata artist='asdf' -c copy
[17:07:03 CEST] <mynick> ?
[17:07:22 CEST] <furq> sure
[17:07:39 CEST] <furq> although you're probably better off using an actual id3 editor
[17:07:53 CEST] <mynick> i have tried to make it with id3
[17:08:02 CEST] <mynick> with eyeD3 python
[17:08:28 CEST] <mynick> but i don't know why in ubuntu doesn't appear
[17:09:37 CEST] <mynick> ffmpeg -metadata param is not the same that opening an ID3 tag?
[17:09:47 CEST] <furq> ffmpeg can't edit files inplace
[17:10:11 CEST] <mynick> yes but i can set output to /tmp/ and then replace it
[17:10:15 CEST] <furq> sure
[17:10:28 CEST] <furq> that seems needlessly cumbersome though
[17:10:42 CEST] <mynick> what? my english level is poor :(
[17:11:11 CEST] <furq> that seems like more effort for no reason
[17:11:32 CEST] <mynick> ah
[17:11:54 CEST] <mynick> so, what whould you do?
[17:12:11 CEST] <mynick> i have tried with eyeD3 but it not displays the metadata in ubuntu.
[17:12:17 CEST] <mynick> can i send a screenshot?
[17:12:20 CEST] <mynick> how?
[17:12:34 CEST] <mynick> i'm very newbie
[17:13:04 CEST] <mynick> does not matter
[17:13:46 CEST] <mynick> the type of metadata that creates ffmpeg is the same that ID3?
[17:13:57 CEST] <furq> that's the only type of metadata that exists in mp3 files
[17:14:12 CEST] <furq> when you say "in ubuntu" do you mean in the file browser or something
[17:14:16 CEST] <furq> that could just be caching the old metadata
[17:14:30 CEST] <mynick> i say in file browser, yes.
[17:14:46 CEST] <mynick> because if i open the tag from python it appears the data
[17:15:06 CEST] <mynick> but i have tried to change the name of the file and the same happends.
[17:15:10 CEST] <furq> have you restarted nautilus
[17:15:17 CEST] <furq> or whichever file browser you're using
[17:15:23 CEST] <mynick> psudo pkill Xorg?
[17:15:32 CEST] <mynick> sudo*
[17:15:39 CEST] <furq> that seems excessive but i guess it would work
[17:15:57 CEST] <mynick> i use unity
[17:17:20 CEST] <mynick> pkill nautilus will work?
[17:18:14 CEST] <furq> probably
[17:24:13 CEST] <mynick> It's not working
[17:24:46 CEST] <mynick> can it be a nautilus bug?
[17:24:58 CEST] <mynick> because i can see the metadata from python
[17:28:53 CEST] <mynick> from stackoverflow:
[17:28:54 CEST] <mynick> >>> import eyeD3 >>> tag = eyeD3.Tag() >>> tag.link("/tmp/song.mp3") 1 >>> tag.header.setVersion(eyeD3.ID3_V2_3) >>> tag.setArtist('Fugazi') >>> tag.update() 1
[17:29:34 CEST] <mynick> that works, because then, i can read the metadata from python
[17:29:45 CEST] <mynick> but i can't from nautilus and i have restarted it.
[17:33:53 CEST] <mynick> somebody?
[17:37:22 CEST] <mynick> does anyone have a idea?
[17:45:06 CEST] <mynick> i'm alone
[18:38:34 CEST] <delicado> hi guys how do i generate the documentation for ffmpeg? i run doxygen in doc directory but it says "SYMBOL_CACHE_SIZE is obsolete" and many other errors that says some stuff are obsolete. But how do you guys build your documentation?
[19:19:04 CEST] <mynick> delicado, i think all the people here are idle
[19:19:11 CEST] <mynick> cheers from spain
[19:21:23 CEST] <JEEB> delicado2: there's a Doxyfile under doc/
[19:22:02 CEST] <JEEB> the official docs are built with doxygen 1.8.6
[19:22:18 CEST] <JEEB> so if you have a much newer version it might not work depending on if anyone has updated the Doxygen file
[19:22:22 CEST] <JEEB> *Doxyfile
[19:23:15 CEST] <JEEB> my first guess would be that no-one has touched it in years... http://git.videolan.org/?p=ffmpeg.git;a=history;f=doc/Doxyfile;h=0c1604e00769087e39a33a6766cc09a4a149530e;hb=HEAD
[19:23:37 CEST] <JEEB> which seems to be correct
[19:23:51 CEST] <JEEB> also there seems to be a doxy-wrapper.sh shell script
[19:24:41 CEST] <JEEB> if you don't necessarily need to compile it yourself, latest docs are available @ http://ffmpeg.org/doxygen/trunk/
[19:29:31 CEST] <DHE> I use doxygen 1.8.11 and running "doxygen docs/Doxyfile" just does it (using ffmpeg from Git)
[19:37:13 CEST] <mynick> heyhey i have solved my problem! the problem was that it wasn't a mp3 file, it was a webm file with .mp3 extension. Thanks a lot!
