[Ffmpeg-devel-irc] ffmpeg.log.20120326

[burek]

[03:54] <grumpy_> Hello I hope someone can help. I get the following:"ERROR: libmp3lame >= 3.98.3 not found" then :"If you think configure made a mistake, make sure you are using the latest version from Git.  If the latest version fails, report the problem to the
[03:54] <grumpy_> ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem." I have no idea how to get the "config.log" can anyone help?
[03:56] <grumpy_> I'm cutting and pasting from "ubuntuforums.org/showpost.php?p=11483157&postcount=1945"  "HOWTO: Install and use the latest FFmpeg and x264 "
[03:58] <grumpy_> I'm on no. 7: cd git clone --depth 1 git://source.ffmpeg.org/ffmpeg cd ffmpeg ./configure --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb \     --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx \     --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab make sudo checkinstall --pkgname=ffmpeg --pkgversion="5:$(./version.sh)" --backup=no \    --deldoc=yes --defa
[03:58] <grumpy_> ult hash x264 ffmpeg ffplay ffprobe" and I get this when I enter .config... This is my second reinstall of ubuntu I thought I messed it up the first time.
[04:24] <grumpy_> can anyone help?
[08:15] <linuxdev> Hi, i'm getting "[libx264 @ 0xa6ffc00] Error setting profile baseline" by using "-profile baseline", this used to work in a previous ffmpeg version. Does anyone know the new syntax?
[10:34] <ikrabbe> what does "DTS -1, next:-200000 st:0 invalid droping" on encoding of a stream means? Actually the audio stream is dropped? (pastbin follows)
[10:35] <ikrabbe> http://paste.pocoo.org/show/571376/
[10:36] <ikrabbe> the effect of this problem seems to be, that the audio stream #0.1 is dropped on output
[10:44] <ikrabbe> I can convert the mic.wav to mic.aac seperately
[10:44] <ikrabbe> but the error still happens on -i mic.aac -acodec copy
[10:57] <ikrabbe> hmm the problem seems the time measurement, as without shortest it works
[10:59] <ikrabbe> so:
[10:59] <ikrabbe> which time rules, without shortest? a- the first stream, b- the longest stream?
[11:01] <ikrabbe> it seems to be the longest then
[13:25] <andrewebdev> Does anyone know what the prefix is for specifying a custom libmp3lame path is? I'm currently combiling with --enable-libmp3lame ... but I dont have it installed globally.
[13:53] <ikrabbe> I would try with --extra-cflags
[13:53] <ikrabbe> I don't see any extra-ldflags in configure
[13:54] <ikrabbe> ah wait, there is an --extra-ldflags actually
[13:57] <ikrabbe> andrewebdev: so simply add --extra-cflags and --extra-ldflags should do the job and as far as I understand the configure script, these flags can be used multiple times without negative side effects
[14:21] <andrewebdev> ikrabbe, thanks, I actually found that out just a couple of mins ago :)
[15:29] <ikrabbe> luckily the ffmpeg configure is no autoconf script, so it's readable :)
[15:45] <orangey1> hello all!
[15:46] <orangey1> I'm trying to use ffmpeg (?avconv??) to resync a file. I would like to avoid recompressing the video / audio, though. How do I get it to just fix the audio / video location, and do nothing else?
[15:47] <ikrabbe> orangey1: I would first try with -acodec copy, -vcodec copy
[15:47] <orangey1> Currently, this is the command I'm trying, but it compresses the video and audio to something else: avconv -i in.avi -itsoffset -0.3 -i in.avi -map 1:0 -map 0:1 out.avi
[15:47] <orangey1> ikrabbe: thank you.
[15:47] <ikrabbe> no guarantees, I'm just a noob user too
[15:48] <orangey1> ikrabbe: Worked!!!!
[15:48] <ikrabbe> hmm, maybe I'm not that noobish anymore :D
[15:48] Action: orangey1 bows
[15:49] <orangey1> ooh. and wickedly fast.. 500 fps over my NAS
[15:49] <orangey1> for 1080p
[15:59] <orangey1> aaah
[15:59] <orangey1> this problem is going to have me pull out my hair.
[15:59] <orangey1> initially I thought it was 0.3 seconds off no matter what
[15:59] <orangey1> but it looks like the sync problem is variable
[16:10] <orangey1> ah. OK, so, the sync issue in the original file *is* 0.3 seconds *no matter what*
[16:10] <orangey1> however, once I render it with: avconv -i in.avi -itsoffset -0.3 -i in.avi -map 1:0 -map 0:1 -acodec copy -vcodec copy out.avi
[16:11] <orangey1> it becomes a *variable* sync problem
[16:11] <ikrabbe> orangey1: maybe avi is not the best format to solve sync problems
[16:11] <orangey1> ikrabbe: I thought avi was just a container?
