[Ffmpeg-devel-irc] ffmpeg.log.20120408

burek burek021 at gmail.com
Mon Apr 9 02:05:02 CEST 2012


[01:05] <dadada> hey
[01:05] <dadada> I'm batch converting a bunch of wma
[01:06] <burek> cool :)
[01:06] <dadada> I saw some examples where people use aq 100
[01:06] <dadada> is that necessary, would the quality by default not be 100%?
[01:06] <burek> I think each encoder has its own default settings
[01:06] <dadada> converting to mp3 that is
[01:07] <burek> so you are best off specifying what you want
[01:07] <relaxed> -aq 0 would be highest quality with libmp3lame
[01:07] <dadada> what is the default?
[01:07] <relaxed> probably 128k cbr
[01:08] <relaxed> why do you care what the default is?
[01:08] <dadada> is ogg quality better then mp3 or anything else I should look into?
[01:08] <burek> aac+ ! :)
[01:08] <relaxed> you mean vorbis, yes
[01:09] <dadada> just to know if I can trust the defaults
[01:09] <relaxed> you can not
[01:09] <dadada> k
[01:10] <dadada> why is vorbis good versus mp3?
[01:10] <burek> why is blue better than green?
[01:11] <relaxed> It's much newer for one, google for the rest.
[01:12] <relaxed> If you're so worried about quality use flac
[01:13] <dadada> flac would probably inflate the size too much
[01:14] <burek> try aac+ if you really want to keep the best size/quality ratio
[01:14] <Kuukunen> burek: blue is better because in double blind tests conducted by the University of Wisconsin, it got 7 points better average than green
[01:14] <burek> Kuukunen :)
[01:14] <dadada> lol
[01:23] <diegoviola> hi
[01:23] <diegoviola> can i record a single window with ffmpeg?
[01:23] <diegoviola> a specific region of my desktop
[01:44] <burek> diegoviola, yes
[01:46] <diegoviola> i see, thanks
[01:47] <diegoviola> any pointers please?
[01:48] <burek> what you need is x11grab
[01:48] <diegoviola> ty
[01:48] <burek> if you are on windows, then vfwcap
[01:48] <burek> http://ffmpeg.gusari.org/viewtopic.php?f=25&t=13
[01:48] <diegoviola> i'm on linux, so x11grab, thanks
[01:48] <burek> oh ok
[01:49] <diegoviola> thanks for the help, appreciated
[01:49] <diegoviola> :)
[01:49] <burek> :beer: :)
[01:49] <diegoviola> are x11grab and vfwcap something like ffmpeg modules?
[01:50] <diegoviola> can something like that be written for wayland?
[01:50] <burek> hmh.. you can say something like that, yes :)
[01:50] <diegoviola> or different windowing systems, etc
[01:50] <diegoviola> cool
[01:50] <diegoviola> thanks
[01:50] <burek> yw :)
[02:12] <teresko> hi everyone
[02:13] <teresko> i have two questions regarding http://ffmpeg.zeranoe.com/builds/
[02:13] <dadada> why does it say that vorbis is experimental and I need strict experimental?
[02:14] <teresko> 1. what is the difference between shared and static builds ( i have only encountered such distinction when dealing with opera on bsd with QT libs )
[02:14] <burek> dadada, because we want to make sure that you don't come back here complaining too much, because the codec is still experimental :D
[02:15] <burek> meaning _very_ beta :)
[02:15] <burek> teresko, shared = uses windows dlls, static = doesn't use any external dlls (has got everything it needs in the exe file)
[02:15] <teresko> damn .. nevermind
[02:15] <teresko> its written in the page
[02:15] <teresko> =/
[02:15] <burek> :)
[02:17] <dadada> burek: that's a bummer, then I can forget ogg and use mp3
[02:17] <burek> use aac+ :)
[02:17] <burek> it rocks!
[02:17] <teresko> and the other question was as stupid as first one : can i used the external codecs from ffmpeg's trunk with mplayer ?
[02:17] <dadada> burek: I doubt my router has support for that format :-)
[02:17] <burek> teresko, ffmpeg's "external codecs" are just simple independent libraries
[02:18] <burek> written just for that, so the other projects can include them in their own code :)
[02:18] <teresko> so the answer is "yes"
[02:18] <llrcombs> relaxed: OK, I reencoded the AAC MKA as an AC3 MKA. No issues with the duration, but it still plays very choppily in VLC
[02:18] <burek> so the answer is, obviously, "yes, you can!" :D
[02:18] <teresko> good
[02:18] <teresko> my mplayer has been outdated for years
[02:18] <burek> dadada, google for your router's user manual and check it out :)
[02:18] <teresko> ( and yes , i am using mplayer on windows )
[02:19] <burek> llrcombs, try converting your audio to wav and play it in the vlc, to see if the stream is ok?
[02:19] <llrcombs> burek: it plays fine in any other player
[02:19] <burek> llrcombs, what are the errors in vlc?