[19:45:19 CEST] <Ant___> hello all. potentially silly question: If I am stitching together a series of PNGs to make an progressive animation (eg: animating drawing a rectangle) - does each image/frame have to be complete (i.e: contain all previously drawn pixels) OR can each image be transparent, contain only the new pixels added in that frame, and have ffmpeg continuously add each frame as a cumulative overlay to previously stitched frames?
[19:49:45 CEST] <kepstin> Ant___: if you're just gonna use ffmpeg's builtin support for reading a series of images as video frames, each will need to be complete.
[19:52:18 CEST] <Ant___> thank you Kepstin - that saves me a bunch of time trying a failed experiment to save space & time :)
[19:54:56 CEST] <kepstin> if you want to save time and space; maybe skip the png encoding completely, and simply pipe raw rgb images directly into ffmpeg :)
[19:55:06 CEST] <kepstin> Depends on how you're making the animation, whether you can do that.
[19:57:56 CEST] <furq> you could overlay each frame but the only way i know how to do that is with one overlay filter invocation per frame
[19:58:00 CEST] <furq> so you probably don't want to do that
[19:58:21 CEST] <Ant___> interesting idea. Probably beyond my current skills & deadline.  I'm rendering GPS traces over time as a movie. I'm using python / PIL to make each image, converting the latlngs into screen pixel coordinates for each timeslice, then opening up each previous png, painting in the new pixels for the timeslice, and saving out as the next png frame in sequence
[19:59:22 CEST] <Ant___> its working well, but I'm working on time-saving procedures now - like not bothering to make frames from empty timeslices, etc
[20:02:03 CEST] <furq> i think apng supports overlaying onto previous frames actually
[20:03:10 CEST] <Ant___> however - next question.. I just had to render my first 4k Ultra resolution movie, which meant using -c:v libx265 and using vlan to view. Sometimes the movie plays perfectly, but most times, a few seconds in, it goes from the white lines on a black background to suddenly being grey with flashing squares moving around the screen.  Could this purely be a factor of machine power & viewing 4k Ultra on a mac laptop, or could this perhaps b
[20:05:42 CEST] <Ant___> furq: thank you for the apng tip - I will check that out
[21:14:19 CEST] <Mooniac> any Linux tools to repair broken mp4 files? I've seen such products for Win, but I haven't found anything yet for Linux
[21:16:36 CEST] <mynick> you can try reencoding with ffmpeg, or opening with vlc and broadcasting to a file.
[21:47:08 CEST] <CFS-MP3> I'm following this thread: https://ffmpeg.zeranoe.com/forum/viewtopic.php?t=2558
[21:47:34 CEST] <CFS-MP3> to accomplish exactly the same thing - from a TS that has DVB subtitles generate another one that has the subtitles burned-in
[21:47:51 CEST] <CFS-MP3> but I get this message: sub2video: rectangle overflowing
[21:48:11 CEST] <CFS-MP3> Admitely I don't really know what I'm doing (yet) just following that thread more or less blindly
[21:48:54 CEST] <CFS-MP3> How can I force the overlay filter to reduce the size of the rectangle (if that's the solution), or just trim as needed (as least as a test)
[21:49:16 CEST] <CFS-MP3> or ideally, just do what it should which is use the same dimensions it would on a TV? :-)
[21:49:39 CEST] <CFS-MP3> this is the command I'm using: ffmpeg -i BBC1_TS.ts -filter_complex "[0:v][0:s]overlay[v]" -map "[v]" -vcodec libx264 -preset ultrafast -x264opts keyint=100:min-keyint=50 -threads -0 -f avi copy.avi
[21:50:37 CEST] <BtbN> Doesn't the overlay filter overlay two videos?
