[Ffmpeg-devel-irc] ffmpeg.log.20130203

[burek]

[00:28] <Youka> could it be that ASS subtitle decoding is buggy? i tried it, but it always skip first 15 lines
[00:29] <Youka> https://github.com/Youka/NyuFX/blob/master/src/lva.cpp#L493
[01:46] <taylanub> Today I compiled from git and ffplay (and mplayer) give corrupted output.  ffplay error output:  http://sprunge.us/gOBS
[01:46] <taylanub> Weird thing is, libav has the same issue (tested with mplayer, since libav has no ffplay).
[03:09] <pietro10> is ther ea way I can capture the audio from either my speakers only (not the mic) or from a specific program? -f alsa -i pulse records from my mic. Thanks.
[03:09] <klaxa> pietro10: open pulseaudio volume control and move the record stream to the monitor of your soundcard
[03:11] <pietro10> I could do that but I mean with ffmpeg, so I don't have to do that each time if I actually want to use my mic
[03:14] <klaxa> you could script it with pacmd or something, i think there is a way to choose that with ffmpeg itself though i don't remember it
[03:25] <pietro10> ok thanks
[03:59] <blue_misfit> anyone have experience with Smooth Streaming?
[06:05] <makara> I want to check that my m4a's aren't ALAC. Why doesn't this work: ffprobe *.m4a | grep alac
[06:20] <drv> makara: ffprobe output is on stderr, you could use e.g. ffprobe *.m4a 2>&1 | grep alac
[06:23] <makara> drv, could you please explain "2>&1"
[06:23] <drv> it redirects stderr to stdout
[06:33] <makara> drv, it doesn't work
[06:33] <makara> it will output on screen, but can't pipe to grep or a file
[06:42] <makara> drv, it works, thanks
[14:18] <paste> hi, is there an ffmpeg player? reason is when using yadif with mplayer clearly field 1 is incorrect, but when encoding with ffmpeg I get jutters regardless if I choose top (0) field or bottom (1) field. Or I'm DIW?
[14:19] <saste> paste: ffplay
[14:19] <paste> cool thanks.
[14:20] <paste> hmm.
[14:20] <paste> looks like its not available.
[14:21] <paste> ugh
[14:21] <paste> nvm
[14:21] <paste> bed time.
[14:21] <paste> thanks.
[14:30] <jastr> hi.. can anyone help me with a raw video format issue?
[14:30] <jastr> when i run this ./ffmpeg -i MyStream_0.f4v -vb 1200k -f rawvideo -t 00:05:00 -pix_fmt yuv420p -s 832x468 - | ./ffmpeg -f rawvideo -r 30 -pix_fmt yuv420p -s 832x468 -crf 0 -i - out.mp4  the output keeps being in 200kb/s i've tried -qscale and -b but either helps
[14:31] <jastr> now it detects the input to be 1200kb after i set the -vb on the raw input but the output it writes is always approx 260kb/s no matter what option i give the output
[14:32] <jastr> here is the pastebin http://pastebin.com/hz27cXEj
[14:33] <spyvingen> anyone know how to set swe as first choice in winff and if swe dont exist then eng should be choosen?
[14:34] <spyvingen> audio that is
[14:35] <Mavrik> jastr: why are you trying to set bitrate on input?
[14:35] <Mavrik> jastr: input has a bitrate of what it has, forcing anything won't change things
[14:35] <Mavrik> plus... why two ffmpeg processes?
[14:36] <Mavrik> what are you trying to achieve?
[14:36] <jastr> well when i didn't use the -vb it detected the input at 116812
[14:37] <Mavrik> it doesn't matter
[14:37] <Mavrik> you can't change what you already have :P
[14:37] <jastr> Mavrik i have a movie that looks fine when played back in quicktime but it's a movie recorded in 25fps on a 30fps canvas so it looks a bit stuttering when played back in flash. So what i wanted was ffmpeg to kind of playback the movie and reencode the video stream. If i reencode from file to file in ffmpeg either at 25fps or 30fps it's the same.
[14:38] <jastr> i know but the input is 1200kb
[14:39] <Mavrik> again, last time: detection doesn't friggin matter
[14:39] <Mavrik> the decoder reads stream as it is
[14:40] <Mavrik> plus, you're probably looking at totally wrong output anyway
[14:40] <Mavrik> now.
[14:40] <Mavrik> WHY are you using 2 ffmpeg processes that essentially do the same a simple one-process command line would do?
[14:40] <Mavrik> also, crf 0? losless?
