[Ffmpeg-devel-irc] ffmpeg.log.20121215

[burek]

[01:57] <klaxa> is there a way to tell ffmpeg h264 raw stream codec parameters? i have a file split at IDR frames and i want to mux them to an mp4 or whatever, if i use the file concatted i can mux it into mp4 (with a lot of "pts has no value" errors)
[01:57] <klaxa> the file plays fine though
[01:57] <klaxa> however, if i try to mux the first block ffmpeg fails to recognize codec parameters, increasing probesize and analyzeduration doesn't help
[01:58] <klaxa> i know stuff like resolution, framerate and profile though, can i tell ffmpeg those parameters?
[02:22] <an3k> klaxa: mp4box
[02:22] <an3k> ffmpeg is for encoding, mp4box is for muxing
[02:23] <klaxa> where do i get... sourcecode or linux builds?
[02:23] <klaxa> first google page doesn't seem to turn it up
[02:24] <an3k> http://gpac.wp.mines-telecom.fr/downloads/
[02:24] <klaxa> ah nice, thx
[02:27] <an3k> to mux: mp4box -add "file.264#trackID=1:fps=23.976" -add "file.aac#trackID=1:lang=eng" -ipod -tmp "MUST BE THE SAME DIRECTORY AS INPUT STREAMS" -new "MyMuxedFile.m4v"
[02:29] <klaxa> what does -ipod do?
[02:30] <an3k> it creates a file using CodecID: M4V
[02:31] <klaxa> i see
[03:03] <Nolaan> Those are my configuration switches : http://fpaste.org/HxWf/
[03:03] <Nolaan> thank you for helping
[04:32] <CoveGeek> Is this a place to discuss f4fpackager?  I can't find it.
[04:40] <roboman2444> hey hey hey, is there a way i can record alsa output with ffmpeg?
[04:40] <roboman2444> i have googled and googled and tried almost everything
[04:51] <seablade_> Quick request for help, trying to create a slideshow that is made up of images, each lasting 10 seconds.  When converting the images to 10 seconds of video I have some nasty compression artifacts around text on colored backgrounds etc.  Was wondering if anyone can help me with a better CLI string than this...
[04:52] <seablade_> http://pastebin.com/VceER5V7
[04:52] <seablade_> ${i} is the file name of the JPEG file
[04:55] <klaxa> try out a different preset
[04:56] <klaxa> and maybe update your ffmpeg, since you can use -vpre hq it seems outdated
[04:58] <seablade_> klaxa, Yea updating ffmpeg is a bit difficult at the moment sadly, but if that is the only option I will have to start working on compiling it
[04:59] <seablade_> klaxa, I tried a couple of presets, thus far no good luck, any suggestions?
[05:02] <seablade_> hmm lossless_max is definitely better, but still some nasty compresion artifacts on certain colors it seems
[05:02] <seablade_> But at least it is usable for the time being
[05:53] <roboman2444> seablade_, the lossy encoding can create baddies for lossy
[05:53] <roboman2444> err
[05:53] <roboman2444> text
[05:54] <roboman2444> and i would suggest keeping 1 frame every ten seconds to keep the filesize down if you are going lossless
[05:55] <klaxa> and probably remove audio stream at all since you are using a silent track anyways?
[05:56] <klaxa> roboman2444, as to recording alsa, dunno, if you have pulseaudio on top though (adds lulz 9001 overhead points) you can use output monitors
[06:00] <roboman2444> klaxa, i dont use pulse, its too annoyinh
[06:00] <roboman2444> klaxa, i dont use pulse, its too annoying
[06:01] <klaxa> well i use it because i find it convenient for quite a lot of things, still adds a lot of overhead though
[06:01] <klaxa> and i can totally understand why people don't use it
[06:01] <klaxa> and i don't want to tell anyone to use it, however, for this purpose it would make the solution easier
[06:02] <roboman2444> i know... i think there is a way to do it in alsa though
[06:02] <klaxa> if so, please cc the solution to me in a private message :)
[06:19] <lake> hi all, i'm interested in cropping the top and bottom of my video by 10 pixels.
