[Ffmpeg-devel-irc] ffmpeg.log.20140622

[burek]

[00:53] <RadSurfer> there is a way to silence an audio stream, (make volume 0, or audio track null?) how would I do that please?
[00:54] <c_14> If you want to silence the entire audio stream, couldn't you just remove it? ie remux without it? If you want an audio stream, but want it to be silent, use the aevalsrc filter.
[00:55] <RadSurfer> Some players require an audio stream be there; so I just want to silence it
[00:56] <c_14> use -an and -af aevalsrc=0
[00:57] <c_14> hmm, wait.
[00:57] <RadSurfer> ?
[00:57] <c_14> looks like -an kills the implicit mapping in the -af filter
[00:57] <RadSurfer> meanwhile, I'll edit the video that needs silencing...
[00:59] <c_14> Right, -filter_complex 'aevalsrc=0[a]' -map '[a]' should work.
[00:59] <c_14> You'll have to explicitly map the video/subtitle streams you want though.
[00:59] <c_14> use something like -map 0:v if you want all the video streams
[01:00] <RadSurfer> I want video UNtouched; just silence it
[01:00] <c_14> add -c:v copy
[01:00] <c_14> That'll copy the video stream.
[01:06] <c_14> Oh, you'll want to add -shortest.
[01:06] <c_14> Or else it'll just keep going.
[01:06] <RadSurfer> what does that mean?
[01:07] <c_14> aevasrc creates a source file with null audio of infinite length. So if you try muxing that with your video it'll go on forever. -shortest says that it should only create a source with the length of the shortest input stream.
[01:09] <RadSurfer> you syntax is rejected by ffmpeg
[01:11] <RadSurfer> http://pastie.org/9312691
[01:13] <c_14> Pretty sure you can't have a -vf and a filter_complex at the same time. use -filter complex 'aevalsrc=0[a];[0:v:0]yadif[v]' -map '[a]' -map '[v]' -c:v libx264 <file>.mkv
[01:13] <RadSurfer> could have included error message...
[01:18] <RadSurfer> what it is you are trying to do: ffmpeg REJECTS syntax
[01:19] <RadSurfer> I'll try to paste whole thing this time, w/error
[01:22] <RadSurfer> http://pastie.org/9312715
[01:22] <RadSurfer> Meanwhile, I'm stuck.
[01:22] <RadSurfer> I can't post this with audio it has in it. :-)
[01:23] <c_14> Ok, in that specific case I accidentally wrote -filter complex instead of -filter_complex
[01:24] <RadSurfer> I didn't catch it: still learning syntax myself
[01:24] <c_14> Which is why it thinks that everything in the quotes is an output file.
[01:25] <RadSurfer> your syntax FAILS: 3 times now
[01:26] <RadSurfer> http://pastie.org/9312722
[01:28] <c_14> The command I gave you works just fine on my machine.
[01:29] <c_14> `ffmpeg -i in.mkv -aspect "16:9" -filter_complex 'aevalsrc=0[a];[0:v:0]yadif[v]' -map '[a]' -map '[v]' -c:v libx264 -shortest out.mkv'
[01:31] <RadSurfer> why do I get errors, 3
[01:31] <RadSurfer> one last time, I'll just have to think of something else, or use avidemux later
[01:32] <RadSurfer> SAME ERROR
[01:32] <RadSurfer> [a] invalid
[01:33] <RadSurfer> try that with an MP4  still work for you?
[01:35] <c_14> yep, switched to mp4 as input and that worked, also worked with mp4 as both input and output
[01:35] <RadSurfer> then why should it FAIL here
[01:35] <c_14> Are you using a recent version of ffmpeg?
[01:35] <RadSurfer> it was just downloaded 3 days ago!
[01:37] <c_14> You're using the windows terminal, right?
[01:37] <c_14> Are [] considered special characters?
[01:37] <RadSurfer> Win8.1
[01:37] <RadSurfer> cmd.exe
[01:37] <c_14> if not you could try removing the single quotes around [a] and [v]
[01:37] <c_14> and maybe around the aevalsrc part
[01:38] <c_14> Those quotes are only really there for shell level escaping, it might be that windows is mucking about with that
[01:38] <c_14> start by removing the single quotes around [a] and see if the error message changes
[01:38] <RadSurfer> anyone here willing to try that syntax in a winblows box?