[14:40] <jastr> thought i had to. that's how i did it in the old days.
[14:41] <Mavrik> hmm :)
[14:41] <Mavrik> jastr: so if I understand this correctly, you're just trying to convert 25fps video to 30 fps video?
[14:41] <jastr> the reason for the two processes was that i just wanted ffmpeg to playback the movie and recapture it hoping to get a little more smooth result
[14:42] <jastr> no i'm trying to smooth it. It is 25fps and should be 25fps BUT the program that encoded the file was working with 30fps so it's not smooth when it's played back in flash but quicktime handles the file fine
[14:44] <Mavrik> I see.
[14:44] <Mavrik> well, your command line won't have the result you seek
[14:44] <jastr> i see.. :D any other commandlines that could have?
[14:45] <Mavrik> it's equivalent of just reencoding video with parameters with added overhead of piping :)
[14:45] <spyvingen> is it possible to choose what audio languege to choose
[14:45] <spyvingen> *to rip
[14:45] <Mavrik> jastr: well, you can try just dropping frames
[14:45] <Mavrik> down to 25fps
[14:45] <jastr> also when the file is 25fps?
[14:46] <Mavrik> oh doh
[14:46] <Mavrik> hmm
[14:46] <Mavrik> now that's quite a problem
[14:46] <jastr> hehe exactly.. that's what the program did and now i just want to smoothen it as much as possible
[14:48] <Mavrik> ffmpeg -r 30 -i yourfile.flv -codec:v libx264 -crf 20 -codec:a copy -r 25 output.mp4
[14:48] <Mavrik> try this :)
[14:55] <jastr> that dropped a lot of fps but didn't really get more smooth. more like the other way around.
[14:55] <jastr> maybe -f 25 and exporting a 30fps?
[14:56] <Mavrik> probably, but I'm afraid you already lost frame info :\
[14:59] <jastr> yeah i know.. that's why people who don't know about video encoding shouldn't do it.. arhh
[15:34] <Guest17665> Hi I'm using this options to convert avi to mp4:
[15:35] <Guest17665> "-i $INPUT -vcodec libx264 -threads 0 -r 25 -g 25 -b 500k -bt 500k -acodec libvo_aacenc -ar 44100 -ab 64k -y $OUTPUT"
[15:35] <Guest17665> But I get:
[15:35] <Guest17665> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[15:36] <Mavrik> Guest17665: there has to be a message actually explaining what's wrong.
[15:37] <Guest17665> Mavrik: you mean, to pastebin it?
[15:37] <Mavrik> well
[15:38] <Mavrik> you can read it as well :P
[15:40] <Guest17665> ok
[15:41] <TheLink> if I set hqdn3d=:::0, does this leave the first 3 parameters at default and the fourth at 0?
[15:41] <Guest17665> http://pastebin.com/k8t58KGi
[15:42] <Guest17665> What are advantages of avconv vs ffmpeg
[15:42] <Mavrik> *shrug*
[15:43] <Mavrik> TheLink: hmm, I'm not sure that's even valid syntax
[15:43] <Mavrik> TheLink: newer ffmpegs prefer hqdn3d=chroma_tmp=0 type syntax
[15:43] <Mavrik> Guest17665: [libvo_aacenc @ 0x1ba5b60] Unable to set encoding parameters
[15:43] <Mavrik> is seems your audio encoder doesn't initialize properly
[15:44] <Mavrik> ah
[15:44] <Mavrik> Guest17665: you have 5.1 audio
[15:44] <TheLink> well, ffmpeg didn't throw an error but I guess your syntax is much cleaner
[15:44] <TheLink> thanks
[15:44] <Mavrik> Guest17665: libvo_aacenc doesn't support 5.1 audio
[15:48] <Guest17665> so i should pick some other acodec?
[15:48] <Mavrik> yep, you'll need another aac encoder if you want to keep 5.1 audio
[16:14] <Guest17665> well what's 5.1 audio?
[16:16] <TheLink> multi channel audio
[16:16] <TheLink> in your case 6 channels
[16:21] <Guest17665> why it says: Unknown encoder 'libfaac'
[16:25] <Guest17665> I get now:
[16:25] <Guest17665> [ac3 @ 0x2684e60] frame sync error
[16:25] <Guest17665> Error while decoding stream #0:1
[16:25] <Guest17665> Application provided invalid, non monotonically increasing dts to muxer in stream 1: 22906 >= 22871
[16:25] <Guest17665> av_interleaved_write_frame(): Invalid argument
[16:25] <Guest17665> I change acodec  to libfaac
[16:34] <LithosLaptop> ewww :)
[16:34] <Guest17665> and what is that:
[16:34] <Guest17665> encoder 'aac' is experimental and might produce bad results.