[21:50:58 CEST] <llogan> it can overlay image based subtitles
[21:51:58 CEST] <JEEB> (because it creates a video track out of them internally, check ffmpeg.c)
[21:52:29 CEST] <furq> i guess you could scale down the subs before overlaying them then
[21:53:38 CEST] <furq> -filter_complex "[0:s] scale:720:-2 [subs]; [0:v][subs] overlay [v]"
[21:54:50 CEST] <CFS-MP3> furq: No such filter: 'scale:720:-2'
[21:55:00 CEST] <furq> scale=720:-2
[21:55:45 CEST] <furq> whether that will produce acceptable output i don't know, but it should at least work
[21:57:52 CEST] <CFS-MP3> It does run but I get the same error.
[21:58:58 CEST] <CFS-MP3> Actually I have change the scale to 704:-2 (because the dimensions are 704x578) the same thing happens
[22:00:38 CEST] <CFS-MP3> I mean "If I change"
[22:05:35 CEST] <CFS-MP3> This seems reasonable
[22:05:36 CEST] <CFS-MP3> Input #0, mpegts, from 'BBC1_TS.ts':
[22:05:36 CEST] <CFS-MP3>   Duration: 00:02:24.43, start: 1.400000, bitrate: 1856 kb/s
[22:05:37 CEST] <CFS-MP3>   Program 1
[22:05:37 CEST] <CFS-MP3>     Metadata:
[22:05:37 CEST] <CFS-MP3>       service_name    : Service01
[22:05:37 CEST] <CFS-MP3>       service_provider: FFmpeg
[22:05:37 CEST] <CFS-MP3>     Stream #0:0[0x100]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv), 704x576 [SAR 16:11 DAR 16:9], max. 15000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
[22:05:38 CEST] <CFS-MP3>     Stream #0:1[0x101](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 256 kb/s
[22:05:38 CEST] <CFS-MP3>     Stream #0:2[0x102](eng): Subtitle: dvb_subtitle ([6][0][0][0] / 0x0006)
[22:05:39 CEST] <CFS-MP3> [mpegts @ 0x151fe80] sub2video: using 704x576 canvas
[22:32:53 CEST] <MuzerLG> I'm abroad and want to listen to the EU referendum results coming in in an hour and a half but it turns out the lowest bandwidth audio stream is far too high bandwidth for the connection available to me. I have a computer at home, is it possible (using ffmpeg or something else) to re-encode this stream into something super low bitrate that I can easily stream to my phone? Audio quality is unimportant as long as I can discern speech.
[22:33:03 CEST] <MuzerLG> http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio4fm_mf_p is the url
[22:38:37 CEST] <furq> MuzerLG: probably something like ffmpeg -i $url -c:a libopus -b:a 16k -f mpegts tcp://0.0.0.0:1234?listen
[22:39:06 CEST] <furq> that'll only serve one client and it'll drop when the client disconnects, so you might need to run it in a loop
[22:39:39 CEST] <furq> alternatively you could use something like hls or dash but that would add a bit of overhead
[22:40:09 CEST] <MuzerLG> Cheers, I'll give it a go. Normally I'd try to work it out myself but it took me by surprise, this, and it's hard to figure things out on a phone!
[22:40:22 CEST] <MuzerLG> (i assumed the bbc's low bandwidth streams were still there)
[22:41:58 CEST] <CFS-MP3> OK, my issue was caused by a specific ffmpeg version
[22:42:54 CEST] <furq> MuzerLG: there's a 64kbps aac stream but i have no idea if it's available overseas
[23:08:38 CEST] <drewbert> Hi all, I have an h.264 live video stream being made available on the local network. I have a system that connects to it and publishes to a wowza streaming engine when the user requests using the command `ffmpeg -f h264 -i tcp://localhost:5601 -an -vcodec copy -f flv rtmp://user:password@my.server.com/applicationName/streamName`
[23:09:50 CEST] <drewbert> The ffmpeg process is running on a system that connects through a mobile network.  When the network strength is low, I want to reduce the bandwidth ffmpeg uses without reencoding the video.
[23:11:05 CEST] <drewbert> I tried adding -discard nokey to the input, so that only keyframes would be sent, but I guess the copy vcodec doesn't support this?  Is there a way to achieve what I want?
[23:17:11 CEST] <royendgel> Hi there : is it possible to have the current time in playlist ? its not important which playlist, I want to be able to see the time of each segment http://pastebin.com/N7bRpU4i
[23:17:56 CEST] <royendgel> With current time I mean the real time (good if it is utc) date + time
[23:59:02 CEST] <MoSal> Hello. loglevel seems to be not respected when creating TS files with only an Opus stream (no video).
[23:59:10 CEST] <MoSal> Can anyone confirm?
[00:00:00 CEST] --- Fri Jun 24 2016