[01:41] <debianuser> Hello. Trying to join multiple audio (wav, ogg) files into a single ogg. Ended up with: ls -1 dir/* | sed 's/^/file /' | ffmpeg -f concat -re -i - -acodec vorbis -ac 2 -ar 48000 -strict experimental 0.ogg  But I get a lot of "invalid dropping" messages (http://pastebin.com/Bmqp0zDp). Do I miss some important params?
[01:42] <RadSurfer> crying out loud...
[01:42] <RadSurfer> Syntax has been fixed. :-)
[01:43] <RadSurfer> on a winblows box: it only wants " not '  delimiters
[01:43] <c_14> I'll remember that for future reference.
[01:44] <RadSurfer> Okie
[01:45] <c_14> debianuser: Does the output audio sound ok/is as long as it should be?
[01:45] <RadSurfer> Just hope it works now
[01:51] <debianuser> c_14: It's shorter. On a small subset of 18 files the resulting 0.ogg is 00:01:04.06, while sum of initial files length is 71.46 (ls -1 dir/* | xargs -L 1 ffprobe 2>&1 | tr ':,' '  ' | awk '/Duration/{a+=$4}END{print a}')
[01:53] <c_14> question, are you sure you need -re?
[01:53] <debianuser> No, I don't.
[01:53] <c_14> But that shouldn't be the problem.
[01:53] <c_14> hmm
[01:54] <debianuser> (I'm almost sure that I don't, but I tried to add it recently, just to check whether it changes anything, it does not)
[01:55] <debianuser> I can share those 18 wav files (5MBytes) if it can help.
[01:56] <c_14> Sure, I can test it on my system. See what happens.
[01:58] <c_14> Those errors tend to hint that there is something wrong with the source files. At least with the timestamps.
[01:59] <debianuser> http://ge.tt/1Eg60nl1
[02:00] <RadSurfer> this encoding has been going on a long time!
[02:01] <RadSurfer> Appears that your syntax is really wrong after all.
[02:01] <RadSurfer> this video is <6min long
[02:01] <RadSurfer> it went WAY PAST that
[02:01] <c_14> did you add -shortest?
[02:02] <RadSurfer> whatever you had in there. I used your example
[02:02] <RadSurfer> lets see if this even plays
[02:02] <RadSurfer> its a mess
[02:02] <RadSurfer> time index is all screwed up!
[02:02] <RadSurfer> but at least it is silent
[02:03] <RadSurfer> I just want SILENCE for crying out loud... lol
[02:03] <RadSurfer> @least input file is undamaged
[02:14] <debianuser> c_14: Maybe I'm using a wrong command and people usually join files differently? It's hard to get something wrong with timestamps in wav-files...
[02:14] <c_14> Nah, command looks right.
[02:17] <c_14> Ok, so what I just did was resample all the input files to 48kHz with a for loop and then concatted that. It still shows the DTS/PTS warnings but the output file is 01:11:49 long which corresponds to the correct length.
[02:19] <c_14> Also, IIRC libvorbis is recommended over vorbis as an encoder.
[02:19] <c_14> you won't need -strict experimental anyway
[02:19] <RadSurfer> meanwhile: Should any one else have an idea how to SILENCE an MP4 audio track PLEASE let me know /msg welcome.
[02:24] <RadSurfer> I wonder if avidemux can even mute audio tracks?
[02:24] <RadSurfer> there's no obvious filter
[02:25] <debianuser> RadSurfer: what about something like -af volume=0 ?
[02:26] <RadSurfer> Can't hurt at this point
[02:26] <RadSurfer> :-)
[02:30] <RadSurfer> Here goes...
[02:31] <RadSurfer> I'd say perfect! Just what I wanted; and straight-forward too. embarassingly so.
[02:34] <RadSurfer> solved one problem, discovered another. But that works.
[02:50] <debianuser> c_14: I just had this ffmpeg built without libvorbis. I also tried to encode into .wav, but still got those "invalid dropping" messages. So, is it a bug? Or maybe "concat" hates different codecs/rates?
