[Ffmpeg-devel-irc] ffmpeg.log.20120317

burek burek021 at gmail.com
Sun Mar 18 02:05:01 CET 2012


[00:55] <akovia> Hi, I would like to convert some old avi files of different formats (mostly xvid) to mp4 without transcoding the video and audio to aac. The main purpose is being able to add meta information for cataloging and searching so I'm not really worried about transportability.
[00:59] <sacarasc> Was that you want to transcode the audio or not?
[01:00] <akovia> Yes
[01:02] <sacarasc> ffmpeg -i input -c:v copy -c:a aac -b:v 256k -metadata "title=blah" -metadata "episode=foo" output.mp4
[01:02] <sacarasc> Something like that...
[01:03] <akovia> ty very much, exactly what I was looking for
[01:08] <akovia> ffmpeg: unrecognized option '-c:v'
[01:08] <ubitux> -vcodec copy, or upgrade your ffmpeg
[01:09] <akovia> ok
[01:15] <akovia> are the commands of newer versions deprecated? I have some scripts I don't want to break.
[01:16] <ubitux> on ffmpeg, they are not considered deprecated
[01:16] <ubitux> they're just simpler
[02:48] <nanoflowe> does ffmpeg support ffmpeg -i input -c:v copy -c:a copy output.mov
[02:50] <sacarasc> ffmpeg -formats|grep mov # that should tell you. :D
[02:51] <sacarasc> But mine does, so yours should too in theory.
[02:52] <nanoflowe> what is better?  .mov or .mp4 container
[02:53] <sacarasc> I'd use Matroska almost every time.
[02:53] <nanoflowe> problem with mkv is that explorer does not support  extravideoinfo for mkv
[02:57] <ubitux> nanoflowe: mp4 & mov are mostly the same; mov is made by apple, and mp4 is the standardized version by MPEG
[02:57] <ubitux> what you can mux in them may differ
[02:57] <nanoflowe> so mp4 container is not by apple?
[02:57] <ubitux> mp4 has more support on some plateform though
[02:57] <ubitux> mp4 is based on the mov
[03:01] <nanoflowe> and i see lot of .flv too now
[03:01] <nanoflowe> is .flv better than mp4
[03:01] <ubitux> it's different, and pretty simple afaik
[03:02] <ubitux> you can't mux much things in it for instance
[03:02] Action: ubitux &
[10:18] <ura> Hi
[10:20] <ura> I have a 1920x1080 yuv420p stream. I want to encode it in 1024x576 with libx264 encoder. Is there any difference if I scale yuv stream first and then encode, or I scale it during encoding?
[10:21] <ura> e.g.  ffmpeg -s 1920x1080 -i 1920x1080.yuv -s 1024x576 1024x576.yuv ; ffmpeg -i 1024x576.yuv ...encoding options... out.264
[10:21] <ura> or ffmpeg -i 1920x1080 ...encoding options... out.264
[10:22] <ura> Do these commands produce the same quality?
[10:24] <ura> How ffmpeg encoding works? It scales frames first then encode, or it take advantage of higher resolution image to encode better in smaller resolution?
[10:27] <juanmabc> i would first encode then scale, as encoding is a lossy proccess
[10:29] <ura> here examples in more readable form: http://pastebin.com/hphKySSE
[10:30] <ura> juanmabc, but it's impossible to scale stream without reencoding
[10:31] <juanmabc> ffmpeg -yuvto264; ffmpeg -264(re)scale
[10:33] <ura> Where I can read about these options? They are not mentioned in ffmpeg man page and I'm unable to find in google (however ffmpeg accepts -yuvto264
[10:39] <juanmabc> ehm, there was a simplification of the process ;P
[10:40] <ura> ah, understood :)
[10:41] <ura> but scaling x264 -> x264 is lossy process (first it decodes, then it encodes again)
[10:41] <ura> so I think better to scale before or during encoding, and that's why I'm curious what is better to scale image before encoding or during encoding
[10:42] <Mavrik_> juanmabc, that'll just make you lose more quality
[10:42] <Mavrik_> ura, scale it during encoding
[10:43] <Mavrik_> ura, you san pass "-sws_flags lanczos" to use lanczos rescale filter which gives slightly better rescale quality
[10:43] <ura> so x264 codec takes benefit of higher resolution input to encode better?