[16:34] <Guest17665> Add '-strict experimental' if you want to use it.
[16:34] <Guest17665> Or use the non experimental encoder 'libfaac'.
[16:35] <Guest17665> literally -strict experimental ?
[16:36] <LithosLaptop> the quality of the experimental aac encoder is worse than libfaac
[16:36] <LithosLaptop> and libfaac aint that great
[16:36] <Guest17665> I've built it with
[16:36] <Guest17665> ./configure --enable-libx264 --enable-libvo-aacenc --enable-libfaac --enable-gpl --enable-nonf
[16:36] <Guest17665> ree --enable-version3
[16:36] <Guest17665> why say not libfaac?
[16:37] <LithosLaptop> not really transparent at 128Kbit/s for 2 channel audio
[16:37] <LithosLaptop> thats why I don't use it
[16:37] <LithosLaptop> even libmp3lame will give you better quality than libffac
[16:38] <LithosLaptop> try to use fdkaac in ffmpeg
[16:39] <LithosLaptop> libvo-aacenc is the worst btw
[16:39] <Guest17665> LithosLaptop: well, I need whatever gets the job done
[16:40] <LithosLaptop> --enable-libfdk_aacenc
[16:40] <LithosLaptop> oh
[16:41] <LithosLaptop> well I haven't really tried more than 2 channels before
[16:42] <Guest17665> Unknown option "--enable-libfdk_aacenc".
[16:43] <Guest17665> oh it's libfdk_aac
[16:44] <Guest17665> oops not found
[16:54] <taylanub> So is ffmpeg from git broken for anyone else ?  Doesn't play H.264, gives corrupted video.
[16:58] <Guest17665> taylanub: it isn't suppoused to play, it's for converts
[16:58] <taylanub> Eh ?  Doesn't mplayer use libav to decode H.264 for playing ?  What does ffplay do ?
[17:02] <taylanub> I'll be compiling old commits and see what commit broke it.  Assuming the problem isn't elsewhere.
[17:03] <Guest17665> LithosLaptop: how do I use libmp3lame?
[17:03] <Guest17665> instead of aac
[17:03] <LithosLaptop> -acodec libmp3lame
[17:04] <Guest17665> Unknown encoder 'libmp3lame'
[17:05] <LithosLaptop> strange
[17:05] <LithosLaptop> maybe it is not built with it
[17:07] <Guest17665> oh
[17:20] <Guest17665> I cloned git://github.com/mstorsjo/fdk-aac.git how to build it?
[17:22] <Guest17665> after I install libmp3lame do I rebuild avconv?
[17:24] <saste> Guest17665, avconv != ffmpeg
[17:25] <saste> and we can't support avconv here
[17:25] <Guest17665> saste: sorry
[18:18] <Zeranoe> Is qt-faststart now part of FFmpeg? If so, how do I use it? If not, are they any other ways to move the moov atom?
[18:19] <JEEB> there's a movflag for that IIRC
[18:19] <beastd> Zeranoe: movflags IIRC
[18:20] <JEEB> movflags +faststart was it?
[18:20] <beastd> "-movflags faststart", according to the fine manual
[18:21] <JEEB> :)
[18:24] <taylanub> OK, I'm back to some December commit, and ffmpeg works again. :\
[18:24] <taylanub> This is OpenBSD 5.1 by the way.
[18:25] <beastd> taylanub: i hope you did ffbisect. might have been lots of work otherwise
[18:26] <taylanub> What's that ?  I use the magit Emacs library to visualize the log and reset from there.
[18:28] <beastd> taylanub: ffbisect is a script to ease biscecting ffmpeg code (because of merges). bisecting is the process of choosing an old (working commit) can be way back in time. and than doing a binary search on the interval
[18:29] <beastd> anyway could you pinpoint exact commit that is causing you trouble?
[18:36] <taylanub> I have yet to find that out ..  Indeed, traversing the history seems hard.
[18:36] <taylanub> I'm on some December 22 commit now, but the log also shows commits from Jan 23. :\
[18:36] <taylanub> (Not very good at git yet.)
[18:39] <beastd> taylanub: maybe you should try ot bisect from  now on
[18:40] <beastd> i assume you have git clone of ffmpeg and found a commit in the past that worked your you (e.g one in the beginning of december)
[18:44] <beastd> taylanub: are you testing for the regression with ffplay?