[02:58] <OnkelTem> Hi
[02:59] <OnkelTem> I'm trying to use this guide https://wiki.ubuntu.com/ScreenCasts/ffmpeg to make a screencast, but it doesn't work
[02:59] <OnkelTem> Unknown input format: 'x11grab'
[02:59] <OnkelTem> This error
[02:59] <OnkelTem> My command line is: ffmpeg -f x11grab -s 1280x760 -r 25 -i :0.0+3840+0 -f alsa -ac 1 -i pulse out.webm
[03:04] <OnkelTem> How comes that x11grab became "unknown"?
[03:04] <OnkelTem> ffmpeg version 2.1.3-   http://johnvansickle.com/ffmpeg/    Copyright (c) 2000-2013 the FFmpeg developers
[03:04] <OnkelTem> built on Jan 22 2014 19:50:14 with gcc 4.8 (Debian 4.8.2-14)
[03:15] <klaxa> static builds don't support x11grab
[03:17] <OnkelTem> wow
[03:17] <OnkelTem> good news
[03:17] <OnkelTem> I'm pretty sure, for a reason :)
[05:15] <debianuser> Hello. I'm trying to join many (hundreds of) audio files into one, and get some errors whatever I try even for a small subset.
[05:16] <debianuser> Command `echo -e "file 1.wav\nfile 2.ogg\nfile 3.wav" | ffmpeg -f concat -i - -ac 2 -ar 48000 0.wav` gives incorrect output file and lots of errors: pastebin.com/KQQSCGM4
[05:16] <debianuser> Command `ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex 'concat=n=3:v=0:a=1[a]' -map '[a]' -ac 2 -ar 48000 0.wav` gives a short file (10.92+3.43+12.89!=23.83) and many errors: pastebin.com/nEYL9RLa
[05:16] <debianuser> Those 3 test files: ge.tt/1Eg60nl1/v/1 What am I doing wrong? How to correctly join files without errors?
[11:25] <_Crash_Laptop> Hi, just installed ffmpeg via "sudo apt-get installed ffmpeg" in Ubuntu 12.04 and when i try and run "ffmpeg" through terminal it cannot locate it "The program 'ffmpeg' is currently not installed.  To run 'ffmpeg' please ask your administrator to install the package 'libav-tools'". I had removed the binary ffmpeg when i compiled it but i've ran the install three times and even removed it fully before re-installing with
[11:25] <_Crash_Laptop>  "remove --purge" - any ideas?
[11:27] <sacarasc> Did you install the package it asked you to?
[11:27] <_Crash_Laptop> yes
[11:27] <sacarasc> (Also, Ubuntu doesn't have ffmpeg any more, it has libav which is a fork.)
[11:28] <_Crash_Laptop> i know, but i need ffmpeg for serviio
[11:28] <sacarasc> Grab one of them instead?
[11:29] <sacarasc> Wait, did you compile yourself? Did you follow the Ubuntu compile guide?
[11:29] <sacarasc> https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide
[11:29] <_Crash_Laptop> i compiled it before, but i removed it and tried via apt-get which doesn't seem to be placing the binary where it is supposed to
[11:30] <sacarasc> In that case, ask in #ubuntu or #libav, because you're not installing ffmpeg, anyway.
[11:31] <_Crash_Laptop> right
[11:33] <_Crash_Laptop> well i am trying to install ffmpeg..
[11:33] <sacarasc> No, Ubuntu does NOT have ffmpeg any more.
[11:34] <brontosaurusrex> _Crash_Laptop, either compile it or use the gusari static builds
[12:13] <SimAV> Hi! I want to produce a livestream using ffmpeg/ffserver and wonder whether there is a way to specify a "fallback" feed that is used as input for the stream as long as the feed that will be live isn't feeded.
[12:15] <maksimkaaa> Hi, i add text on some frames with draw option specifying the X,Y coordinates for my text, the text is printed successfully on the video but there is a blinking cursor that keeps appearing on random areas of the video..! Isn't there a way to remove that blinking cursor?
[13:30] <oakwhiz> I'm trying to make a video out of an animated GIF and a FLAC file. The length of the FLAC is a multiple of the length of the GIF. How do I repeat the GIF as many times as is necessary to fill the length of the FLAC?
[13:30] <oakwhiz> I'm thinking it might be -loop but I'm still reading through man pages
[13:41] <iamtakingiteasy> hello, i am thinking of writing a ffmpeg video filter, which will convert incoming image into ascii-art using aalib, but i am not very sure from where to start. i've found a nice guide overviewing general aspects of writing filter for libavfilter component of ffmpeg: http://wiki.multimedia.cx/index.php?title=FFmpeg_filter_HOWTO ; but it is not covering it in great detail and missing complete
[13:41] <iamtakingiteasy> example sources. is there any other guides, especially covering ffmpeg build-process integration and development-time simplifications of build routine?