[10:43] <ura> Malmis_, thanks for the answer, by the way :)
[10:44] <ura> oops Mavrik_ of course :)
[10:44] <Malmis_> ;P
[10:44] <Mavrik_> ura, you just don't introduce quality loss because of two reencodings
[10:44] <ura> I have raw video stream on input. I don't loose on reencoding when I scale it.
[10:45] <Mavrik_> nothing will change - raw images will get passed through the "scale" filter one way or another
[10:45] <Mavrik_> unless you can provide a better filter than sws_scale with lanczos you'll gain nothing~
[10:45] <ura> So my both examples http://pastebin.com/hphKySSE produce the same quality?
[10:46] <Mavrik_> yep
[10:46] <Mavrik_> first is just wasting more of your time :)
[10:46] <ura> Mavrik_, I just don't want to store big 1920x1080 stream on disk, so I want to scale it on the fly to save a disk space.
[10:47] <ura> I want to do two pass encoding, thats why I can't just feed 1920x1080 input, and on other hand I don't need such big resolution, so I want to scale it first :)
[10:48] <Mavrik_> as I said.
[10:48] <Mavrik_> both of those are doing exactly the same thing
[10:48] <Mavrik_> one just saves intermediate result to disk
[10:49] <ura> Thank you very much! I was doubt much about quality :) And thank you for suggesting scaling options, I was searching for such option but I probably missed it in manual.
[11:52] <juanmabc> two questions, is formatcontext->duration always valid? and is codeccontext->time_base variable for vbr?
[11:53] <juanmabc> also, the library itself must provide a valid presentation time stamp (even computing itself what ffplay does), fighting with it it's almost like performing a decoder job
[12:23] <juanmabc> ...
[15:32] <MrHanjrah> hello, i want to convert .mkv file to avi, what should i put in -vcodec flag? xvid, divx ?
[15:32] <rav0> you have to choose. xvid sounds good. or mpeg4 if you like
[15:35] <MrHanjrah> rav0: thanks, so say for using xvid, what should be the argument for -vcodec flag? libxvid or just xvid
[15:35] <rav0> should be the same thing. i use pre-compiled versions where things are named one way or the other and i don't find out why
[15:36] <MrHanjrah> hmm, alright
[15:39] <MrHanjrah> thanks rav0 ,its working, i used libxvid
[16:44] <Goffredo> what's the status of aac encoding? why is it turned off by default?
[16:46] <Mavrik_> because the encoder doesn't really work well, use libfaac or libvo_aacenc
[16:48] <Goffredo> which are not compiled by default (at least the one in the arch repo)
[16:48] <Goffredo> right?
[16:52] <sacarasc> Correct.
[16:54] <Goffredo> what about windows? is the prebuilt binary compiled with aac support?
[16:55] <Goffredo> where can i find all of this information? I'm having trouble finding good information
[16:55] <sacarasc> Run ffmpeg and look through the configure options.
[16:56] <Goffredo> I don't have windows right now, I only wanted to know, if the majority of people who happen to have ffmpeg ona windows machine, would be able to encode aac out of the box
[16:57] <Goffredo> (since aac is needed to encode mp4 for ipods as far as I'm aware)
[16:58] <sweb1> how can i install last stable version of ffmpeg on RH family, cent os
[16:58] <sweb1> any repository for this ?
[17:18] <sweb1> any one here ?
[18:19] <sweb1> i have a problem during compile on Centos as ffmpeg wiki
[18:19] <sweb1> http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide
[18:20] <sweb1> config.log : http://pastie.org/3616056
[18:27] <sweb1> morkia: did you can compile opencv with ffmpeg ?