[06:19] <lake> so 20 pixels total.
[06:20] <lake> my input is 640x480, i would like to crop 10 pixels from the top and ten pixels from the bottom, and then scale it back up to 640x480.
[06:21] <lake> I'm playing around in mplayer: "mplayer in.avi -vf crop= crop=640:480:???,scale=640:480"
[06:21] <lake> i need some help with the ??? marks
[06:22] <lake> i've also tried reading about how to crop online but it confuses me a lot
[06:31] <seablade_> roboman2444, klaxa I was trying to keep framerate down, but couldn't get it working right, any framerate runder 1fps just didn't encode here when I tried that.  Audio is there because the intent is to put these files in front of and behind a video file, but I am having problems on that aspect of things as well
[06:40] <roboman2444> seablade_, why dont you just gif the thing?
[06:41] <seablade_> roboman2444, Can't gif the video file
[06:41] <roboman2444> no i mean gif the images
[06:42] <seablade_> roboman2444, I am looking to create a single video file, at the beginning and end of this video file would be the slideshow of X number of slides(Usually less than 10) with a video file in the middle.  This file would be looped for digital signage
[06:42] <seablade_> roboman2444, The images are provided to me:)
[06:42] <seablade_> roboman2444, I don't create them
[06:42] <roboman2444> ah
[07:17] <alucinor> So I am having a very strange encoding issue... I am encoding x264 video files
[07:18] <alucinor> ffmpeg does its job with no errors, ffplay can play it just fine.. BUT when I try to play it outside of ffmpeg, such as with xine or gstreamer, a couple minutes into the video the video completely blows up, it looks like it is fast forwarding very very quickly, the audio stays normal though
[07:19] <zap0> try adjusting a variable, and re-encoding
[07:20] <alucinor> zap0; I leave almost everything to defaults, any suggestions for variables to change?
[07:22] <zap0> bit rate
[07:22] <zap0> even by a tiny amount;  might result in a very different encoding
[07:23] <alucinor> hmm ok
[07:29] <alucinor> zap0; changing bit rate multiple times with different settings did absolutely nothing, it still freezes at the same exact time
[07:30] <zap0> ffmpeg version?
[07:30] <roboman2444> mplayer?
[07:30] <zap0> is the freeze point some magic number?    like exaclty 1M. or PI or something?
[07:31] <zap0> does ffprobe give you all the details your expect?
[07:31] <zap0> are all those details correct?
[07:31] <alucinor> zap0; no it is random between different files, usually somewhere in the first 2 minutes
[07:32] <alucinor> ffmpeg version 0.10.3, built dec 2 2012
[07:32] <zap0> tried a nightly build version?
[07:32] <zap0> or an older version?
[07:32] <alucinor> nope have not tried newer versions
[07:33] <alucinor> I'll give that a go
[07:33] <zap0> also as R said above,  mplayer!??!
[07:34] <zap0> after its encoded, can you decode it into another format or individual frames
[07:35] <alucinor> mplayer plays the video just fine, with no corruption
[07:35] <alucinor> and yes, I can decode it just fine
[07:36] <alucinor> into other formats that is, with ffmpeg
[07:38] <alucinor> the latest ebuild version of ffmpeg for gentoo is 0.10.6 I will try that
[07:40] <alucinor> nope, same bug in that version
[07:45] <alucinor> is there something in the x264 codec that causes framerate to speedup or slow down?
[07:45] <lake> can anyone explain this?  Unable to parse option value "-1" as pixel format
[07:45] <alucinor> the reason I ask is the files that do not work, if I play them in xine, the time when it fails in the others it goes VERY fast in xine and it complains about the frame rate being too high
[07:46] <alucinor> so it seems like the frame rate is all of a sudden multiplied by a very large number and blows up
[07:46] <zap0> does ffprobe give you all the details you expect?