[13:43] <oakwhiz> iamtakingiteasy: you should look into libav/avconv, it seems to be some kind of fork of ffmpeg
[13:44] <sacarasc> It has less features than ffmpeg. \o/
[13:44] <oakwhiz> Sometimes I come across files that don't work in ffmpeg but work in avconv, or vice versa
[13:45] <oakwhiz> damn, -loop 1 just returns "Option loop not found."
[13:45] <sacarasc> -loop_input maybe?
[13:45] <sacarasc> And also -shortest
[13:46] <oakwhiz> -loop_input 1 -shortest gives Unrecognized option 'loop_input'.
[13:46] <sacarasc> What version are you using?
[13:46] <oakwhiz> -loop 1 -shortest gives "Option shortest (finish encoding within shortest input) cannot be applied to input file [...] -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to."
[13:47] <oakwhiz> ffmpeg version 2.1.4
[13:47] <oakwhiz> built on Feb 25 2014 08:24:47 with gcc 4.8.2 (GCC) 20131212 (Red Hat 4.8.2-7)
[13:48] <oakwhiz> You're supposed to put those arguments at the beginning before the inputs, right? e.g. -loop 1 -shortest -i whatever.gif
[13:49] <sacarasc> Yeah.
[13:49] <sacarasc> Well, -shortest is an output option, maybe.
[13:50] <iamtakingiteasy> hmm
[13:50] <oakwhiz> -loop 1 on the input and -shortest on the output gives "Option loop not found."
[13:50] <oakwhiz> I might try another method of accomplishing this
[13:50] <iamtakingiteasy> looks like there was a division in ffmpeg' developers which made separate project named avconv
[13:51] <iamtakingiteasy> but where should i go?
[13:51] <oakwhiz> Since I know that the two files share a least common multiple, I can repeat them until they match in length
[13:52] <sacarasc> iamtakingiteasy: If you use ffmpeg, here. If you use libav/avconv, #libav.
[13:54] <iamtakingiteasy> i am a bit confused about "libav" things, since ffmpeg seems to be using libavfilter as well. is that wrong?
[13:55] <sacarasc> Libav is a project, libav* is the group of libraries both ffmpeg and libav create/use.
[13:56] <iamtakingiteasy> aha, so it should be ffmpeg/libav agnostic if i want to write a libavfilter, which means it be available for both projects; but is there any separate channel dedicated to libav* libraries?
[13:57] <iive> iamtakingiteasy: ffmpeg merges almost everything from libav, so it libav superset.
[13:57] <iive> however afaik the people who designed and implemented avfilter are still ffmpeg developers, and ffmpeg have more filters.
[13:58] <iive> this means that if you write the filter, it should be trivial to port it to the other project.
[14:00] <iamtakingiteasy> okay, thanks for clarifying it
[14:07] <oakwhiz> It turns out that I have no idea how to properly use imagemagick, so I just copied and pasted the same command line argument 2^5 times and now I have a 40MB GIF.
[14:11] <oakwhiz> according to "identify -verbose test.gif | grep -i elapsed" it's within a few seconds of the correct length... close enough for me
[14:12] <oakwhiz> probably should have made that "identify -verbose test.gif | grep -i elapsed | tail"
[14:12] <iive> oakwhiz: what exactly are you doing?
[14:12] <oakwhiz> I have an animated GIF and a sound
[14:12] <oakwhiz> they are different lengths
[14:12] <oakwhiz> I want them to loop
[14:13] <oakwhiz> Their lengths share a least common multiple.
[14:13] <iive> aha
[14:13] <oakwhiz> I wanted to make it so that they match up exactly on that LCM.
[14:14] <oakwhiz> Unfortunately there is no built in functionality for this, so I had to manually expand each input file
[14:16] <oakwhiz> here goes nothing
[14:17] <oakwhiz> Wow it actually worked
[14:18] <oakwhiz> Actually scratch that... halfway through it gets stuck
[14:18] <oakwhiz> :-(
[14:22] <iive> yeh, loop seems to be  gif output option and imd2dec input option.