[18:27] <morkia> sweb1: not yet
[18:28] <morkia> sweb1: i have to apply the patch provided by opencv community but it doesn't work may be it's not written for this distribution im using i.e ubuntu
[18:30] <sweb1> sweb1: source files is'nt depending to dist. i have situation here . cant compile ffmpeg in centos.
[18:30] <morkia> i have to change the line: CODEC_TYPE_VIDEO to AVMEDIA_TYPE_VIDEO but when i look up into cap_ffmpeg.cpp i don't see it there
[18:31] <morkia> neither the patch nor manual change is possible
[18:31] <morkia> sweb1: did you try ./configure?
[18:32] <sweb1> morkia: `./configure` is first part of each compile dude
[18:32] <morkia> sweb1: yes i  know but you might have to disable few things before compiling
[18:33] <sweb1> this is the guide for CentOS http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide
[18:33] <sweb1> morkia: ^
[18:33] <morkia> sweb1: what's the error you're getting?
[18:34] <sweb1> morkia: http://pastie.org/3616122 problem is for lamemp3
[18:34] <sweb1> shared and internal method not working
[18:36] <morkia> sweb1: have you tried: ./configure --enable-shared
[18:36] <sweb1> sweb1: yeah
[18:38] <morkia> sweb1: you might need to install some prerequisites for ffmpeg try doing this: sudo apt-get remove ffmpeg x264 libx264-dev
[18:38] <morkia> sudo apt-get install git-core checkinstall yasm texi2html \
[18:38] <morkia>  libfaac-dev libjack-jackd2-dev libmp3lame-dev libopencore-amrnb-dev \
[18:38] <morkia>  libopencore-amrwb-dev libsdl1.2-dev libtheora-dev libva-dev libvdpau-dev \
[18:38] <morkia>  libvorbis-dev libx11-dev libxfixes-dev libxvidcore-dev zlib1g-dev
[18:39] <sweb1>  sweb1: i'm using CentOS
[18:39] <sweb1> RH havnt packages like Debian based dists
[18:40] <morkia> sweb1: oops my bad
[18:40] <morkia> sweb1: totally forgot
[18:41] <sweb1> sweb1: debian most useful than RH based dist
[18:44] <morkia> any here facing the same problem?
[18:44] <morkia> opencv not compiling
[18:54] <morkia> sweb1: you there?
[18:57] <sweb1> morkia: yeah
[19:01] <greenythebeast> Question for all you smart people. I'm trying to use FFmpeg to convert some nellymoser audio to a .wav file. However, when the .wav file is produced it is a different length than the nellymoser audio. What would cause this?
[19:01] <greenythebeast> I've pasted the terminal output to http://pastebin.com/tqm1faZ7
[19:02] <morkia> sweb1: please come on google talk
[19:51] <DrSlony> Hey, can someone please update this? http://ffmpeg.org/ffserver.html#The-audio-and-video-lose-sync-after-a-while_002e
[19:52] <DrSlony> which versions does it apply to, does it always happen and is there a fix or at least a workaround?
[20:38] <Goffredo> hi, I'm trying to compile ffmpeg on arch. I get this error: ERROR: libpulse-simple not found even though I have libpulse installed, what could be the problem?
[21:49] <akovia> Hi, I am trying to convert some older avi files to an mp4 container with aac audio and am getting some errors. I haven't used pastebin b4 so I hope this is right. http://pastebin.com/s3w8EaFb
[22:48] <IamTrying> Latest ffmpeg from git, is not compiling any idea what is wrong? e.g: https://gist.github.com/2065586
[22:55] <akovia> i built from git this morning. updated since then i wonder?
[22:56] <akovia> ffmpeg version git-2012-03-17-d9f26a0 Copyright (c) 2000-2012 the FFmpeg developers   built on Mar 17 2012 08:48:05 with gcc 4.4.3
[23:02] <IamTrying> akovia, i keep getting this make: *** [libavutil/libavutil.so.51] Error 1
[00:00] --- Sun Mar 18 2012


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