[07:47] <alucinor> yes
[07:47] <zap0> does $MEDIA_PLAYER show the same values in its 'about media' dialogs?
[07:48] <alucinor> what is that?
[07:49] <alucinor> $MEDIA_PLAYER is blank
[07:50] <zap0> i mean  "insert your favourite media player app's name here"
[07:50] <zap0> does xine show the same values in its 'media info' dialogs as ffprobe does.
[07:50] <zap0> like fps,size,bitrate,codec
[07:51] <alucinor> let me check
[07:53] <alucinor> yep all the same
[07:54] <alucinor> well I can describe the problem in detail now with xine
[07:55] <alucinor> when the error occurs... the speed of the video increases by 10 fold, going 10 times too fast like it is fast fowarding, meanwhile the audio is going normal speed
[07:55] <alucinor> dunno if that helps
[09:14] <alucinor> fixed my issue... apparently the bitrate information was corrupted from the origional file I was encoding from... using -r 28 fixed my problem
[12:08] <divVerent> I need some help using ffmpeg... I use this command to convert audio files to mp3:
[12:09] <divVerent>         ffmpeg -nostdin -vn -sn -i "$1" -f mp3 -codec:a libmp3lame -compression_level:a 2 -qscale:a 1 -map_metadata:g -1 -map_metadata:s -1 -map_metadata:c -1 -map_metadata:p -1 "$2"
[12:09] <divVerent> but it retains album art. I want it stripped out. How would I do that?
[12:15] <Nolaan> hi!
[12:15] <Nolaan> i'm trying to cross compile ffmpeg for arm
[12:15] <Nolaan> but it cannot find libpulse-simple?
[12:15] <Nolaan> how to do?
[12:16] <Nolaan> Those are my configuration switches : http://fpaste.org/1Sxp/
[12:16] <cbreak> how about disabling libpulse?
[12:16] <Nolaan> i need it for the sound?!
[12:17] <cbreak> why?
[12:17] <Nolaan> because i need to record both camera and sound
[12:17] <Nolaan> i wanna stream those
[12:21] <cbreak> and you want to use pulse for input?
[12:21] <cbreak> then you'll have to get the libraries for it
[12:41] <Nolaan> cbreak: I clearly understood that, can you more specific please?
[12:41] <cbreak> I don't know the specifics for that.
[14:40] <hiptobecubic> is there a no-op set of parameters for ffmpeg?
[14:40] <hiptobecubic> so that it essentially just copies the file
[14:40] <sacarasc> -c:a copy -c:v copy -c:s copy
[14:42] <hiptobecubic> hmm
[14:44] <hiptobecubic> I am having trouble with this file. It seems like it's borked somehow. I'm not sure how to explain but the file is available here: http://www.opengamma.com/downloads/A0E76160-46B8-11E2-95FF-7FB6DF4B3F91/Algorithmic-Differentiation-in-Finance-Webinar-12-12-2012.mov
[14:44] <hiptobecubic> the first 30 seconds or so are "empty" somehow
[14:44] <hiptobecubic> it doesn't render properly
[14:44] <hiptobecubic> but the real content doesn't start until 2:30 anyway so i am trying to trim it
[14:50] <hiptobecubic> ffmpeg -i somefile.mov -ss 00:02:30 -c copy output.mov   only seems to trim the video, the audio still starts from the beginning
[14:54] <hiptobecubic> is -ss supposed to shift both audio *and* video?
[14:55] <sacarasc> Yes.
[14:55] <hiptobecubic> well i wonder what i'm doing wrong then
[14:55] <hiptobecubic> ffmpeg -i Algorithmic-Differentiation-in-Finance-Webinar-12-12-2012.mov -c copy -ss 00:02:30 o.mov
[14:55] <hiptobecubic> should produce a copy of Algo....mov  with the first 2 and half minutes cut off, right?