[14:30] <oakwhiz> One of my clips has a bit of a problem... the LCM of two adjacent prime numbers is usually a pretty big number.
[14:31] <oakwhiz> video is 251 seconds long and audio is 193 seconds long... LCM is 48433 seconds
[14:31] <oakwhiz> 13 hours...
[14:32] <oakwhiz> lol
[14:40] <Hello71> iamtakingiteasy: originally there was ffmpeg which had libraries called libav*, then some people split and called their fork libav to confuse everyone
[14:41] <oakwhiz> now instead of having 1 broken video converter with a standard interface, we can divide our efforts amongst 2 broken video converters with different interfaces :-)
[14:43] <oakwhiz> Joking of course, ffmpeg has saved my butt on numerous occasions when neither random windows programs nor VLC Player worked
[14:54] <sacarasc> zack6849: ffmpeg -i blah -target dvd output.mpg
[14:54] <sacarasc> Sorry.
[14:54] <sacarasc> zybi1: ffmpeg -i blah -target dvd output.mpg
[14:54] <sacarasc> Maybe -target dvd-pal, as you seem to be in Germany.
[14:54] <zybi1> ntsc
[14:55] <Fjorgynn> so
[14:55] <Fjorgynn> never used dvd-pal
[15:13] <zybi1> re
[15:13] <Fjorgynn> re
[15:23] <zybi1> how do i make ffmpeg combine audio (flac) with video (avi) to lossless mkv-video?
[15:23] <sacarasc> ffmpeg -i blah.flac -i blah.avi -c copy output.mkv
[15:28] <g0twig> Hello
[15:29] <g0twig> I need to compress 250GB to something smaller, in a lossless way, any suggestion? It should be played back on a samsung smart tv
[15:29] <sacarasc> What does your TV support?
[15:29] <g0twig> sacarasc: second
[15:29] <g0twig> sacarasc: http://www.samsungdforum.com/Guide/rel00010/index.html
[15:31] <sacarasc> Try -c:v libx264 -q 0, but lossless video is usually VERY big.
[15:31] <g0twig> sacarasc: but half the size, or?
[15:31] <g0twig> sacarasc: you mean I can use h264 with lossless?
[15:31] <sacarasc> Yes.
[15:31] <g0twig> sacarasc: I compressed lossy h264 250 gb to 21 gb already
[15:32] <g0twig> sacarasc: how big do you think it will become
[15:33] <sacarasc> I don't know.
[15:33] <zybi1> thanks sacarasc !!
[15:33] <g0twig> sacarasc: Maybe its not even a compression?
[15:33] <g0twig> Maybe it will just change the container :(
[15:33] <sacarasc> It does compress, but you almost definitely want to re-encode.
[15:34] <sacarasc> As iive is saying in #mplayer. Lossless means something rather specific in these quarters.
[15:45] <zybi1> got an error: http://www.pasteall.org/52379
[15:46] <sacarasc> zybi1: You didn't use my command. :D
[15:56] <zybi1> sacarasc: I tried now like you wrote: http://www.pasteall.org/52380
[15:56] <zybi1> still no luck!
[16:05] <zybi1> I found this: http://askubuntu.com/questions/287461/using-avconv-when-remuxing-to-mkv-is-there-a-way-to-fix-packed-avi-input-files
[16:05] <zybi1> maybe it's for ffmpeg the same?
[16:05] <zybi1> i have to first convert to mp4 and then to mkv?
[16:08] <zybi1> this also doesn't work
[16:08] <zybi1> http://www.pasteall.org/52381
[16:16] <zybi1> re
[16:16] <zybi1> i dunno why, but what finally worked is$ ffmpeg -fflags +genpts -i audio.flac -fflags +genpts -i video.avi -c copy out-va.mkv
[16:17] <zybi1> and how to output ac3 and mpeg for authoring a DVD later? is it possible in one command or do i have to do it separately?
[16:18] <c_14> just create another output with different output options
[16:19] <zybi1> is this a good howto: http://ffmpeg.gusari.org/viewtopic.php?f=25&t=1235
[16:19] <zybi1> ?
[16:19] <luc4> Hello! Im trying to transcode an input from stdin with cat in.mp4 | ffmpeg -i - -acodec ac3 -vcodec h264 -preset veryslow -crf 18 out.mov. It seems this is not working. Is the command wrong?