[16:11] <orangey1> ikrabbe: what do you suggest?
[16:11] <orangey1> mp4?
[16:11] <ikrabbe> though that leads you to reencoding again
[16:12] <orangey1> ikrabbe: At this point, no problem if it'll work.
[16:12] <orangey1> I can also re-render
[16:12] <orangey1> which isn't a big deal.
[16:13] <ikrabbe> for sync problem it might be best to demux the audio and video into mp2 video and some audio format (mp3 or aac, or even wav), and then recompose it
[16:14] <orangey1> should I render as mpeg2 to begin with?
[16:15] <ikrabbe> As a result I would choose a h264/aac .mp4
[16:15] <orangey1> ah. that's what i'm rendering to right now..
[16:16] <ikrabbe> yes I would do mpeg2
[16:16] <orangey1> ikrabbe: OK, thank you
[16:16] <ikrabbe> (because that's what I generally get anyway from my tv card ;)
[16:17] <ikrabbe> you might need to set a high bitrate, to not loose quality
[16:25] <gues0x00> Hi there! :-)
[16:26] <gues0x00> How can I convert an .ogg stereo file to mono?
[16:26] <gues0x00> is ffmpeg the tool I should use for doing this?
[16:28] <teratorn> gues0x00: Audacity might have an easy thing for that
[16:28] <teratorn> gues0x00: but ffmpeg probably has a thing
[16:29] <ikrabbe> gues0x00: you can use -ac 1 for the output to set the number of channels in the output
[16:29] <ikrabbe> but I don't know what it does if the stereo differs much
[16:31] <ikrabbe> there are some downmixing options from 7.1/5.1 or else to stereo
[16:33] <ikrabbe> see ffmpeg manpage Mixing examples
[16:33] <gues0x00> -ac 1 seems to work, thanks ikrabbe :-)
[16:33] <ikrabbe> For example, if you want to down-mix from stereo to mono, but with
[16:33] <ikrabbe>        a bigger factor for the left channel:
[16:33] <ikrabbe> for more advanced fun
[17:01] <OneFix_Work> Is ffmpeg no longer being developed?  When I run a newer copy of ffmpeg, I am getting "This program is not developed anymore and is only provided for compatibility. Use avconv instead (see Changelog for the list of incompatible changes).
[17:01] <OneFix_Work> "
[17:02] <OneFix_Work> I'm assuming that this is caused by one of the libraries that it has been built with...
[17:03] <TimNich> OneFix_Work:  It is vey actively developed. However those responsible for the avconv fork would wish you to believe otherwise...
[17:03] <OneFix_Work> TimNich: So, how did that line get included in the Ubuntu version???
[17:03] <TimNich> because Ubuntu switched to the fork by default...
[17:15] <Gepmff> Hey guys, I've written some code that decodes video's, all the h264 files I throw at it are decoded fine, but I can't seem to decode mkv's or mp4's I get "damaged header" during the decoding process, anyone have an idea as to why?
[17:30] <vivienschilis> does aacplus supports sample rates < 44100 ?
[17:39] <Aegwyn11> Hi there, I have some feedback regarding Prores decoding and not sure where to send it. I have a Prores 422 file that has some periodic issues. The libavcodec decoder manifests the issues as grey or grey/green macroblocks, while the quicktime decoder does a VERY good job of concealing them to the point that most of them aren't visable at all. The errors are all shown as "[prores @ 02cb9b80] ac
[17:39] <Aegwyn11> tex damaged -3, 1024".
[19:27] <teratorn> I'm encoding a VP8 stream using libvpx - using codec default values - but it looks pretty bad. what should I do to (presets?) to set up a higher-quality encode?
[19:28] <Mavrik> teratorn, http://www.virag.si/2012/01/webm-web-video-encoding-tutorial-with-ffmpeg-0-9/
[19:28] <teratorn> Mavrik: thx
[19:30] <teratorn> Mavrik: I'm doing this in C code btw - do you know anything about setting up these parameters using the API?
[19:32] <ubitux> check doc/examples/*.c
[19:32] <Mavrik> teratorn, you'll have to checkout libvpx.c in avcodec folder
[19:32] <Mavrik> to see how those vpx-specific parameters are parsed
[19:34] <teratorn> OK, thanks
[19:37] <teratorn> is there not like a preset I can use?