[18:47] <taylanub> Yes.
[18:51] <taylanub> Is there a way to not re-compile everything when changing commits ?  A full compilation takes 5-10 minutes.
[18:51] <beastd> taylanub: not sure that is a goode idea
[18:51] <taylanub> OK then, will just take my time.
[18:51] <beastd> taylanub: try it this way:
[18:52] <beastd> taylanub: remember the bad revision or just checkout the bad revision
[18:53] <beastd> then execute tools/bisect-create
[18:53] <beastd> tools/ffbisect start
[18:54] <beastd> tools/ffbisect need ffplay
[18:54] <beastd> tools/ffbisect bad # marks current revision as bad
[18:54] <beastd> oop wrong
[18:54] <beastd> tools/ffbisect good # assuming you checked out a revision where it was working for you
[18:55] <beastd> tools/ffbisect bad master
[18:55] <beastd> then it should chose a commit for you to test
[18:56] <beastd> make disclean; ./configure; make  # use -j4 or whatever brings speed up on your systems, i assume you have multiple cores
[18:56] <taylanub> -j4 tends to randomly break; or is that just due to bad Makefiles ?
[18:57] <beastd> taylanub: too bad it shouldn't :(
[18:57] <taylanub> I mean in-general .. dunno about ffmpeg.
[18:57] <beastd> test if it works. depending if it works/or doesn't work you shoult then mark the revision as good or bad with: tools/ffbisect good/bad
[18:58] <beastd> continuing that cycle you should arrive at the offending commit in i guess about 10 steps
[18:58] <taylanub> OK
[18:59] <beastd> taylanub: i don't know. GNU make with -j<n> usually works for me. i remember that on windows i had some warnigns from make that some assertions weren't met. on linux it always worked for me so far. and i use it on ffmpeg all the time.
[19:00] <taylanub> I'll just use it then.
[19:01] <beastd> taylanub: good luck! if you found the offending commit please read http://ffmpeg.org/bugreports.html and report it as regression
[19:02] <taylanub> OK
[20:49] <taylanub> On another note, the configure script bails out with "too many arguments" for me.  Bash works.  I guess I should just report a bug.
[20:55] <taylanub> The first bad commit could be any of:
[20:55] <taylanub> 528878ee7b377e23a194d7c801571d97793047e0
[20:55] <taylanub> 9c555bca68cf3f168c2e4ffcb5b4dacbda8366bf
[21:00] <taylanub> [528878ee7b377e23a194d7c801571d97793047e0] openbsd: configure: Stop enabling PIC by default
[21:00] <taylanub> Since it mentions OpenBSD, I'm betting on that.
[21:00] <taylanub> Oh, in fact the other one just merges that.
[21:05] <durandal_1707> taylanub: you reverted that commit and confirm it fixes your problem?
[21:05] <durandal_1707> or you are just using old release with broken shell?
[21:06] <taylanub> durandal_1707: Excuse me ?  The configure problem is something unrelated; I've always had it and just change the script to use bash.
[21:06] <taylanub> The PIC commit is likely what introduced corrupt H.264 video playback.
[21:07] <taylanub> (And re. configure, OpenBSD's ksh is a POSIX-compliant sh, so it would be configure script that's somehow broken.)
[21:07] <taylanub> (Assuming that this ksh is not simply buggy itself.)
[21:08] <durandal_1707> taylanub: not really, there is ksh bug that got fixed in later version
[21:08] <taylanub> Oh, when exactly ?  I'm using OpenBSD 5.1.
[21:08] <durandal_1707> how old is that?
[21:08] <taylanub> The previous release .. they release approximately every 6 months, so it would be maximally about a year old.
[21:10] <taylanub> May 1 2012 apparently.
[21:26] <Jiyuu|C2> is it possible to use the libass subtitles filter so it will use an mkv for the source?
[21:27] <Jiyuu|C2> i.e. avoiding the need to use mkvextract to get a subtitles file out of the mkv file
[21:27] <Jiyuu|C2> also
[21:27] <Jiyuu|C2> is is possible to use attached font files or do i have to extract and install them?
[21:34] <trosh> i am on linux, and when i use ffmpeg to convert a part of a movie to still frames, using -ss for starting time, i have to wait for a hell of a long time until it reaches the desired time and then convert normally. is this normal behaviour? thanks.
[21:34] <Fjorgynn> nah
[21:35] <nronksr> trosh: try placing the -ss before the input file for a faster seek.  add -ss for exact accuracy after the input file for a slower seek once you are in the general timeframe vicinity.