[16:19] <zybi1> okay: like ffmpeg -i in.mkv -o video.mpeg -o audio.ac3
[16:19] <zybi1> but is it DVD-compatible ntsc format then?
[16:20] <zybi1> should i use somthing like -target ntsc-dvd
[16:20] <c_14> I'd probably just use -target ntsc-dvd, it usually just works.
[16:20] <c_14> ffmpeg doesn't have a -o option iirc
[16:21] <sacarasc> luc4: MP4 has its index at the end of the file, and so catting it often won't work.
[16:21] <luc4> sacarasc: oh so not possible. Ok, thanks!
[16:21] <sacarasc> *usually has its
[16:23] <luc4> sacarasc: any idea if it is possible to get some kind of progress bar when transcoding?
[16:24] <sacarasc> Not sure you can without patching ffmpeg.
[16:24] <luc4> sacarasc: ok, no problem then.
[16:24] <luc4> sacarasc: thanks.
[16:28] <zybi1> c_14: the resulting file by using -target ntsc-dvd has crappy quality, it's totally blurry
[16:34] <c_14> I just tried it on a couple of my files and the output looks fine.
[16:40] <zybi1> can't I tune the bitrate higher somehow or does this command use the highest possible bitrate for dvd automatically?
[16:41] <zybi1> or some animation mode for fast movements and so on
[16:51] <c_14> I'm pretty sure the dvd target automatically sets the highest bitrate it can.
[16:52] <zybi1> ok
[17:20] <Fjorgynn> so what's up with this -target pal-dvd?
[17:20] <g0twig> video stabilization is included in ffmpeg, how can I use it on an existing video, and produce a new video with same codec/same container
[17:27] <zybi1> what does this mean? while creating ntsc-dvd-compliant mpg file from mkv: [swscaler @ 0x245ac00] deprecated pixel format used, make sure you did set range correctly
[17:31] <zybi1> Fjorgynn: what's your question? there is pal or ntsc dvd, that's what it means, you gotta decide wheater your dvd will have 29.97fps or 25fps
[17:31] <Fjorgynn> zybi1: no I just saw someone earlier talking about dvd or pal-dvd
[17:50] <user3> why can't i change the title with this, and how can i speed it up? ffmpeg -i in.mp3 -metadata title="abc" out.mp3
[17:56] <user3> it takes a long time to runs and has not effect on the title
[17:56] <user3> no effect
[18:01] <sfan5> user3: use "-c copy" to avoid re-encoding anything
[18:01] <user3> trying it...
[18:04] <user3> it definitely doesn't change the title much faster, any alternative?
[18:05] <user3> i mean this still won't change the title
[18:07] <user3> but whatever this does it does it much faster
[18:09] <sacarasc> user3: Which OS are you on?
[18:09] <user3> windows 7
[18:09] <sacarasc> Try using something like mp3tag.
[18:09] <user3> is that included in the ffmpeg package?
[18:10] <sacarasc> No, it's a different programme.
[18:10] <user3> ok, googling it out...
[18:12] <relaxed> user3: try ffmpeg -i in.mp3 -map_metadata -1 -metadata title="abc" out.mp3
[18:12] <user3> trying...
[18:12] <relaxed> you'll probably have to add all the tags, though
[18:13] <Fjorgynn> ligga
[18:15] <user3> still doesn''t work but i can try it on linux too, I have to reboot...
[18:15] <relaxed> user3: your first example works here
[18:15] <relaxed> which version are you using?
[18:16] <user3> maybe the problem is with my file, maybe i mess too much with it
[18:16] <user3> messed
[18:16] <user3> i'll try again, this time with a brandnew file...