[19:44] <_klk_> Hi all - I have a bunch of images that I would like to merge into a video file using ffmpeg.  I know how to do this, but is there a way to tell ffmpeg to repeat each image for N frames.  (e.g. 30 frames showing image1, 30 frames showing image2, etc.)
[19:46] <teratorn> _klk_: as a hack you could create your direction of images with symlikes for 29 of 30 images
[19:46] <teratorn> *directory
[19:53] <_klk_> teratorn:  i suppose i can try that, thanks
[19:53] <teratorn> well
[19:53] <teratorn> I supposed you can slow down playback by 30x
[19:53] <teratorn> and achieve the same effect
[19:56] <GreatEmerald> Is it possible to record audio and video with one instance of ffmpeg, but have it output two files (audio and video, separately)? The program segfaults every time I try to do that...
[19:58] <GreatEmerald> This is what I'm attempting to use: http://pastebin.com/raw.php?i=cERbgBda
[20:14] <juanmabc> hi, i get a weird ~1fps on a flv file on my own opengl/openal player based on ffmpeg, i use the pts correctly but i have a time_base issue, perhaps this could help: http://stackoverflow.com/questions/9854293/encoding-video-only-flv
[20:16] <juanmabc> also other issue is when i seek, it haves to wait for a non delta frame (but audio doesn't) so audio starts, and video keeps going with pts non drawable until scene gets moved to non delta frame
[20:38] <juanmabc> so i got pts and i do: if(pts * av_q2d(media.video.codec->time_base) * 1000000.0 <= av_gettime() - media->masterclock) update_texture
[20:46] <_klk_> teratorn:  i did the symlink approach.  ffmpeg barfs when trying to process the symlinks.  i give it an input wildcard like this:  ffmpeg -i image%03d.jpeg , but it says no files found.
[20:47] <_klk_> when i try an equivalent wildcard for non-symlinked files it works.
[21:16] <GreatEmerald> Aha, so my order of arguments was wrong, I have to set all input options first, and then all output options
[21:17] <GreatEmerald> Does ffmpeg make sure the audio and video that is being recorded is in sync?
[21:20] <GreatEmerald> Although either way, I guess that due to the fact that you can't specify individual input for each output file, there is no way to use ffmpeg to sync more than two streams and output them to different files to begin with...
[21:26] <GreatEmerald> Hmm, unless I can map the streams manually...
[21:28] <GreatEmerald> Yes I can map them! Awesome!
[22:03] <Kieran> Hi, I'm having a very strange problem. Apparently commas can't appear in the output filename unless a space has appeared somewhere earlier in the filename. E.g. "test,test.m4a" triggers a "test,test.m4a: No such file or directory", where as "te st,test.m4a" or "test ,test.m4a" works. "test, test.m4a" or "test,te st.m4a" don't work either. Using latest "64-bit (Static)" Windows build found at http://ffmpeg.zeranoe.com/builds/
[22:04] <Kieran> Is this a bug, or is there something I'm missing?
[22:04] <teratorn> _klk_: huh well that sux - I wonder why it doesn't like symlinks
[22:05] <teratorn> sounds like a bug
[22:05] <rav0__> do you have the quotes in your commandline? (i'm thinking about windows' handling of commas for multiple files)
[22:05] <Kieran> yes
[22:06] <Kieran> ffmpeg -i 218.m4v -i metadata.txt -y -c copy -map 0:0 -map_metadata 1 -map_metadata:s 1 -t 00:03:23.400 -ss 00:03:00.220 "test,test.m4a"
[22:06] <Kieran> triggers "test,test.m4a: No such file or directory"
[22:06] <Kieran> putting a space anywhere before the comma works fine
[22:11] <GreatEmerald> Oh, so -vsync and -async can sync the audio and video, that's even more awesome. Too bad both the arguments have very different meanings out of context :\
[22:11] <GreatEmerald> both names, rather (vertical sync and asynchronous)
[22:12] <GreatEmerald> So searching for "ffmpeg async" or "ffmpeg vsync" won't give any good results at all...
[22:16] <GreatEmerald> It says that the -async argument takes samples per second. Does that relate to the sampling rate?
[23:43] <teratorn> could anyone help me setting codec options? I'm using av_dict_set() to set the "rc_target_bitrate" option to "2048" before passing the options in to avcodec_open2(...) - and I've got DEBUG level logging on - when the codec options are dumped out rc_target_bitrate is still 256?
[23:49] <teratorn> OK, well the "b" option does seem to have an effect...
[00:00] --- Tue Mar 27 2012