[14:56] <hiptobecubic> not with
[14:57] <hiptobecubic> not with the audio shifted back 2:30 and the video starting from 0:00 *
[14:57] <hiptobecubic> when i look at the info for the two files though. they also differ with respect the "streams", so maybe that's related
[14:58] <hiptobecubic> http://vpaste.net/Emg8u
[14:59] <hiptobecubic> where 'o.mov' was produced via the command i gave earlier
[15:02] <cbsrobot> hiptobecubic: try
[15:02] <cbsrobot> ffmpeg -ss 00:02:30 -i Algorithmic-Differentiation-in-Finance-Webinar-12-12-2012.mov -c copy -t 10 test.mp4
[15:02] <cbsrobot> see if it works
[15:02] <cbsrobot> then remove the -t 10
[15:02] <hiptobecubic> sure
[15:03] <hiptobecubic> it's not "right", but i think it's a problem with the video file
[15:04] <cbsrobot> what is not right
[15:04] <cbsrobot> ?
[15:05] <hiptobecubic> i think somehow there are only a few frames in the beginning
[15:05] <hiptobecubic> because what happens is it will draw the initial frame, then not update for 30 seconds, and if you do anything like drag the window around etc, it will leave trailing lines everywhere ala windows 95 solitaire
[15:05] <hiptobecubic> but the time counter is increasing normally and eventually it will get to the 'good' part of the video
[15:05] <hiptobecubic> but for example, when i did your command, there was no audio in the beginning and i *know* that there is audio at 2:30 in the video
[15:05] <hiptobecubic> the original
[15:06] <hiptobecubic> I'm not great at explaining this because honestly, i don't know balls about how these various container formats and audio video streams work
[15:08] <cbsrobot> maybe its because the movie is at 2.48 fps and the normaly you can only cut at I frames
[15:08] <hiptobecubic> what's an I frame?
[15:10] <hiptobecubic> cbsrobot, also, there are parts of this where he's moving the mouse around to draw attention to some part of the slides and it is definitely better than 2.5 fps
[15:10] <cbsrobot> hiptobecubic: sorry I have to go
[15:11] <cbsrobot> maybe later
[15:11] <hiptobecubic> cbsrobot, cheers. thanks for the effort
[15:25] <hiptobecubic> it's a shame this video is so screwy.
[15:26] <hiptobecubic> is there a preset i can use to just decode and reencode to try to fix it up?
[15:26] <hiptobecubic> as in from any input to 640x480 h264 output at 23fps or something
[15:27] <hiptobecubic> oh i see... \topic has -preset
[16:11] <lake> Given i have an input video (640x480) and I want to crop out the top border by 10px and the bottom border by 10px, how do i do that?
[16:14] <saste> lake, crop filter
[16:32] <lake> saste: i'm trying this "ffmpeg -vf crop=720:480:60:40 -i clip-christmas-04-001.mkv -vcodec copy -acodec copy out.mkv" but it's not working at all
[16:33] <saste> lake: copy and filtering are conflicting
[16:33] <saste> you can't copy data and filter it at the same time
[16:38] <lake> saste: okay, so if i don't specify vcodec it works. that's great. I just need to figure out how to crop the top by 10px and bottom by 10px
[16:38] <saste> lake: crop=iw-20:ih-20:10:10
[16:39] <saste> lake: crop=iw:ih-20:0:10
[16:40] <lake> saste: wow!
[16:40] <lake> saste: thank you for that! Will i need to scale it back up to 720x480 for dvd compliance?
[16:49] <saste> lake, dunno
[17:16] <lake> i have a lot of motion in my videos and i deinterlace with yadif=0. I notice that sometimes the motion appears very "wavy" or "watery".
[17:16] <lake> it just doesn't look right. Am I doing something wrong?
[17:43] <lake> aaaah, looks like i needed to deinterlace before any other video filters
[17:43] <mazda01> trying to stream to twitch using ffmpeg. using the following line for audio, -acodec libmp3lame -ar 44100 -threads 4 -qscale 3 -b 712000  -bufsize 512k \  does that mean the bitrate of the audio is 750kbps?