[21:35] <trosh> oh thanks
[21:35] <trosh> is -ss relative ?
[21:36] <Fjorgynn> yeah -ss hh:mm:ss -i file.avi -t hh:mm:ss
[21:37] <bkmz> hi guys, what is the exactly computation complexity of AAC-LC ?? where i can find complexity function for it
[21:37] <nronksr> Ideas on extracting 6 still images every second from a 59.96fps clip?
[21:42] <durandal_1707> nronksr: -vf select?
[21:56] <nronksr> durandal_1707: thanks, I thought that too; I was experiencing a typo that boggled me. XD
[22:02] <trosh> nronksr: just to be sure, -r 6 works in your case ?
[22:05] <nronksr> trosh: for 6 frames extracted every second?  I used this:  -vf 'select=not(mod(n\,6))'
[22:05] <vlt> Hello. Does anyone know what pkg I need to install additionally to ffmpeg to be able to use -vcodec libx264 on Ubuntu 12.04?
[22:05] <JEEB> vlt, libavcodec-extra
[22:06] <nronksr> vlt: ubuntu uses avconv by default, but yeah, libavcodec-extra and if I recall, something with 53 in it?
[22:06] <JEEB> and yeah, if you want to use the packaged libav (which is not ffmpeg, long story), you should use the avconv command instead of the ffmpeg one :)
[22:06] <JEEB> just a newer binary in that project
[22:07] <trosh> nronksr: how come?
[22:08] <vlt> JEEB, nronksr: Thank you!
[22:11] <nronksr> trosh: I'm not an expert by any stretch of the imagination and that was the way I've done it and it works?
[22:17] <vlt> :-( Installing libavcodecs-extra-53 breaks my whole pkg system on ubuntu o_O
[22:17] <vlt> I should try a chroot env &
[22:18] <JEEB> more like it wants to install packages that use it instead of the standard libavcodec package
[22:18] <JEEB> it's OK
[22:18] <JEEB> unless you have something weird in there
[22:18] <JEEB> if you look at the list of installed and removed packages you should see it :P
[22:19] <vlt> JEEB: (In the chroot env) apt is removing like 280 pkgs now.
[22:20] <JEEB> and it should be installing something instead of them, no?
[22:21] <vlt> JEEB: I dont know (and care) exactly now. I only know I cant do this on a production machine.
[22:21] <vlt> Hence the chroot.
[22:21] <JEEB> uhh
[22:21] <JEEB> it should just remove the packages that are configured and linked against the standard (without x264 etc.) libavcodec packages
[22:22] <JEEB> and then it should at the same time replace them with packages that are configured with x264 etc. enabled in the libavcodec
[22:22] <JEEB> you should look at what it says exactly
[22:22] <JEEB> because that is what it should do
[22:22] <JEEB> if it does something else then something's wrong
[22:22] <JEEB> but I do know that in the cases I've done it, it has worked rather nicely
[22:23] <vlt> JEEB: Yes, Ill have a look at the apt log files later and show them here in case youre interested.
[22:23] <JEEB> no, I'm not interested. I just want to tell you why it looks like it's going to "break" something
[22:23] Action: JEEB sighs
[22:27] <JEEB> vlt, although it might be apt failing to see the resolution, too
[22:28] <JEEB> http://askubuntu.com/questions/132164/ubuntu-restricted-extras-wants-to-remove-over-200-packages-could-this-harm-the
[22:28] <JEEB> looking at this
[22:28] <JEEB> if this is what happens then take a look at the answer
[22:30] <relaxed> there are static ffmpeg binaries available with libx264 support
[23:28] <Jiyuu|C2> no help for [22:26:47] <Jiyuu|C2> is it possible to use the libass subtitles filter so it will use an mkv for the source?
[23:28] <Jiyuu|C2> ?
[23:29] <burek> Jiyuu|C2, you could use -vf subtitles
[23:30] <Jiyuu|C2> would that work with a .mkv file as the source?
[23:31] <Jiyuu|C2> or do i still have to extract the subtitles from it?
[23:33] <burek> hmh, did you read the docs about that filter?
[23:33] <burek> it's kinda new, so i didnt have time to read it thoroughly
[23:38] <saste> burek: should be possible, since subtitles uses libavformat to read the sub stream from the file
[23:39] <burek> saste, do you have any example/sample cmd line to showcase how to tell it to ffmpeg?
[23:40] <saste> burek, can't be hader than subtitles=in.mkv
[23:41] <burek> oh right
[23:41] <burek> thanks
[00:00] --- Mon Feb  4 2013