[18:21] <user3> i'd better try i on linux,bye folks
[18:57] <user3> update! ffmpeg -i in.mp3 -metadata title="abc" out.mp3 does indeed change the title (both Windows 7 and Linux) when i check with "ffmpeg -i out.mp3" Initially i thought it didnt because i had only looked at what Windows Media Player shows i.e. the file name instead of the metadata title
[19:20] <debianuser> _Crash_Laptop: You can get ffmpeg from some ppa, e.g. jon-severinsson's (apt-add-repository ppa:jon-severinsson/ffmpeg) or samrog131's (add-apt-repository ppa:samrog131/ppa). I've just googled them, never tested. :)
[19:20] <debianuser> iamtakingiteasy: You can find some examples of libaa+ffmpeg in mplayer sources. I.e. mplayer actually decodes arbitrary file with ffmpeg and displays it as ascii art (mplayer -vo aa somefile.avi) It even supports colored-ascii-art (mplayer -vo caca somefile.avi)
[19:31] <zybi1> a film festival is asking to send the video works in  .mov, .mpg, .divx, .avi oder flash format - i don't know why they don't accept .mkv
[19:32] <zybi1> which container would you prefer and how to encode from .mkv lossless
[19:34] <relaxed> zybi1: try, ffmpeg -i input.mkv -map 0 -c copy out.mov
[19:35] <sfan5> zybi1: there is no lossless re-encode, changing the container format is called re-muxing (IIRC)
[19:36] <zybi1> relaxed: it says "file may be unplayable" see http://www.pasteall.org/52383
[19:38] <sfan5> seems like .mov does not support flac directly
[19:38] <sfan5> s/directly//
[19:38] <relaxed> zybi1: ffmpeg -i input -c:v copy -c:a pcm_s16le output.mov
[19:39] <relaxed> still lossless
[19:39] <sfan5> depending on the transmission medium uncompressed audio may be a bad idea
[19:42] <relaxed> I doubt you want to step on the audio for a film festival entry.
[19:44] <sfan5> Doesn't mov support ALAC?
[19:44] <debianuser> zybi1: Don't know whether mov can contain flac, but, maybe, avi can...
[19:45] <sfan5> yes, avi can contain flac
[19:45] <sfan5> ..and mov supports ALAC
[19:48] <relaxed> does recent Windows support alac out of the box?
[19:49] <zybi1> final command line suggestions?
[19:49] <zybi1> :)
[19:50] <zybi1> good quality and good playability by all systems (osx, win, linux)
[19:50] <zybi1> I guess would be nice
[19:50] <debianuser> ffmpeg -i input -c copy output.avi
[19:51] <relaxed> zybi1: I would say my command is the safest bet
[19:52] <debianuser> Have anybody joined many different (wav, ogg) audio files into one with ffmpeg? I'm trying to, and get some errors whatever I try even for a small subset. Either lots of "invalid dropping", or "Non-monotonous DTS". And resulting wav file is either smaller than expected or sometimes larger than expected.
[19:53] <sfan5> join = concat?
[19:53] <zybi1> ok relaxed thanks!
[19:53] <debianuser> sfan5: yes
[19:54] <debianuser> Command `echo -e "file 1.wav\nfile 2.ogg\nfile 3.wav" | ffmpeg -f concat -i - -ac 2 -ar 48000 0.wav` gives incorrect output file and lots of errors: pastebin.com/KQQSCGM4
[19:54] <debianuser> Command `ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex 'concat=n=3:v=0:a=1[a]' -map '[a]' -ac 2 -ar 48000 0.wav` gives a short file (10.92+3.43+12.89!=23.83) and many errors: pastebin.com/nEYL9RLa
[19:54] <sfan5> did you try to convert all three files to wav first?
[19:54] <debianuser> (I can share the files if needed)
[19:55] <debianuser> Two of them are already wav files. :)
[19:59] <Fjorgynn> 22.05 vs 44.1 khz?
[20:00] <debianuser> sfan5: Just tried. I get the errors separately: "Non-monotonous DTS" while unpacking ogg into wav and lots of "invalid dropping" and other errors when joining them.
[20:00] <sfan5> maybe your .ogg is broken
[20:02] <debianuser> Well, maybe (then how can I "fix" it or ignore errors?). But wav files can't be broken, right? These 3 test files: ge.tt/1Eg60nl1/v/1
[20:07] <sfan5> debianuser: ffmpeg -i 2.ogg 2.wav && echo -e "file 1.wav\nfile 2.wav\nfile 3.wav" | ffmpeg -f concat -i - -ac 2 -ar 48000 0.wav    seems to work
[20:10] <debianuser> sfan5: gives me 47.35 seconds file. Seems too long for 10.92+3.43+12.89 sources :)
[20:11] <sfan5> I guess you'll have to resort to using something like Audacity then; I don't know of a way to fix this
[20:23] <debianuser> Maybe it's a bug... Ill try reporting it to #ffmpeg-devel
[21:30] <cuba> anyone here have a working example for using x254 with libav?