[17:45] <mazda01> i only have .44 mbps upload from my ISP
[17:47] <mazda01> my video line is this: -vcodec libx264 -s $OUTRES -crf $CRF -maxrate 350k -preset $QUAL \ with outres being 426x240, CRF being 28, and quality being veryfast
[17:47] <mazda01> with a FPS of 30.
[17:49] <mazda01> can anyone help please? does that audio setting mean 750kbps?
[17:54] <klaxa> no. -b is different from -ab
[17:55] <lake> when i convert my video from mkv to mpg (via -target ntsc-dvd), I get shit tons of buffern underflows and choppy video output
[17:57] <mazda01> klaxa, what does the -b for audio do?
[17:58] <klaxa> in this case i think it is bitrate though... i can't find -b in the manpage, just remove -b 712000 and replace it by -ab 96k or something similar
[17:59] <mazda01> klaxa, if my upload is only 440Kbps and I am already using 350Kbps for video, how do I set the audio to be like 50Kbps?
[18:00] <klaxa> is your upload 440 kbps or kBps ?
[18:00] <klaxa> so kilobits or kilobytes?
[18:00] <klaxa> anywho to set audio to 50 kbps you add -ab 50k
[18:00] <mazda01> klaxa, it's .44 megabits per second
[18:01] <mazda01> klaxa, internet transmission is measured in bites, not bytes correct? That's what speedtest.net shows
[18:01] <klaxa> ah yes in that case, yeah
[18:04] <mazda01> so between the video 350k max and audio 50k max, that would be 400k of data uploaded out of the 440 I have to use
[18:05] <mazda01> klaxa, would it be -b 50000? or -b 5000
[18:05] <klaxa> -ab 50k
[18:06] <mazda01> klaxa, the original line is -acodec libmp3lame -ar 44100 -threads 4 -qscale 3 -b 75000  -bufsize 512k \
[18:06] <mazda01> klaxa, i should be using -ab instead?
[18:07] <klaxa> yes or if you have a recent version of ffmpeg -b:a 50k would work too
[18:07] <mazda01> klaxa, using this ffmpeg version 0.8.4-4:0.8.4-0ubuntu0.12.04.1
[18:08] <mazda01> klaxa, and it actually says it deprecated and to use avconv
[18:08] <klaxa> that version is quite old
[18:08] <klaxa> for support on avconv you are in the wrong channel ;)
[18:09] <mazda01> klaxa, oh, lol
[18:12] <mazda01> klaxa, says it was built on Nov  6 2012
[18:13] <klaxa> that doesn't mean the sourecode was recent, try it with -ab 50k and see if it works, if it does, great, if it doesn't... well we'll see what to do in that case
[18:14] <mazda01> klaxa, i did man avconv and learned that if I use -maxrate for video then I need to use -bufsize. how do I know what -bufsize to set?
[18:15] <klaxa> hmm no idea myself
[18:16] <klaxa> still, for help on avconv visit #libav
[18:16] <JEEB> bufsize is dependant on what you are doing
[18:16] <mazda01> klaxa, do you know why avconv forked off of ffmpeg?
[18:16] <mazda01> JEEB, i am streaming to twitch.tv
[18:16] <JEEB> various reasons, both sides are developing and ffmpeg merges most changes from libav
[18:16] <klaxa> http://stackoverflow.com/questions/9477115/who-can-tell-me-the-difference-and-relation-between-ffmpeg-libav-and-avconv/9477756#9477756
[18:17] <JEEB> mazda01, then twitch.tv should tell you what to set maxrate and bufsize to
[18:17] <JEEB> as they are the limiting factor
[18:17] <JEEB> the buffer is sometimes specified in time
[18:17] <JEEB> then it is maxrate * <amount of seconds>
[18:17] <JEEB> (of buffer)
[18:18] <mazda01> JEEB, it's pretty much dependent on my upload from my ISP I would think?
[18:18] <mazda01> JEEB, does buffsize using RAM?