[21:30] <cuba> *has
[21:31] <cuba> x264.h i meant
[21:34] <JEEB> demux, decode a picture, convert pix_fmt if needed, feed decoded picture to x264, rince and repeat
[21:35] <JEEB> would be pretty much a generic example of lavf/lavc (and possibly swscale) usage, and then pretty much an example of generic libx264 usage
[21:35] <JEEB> there's also the libx264 wrapper in lavc that you can look at as well
[21:46] <SimAV> Hi, I would like to produce a livestream using ffmpeg/ffserver
[21:47] <SimAV> is there a way to specify a "fallback" feed that is used as input whenever the "live"-feed isn't sent yet?
[21:48] <SimAV> I haven't found such an option in ffserver config yet
[21:48] <SimAV> has anyone in here achieved such an setup or maybe found a workaround?
[22:59] <dw2> hi so I want to convert a folder from 10 bit to 8 bit to play on my raspberry pi, is there a way to do a whole folder
[22:59] <dw2> ffmpeg -i /home/dw/video/sao -vcodec libx264 -crf 20 -acodec copy /home/dw/video/sao8.mkv
[23:00] <dw2> that's what I tried
[23:01] <c_14> afaik not ffmpeg-internal. But something like `for file in /path/to/dir/*; do ffmpeg -i "$file" [options] "/path/to/outdir/$file"; done' should work just fine
[23:02] <klaxa> pssst sao was bad
[23:03] <dw2> yeah I know :p
[23:03] <dw2> also converting some good stuff like inferno cop
[23:03] <sfan5> you might also want to use -c copy and -map 0 to make sure attachments are copied too (pretty important if there are subs)
[23:03] <dw2> thanks
[23:03] <klaxa> might as well just hardsub when you are encoding for the pi anyway
[23:03] <sfan5> ^
[23:04] <dw2> I'm not a media expert so idk how to do that
[23:04] <sfan5> https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo
[23:06] <sfan5> would be ffmpeg -i $file -c copy -vcodec libx264 -vf subtitles=$file out/$file then
[23:06] <klaxa> you will need to extract the font-files though
[23:06] <klaxa> because ffmpeg will not find them in the embedded mkv
[23:07] <c_14> IIRC, it actually does.
[23:07] <c_14> At least for me.
[23:07] <sfan5> ^ same IIRC
[23:07] <klaxa> oh? it has been a while since i hardsubbed something
[23:11] <sfan5> klaxa: just tested it again, ffmpeg does find the font files in the mkv
[23:11] <klaxa> coolbeans
[23:13] <dw2> so when I tried
[23:13] <dw2> home/dw/videos/sao/*; do ffmpeg -i "$file" -c -map 0 -vcodec libx264 -crf 21 -acodec /home/dw/videos/$file
[23:13] <dw2> it said unexpected token do
[23:14] <c_14> -c -map is invalid, -c copy -map
[23:14] <dw2> bash: syntax error near unexpected token `do'
[23:14] <c_14> and you forgot the for file in
[23:15] <sfan5> dw2: I'd suggest doing hardsub just to be sure
[23:17] <dw2> what's for file in?
[23:17] <c_14> It's the bash for loop syntax: `for file in /path/to/dir/*; do [command]; done
[23:18] <dw2> gotcha
[23:23] <dw2> after I hit enter all there is on that terminal is an > does that mean it's working?
[23:24] <c_14> that means you're missing something. ie that thy syntax isn't finished
[23:25] <dw2> is it missing something like an output command
[23:25] <sfan5> you probably forgot the "done"
[23:28] <dw2> for file in home/dw/videos/sao/*; do ffmpeg -i "$file" -c copy -map 0 -vcodec libx264 -crf 21 -acodec /home/dw/videos/$file done
[23:28] <dw2> still dosn't work :(
[23:28] <c_14> ;done
[23:29] <c_14> the semicolon is important
[23:29] <dw2> wow, I need to learn bash
[23:43] <elec29a> hi hi could someone help me with ffmpeg on windows cygwin android ?
[23:56] <elec29a> n00ne here for a helping hand ?
[23:58] <iive> don't ask to ask, just ask your question and wait long enough for answer or the entropy death of the universe, whatever comes first.
[23:59] <elec29a> ok thx i try to get it in r3questet format
[00:00] --- Mon Jun 23 2014