[18:19] <JEEB> I'm pretty sure all of those streaming services that restream your stream have bandwidth limits :P
[18:19] <JEEB> but naturally you can't stream more than your internet connection can push
[18:19] <JEEB> also no, not RAM
[18:19] <mazda01> JEEB, right, but i wouldn't come close to limits with only .44Mbps upload
[18:20] <mazda01> JEEB, i am merely trying to get the best settings to stream with my limited upload from my ISP so that the stream isn't laggy etc etc
[18:20] <JEEB> well, see the bufsize definition over at twitch in any case :P
[18:20] <JEEB> then if it is time-based you set the maxrate then according to your connection speed
[18:21] <JEEB> or if it is set as a specific size you get a bigger bufsize for a smaller maxrate :3
[18:21] <mazda01> JEEB, ok, thanks.
[18:21] <JEEB> anyways, you can think of maxrate as the "maximum average bit rate over buffer", and then bufsize sets that buffer
[18:21] <mazda01> JEEB, suggestions I just read said to set the bufsize to match my max video bitrate which is 350k
[18:22] <JEEB> that means that you'd have a buffer of ~1 second
[18:22] <JEEB> thus, as long as the client and the server use the same bufsize and the maxrate is low enough for transfer to be flawless, no extra buffering should be needed during playback :)
[18:23] <mazda01> JEEB, ok, thanks
[19:35] <roboman2444> can i make ffmpeg loop through a bunch of mp3's for input audio?
[19:39] <roboman2444> basically add background music to a long video, without having to manually combine the mp3s
[20:13] <ubitux> roboman2444: yes, look at amerge (and maybe pan) and amix filters, i don't remember exactly which is more appropriate
[21:07] <tasslehoff> I have some huge icod .mov files from iMovie (e.g. 2.8GB for a 6 minute clip). Can ffmpeg help me transcoding this to a much smaller size with a usable format?
[21:10] <barque> does libavcodec handle multi-threaded decoding internally?
[21:11] <barque> or do I have to do use certain functionality?
[21:14] <Rathann> barque: yes it does
[22:24] <lake> whaat is the cause of a buffer underflow when converting an mkv file to mpeg (ntsc-dvd)
[22:46] <lake> klaxa: http://pastebin.com/WbTrBWin
[22:47] <wawrek> Hi, can we get the duraiton of a movie with ffmpeg? Is there a command which returns the length/duration of the clip?
[22:48] <klaxa> wawrek: ffprobe does that
[22:48] <klaxa> lake: sorry, no idea :S
[22:49] <lake> klaxa: all good. i'll hack it out
[22:49] <wawrek> klaxa: thanks, i will check out. but does it also work with avconv?
[22:49] <klaxa> visit #libav for that
[22:50] <klaxa> no idea actually
[22:50] <wawrek> thanks
[22:52] <ubitux> ffprobe is more developped than the equivalent in the fork
[22:52] <ubitux> it has several more writers
[22:52] <ubitux> such as csv, xml, flat&
[22:55] <wawrek> I found avprobe, I guess it's the same as ffprobe
[23:17] <wawrek> but still with ffprobe I have to process the output programatically and find the duration. I cannot find a single command to return the length.
[23:18] <saste> wawrek, correct
[23:19] <saste> there is no reliable way to detect duration, if not by decoding the file
[23:19] <saste> i may add an option to compute the duration in the code, but that will always require decoding
[23:20] <wawrek> saste: thanks for the help, it sounds clear
[23:21] <saste> wawrek, no that's not correct
[23:21] <saste> it would require at least *demuxing* (show_packets)
[23:22] <saste> since the timestamp information is contained in the packets
[23:22] <saste> so discard my comment about decoding
[23:25] <iive> if the container contains timestamps you are ok with just demuxing. probably seeing the first timestamp then seeling near the end and demuxing the packets to the end would give faster result. ideally it could use the index for that. (if there is index).
[23:25] <iive> if the format is raw, then it may need at least parsing of the demuxed packet.
[23:26] <iive> packets.
[00:00] --- Sun Dec 16 2012