[02:20] <llrcombs> buffer too early, down-sampling
[02:21] <llrcombs> buffer way too early, clearing queue
[02:21] <burek> try remuxing with -vsync or -async
[02:22] <dadada> burek: hmm, the file I converted to vorbis plays fine on my desktop but very badly on the router... madplay works only for mpeg and works great on the router... I have to play ogg with sox on the router but that's horrible, maybe ogg needs more hardware resources, sox has bad perfomance, or encoding problem?
[02:22] <llrcombs> Is there a way to do that in MMG?
[02:23] <burek> dadada, did you try using aac? -f aac
[02:23] <burek> -acodec aac (sorry)
[02:23] <dadada> burek: I don't know that there's a single tool that play that on the router
[02:23] <llrcombs> burek: was that for me?
[02:23] <burek> llrcombs, what is mmg?
[02:24] <dadada> burek: it's a very limited env, mips cpu, busybox, totally stripped down, 400mhz
[02:24] <llrcombs> MkvMerge GUI
[02:24] <burek> dadada, which tool do you use to play it on the router
[02:24] <burek> llrcombs, sorry, I've never used it, I thougt you were talking about ffmpeg
[02:25] <llrcombs> burek: I'm encoding in ffmpeg and muxing in MkvMerge
[02:25] <burek> why
[02:25] <llrcombs> burek: because there's a nice, easy GUI for mkvmerge
[02:25] <burek> why not mux it in ffmpeg?
[02:26] <dadada> burek: sox
[02:26] <dadada> burek: and madplay for mp3 (but mp3 is about all madplay can..)
[02:28] <llrcombs> burek: 1. because I've been told it's preferable to mux in MkvMerge (though that doesn't really matter), and 2. because the muxing isn't the issue. Playing the MKA alone is choppy in VLC
[02:29] <burek> dadada, did you try converting audio to aac, just to see if it plays?
[02:29] <burek> llrcombs, I'm asking because: 1. it is easier to use 1 tool instead of 2
[02:30] <burek> and 2. muxing can cause audio/video async
[02:30] <burek> so you get buffer too early for audio stream for example :)
[02:30] <dadada> burek: no, but each tool gives a list of supported codecs, madplay just supports mpeg, sox supports a bunch but AFAIK not aac
[02:30] <llrcombs> burek: it can, yeah, and I understand that if it only happened after muxing, MMG could be to blame
[02:30] <llrcombs> burek: but it happens even when I take an MKA file straight from ffmpeg, so mkvmerge is definitely not the problem
[02:30] <burek> dadada, how do you install things on that router and why don't you install ffmpeg/ffplay?
[02:31] <burek> mka is audio only file right?
[02:31] <llrcombs> it's an MKV with an alternate extension indicating that there are no video streams
[02:31] <dadada> burek: opkg install package
[02:31] <burek> i see
[02:32] <burek> can you remux it into mp4 and try if it plays better, llrcombs ?
[02:32] <burek> dadada, then dpkg -i ffplay :)
[02:32] <dadada> burek: the repository has ffmpeg but no ffplay
[02:33] <burek> then install ffmpeg (it will install ffplay too)
[02:34] <llrcombs> burek: muxing with ffmpeg into any other encapsulation format causes the audio to play smoothly in VLC, but the duration decreases (as if it had been squeezed inward)
[02:34] <burek> llrcombs, that would indicate that vlc might not support mka properly
[02:34] <burek> or your mka is damaged
[02:35] <llrcombs> burek: the stream or the mux?
[02:35] <burek> (remuxing into other container proved that your audio stream is ok)
[02:35] <burek> mka as a format/mux
[02:35] <llrcombs> mka is an encapsulation format, never a codec, FYI
[02:36] <llrcombs> and that doesn't explain the duration decreasing whenever I use any other encapsulation format
[02:36] <dadada> burek: I installed the ffmpeg package, but no ffplay there, as I said, it's stripped down
[02:37] <burek> llrcombs, that's why I would decode audio stream into wav
[02:37] <burek> and check out the real duration
[02:37] <burek> to see if mka is erroneous or not
[02:37] <burek> before I do any other remuxing and testing
[02:37] <llrcombs> burek: alrighty, reencoding as a wav. What -acodec do I give it? Just "wav"?
[02:38] <burek> just set the output to out.wav
[02:38] <burek> ffmpeg will figure it out
[02:38] <llrcombs> writing...
[02:38] <burek> dadada, hmh, let me think
[02:40] <llrcombs> when it's a .wav, it shows the same, smaller duration
[02:40] <burek> that means that is the real duration
[02:40] <llrcombs> so maybe I need to stretch it out somehow, because it needs to be 2h4m31s
[02:41] <burek> now, base your conclusions on that
[02:41] <llrcombs> (the shorter one is 1h59m25s)
[02:41] <llrcombs> I need it longer so it matches with the video
[02:41] <burek> maybe it's damaged somewhere, so it has been truncated
[02:42] <llrcombs> my thought is that the actual stream is 1h59m25s long, there's an MKV header saying to stretch it out, and VLC is interpreting the header wrong
[02:43] <burek> why would the header tell anyone to stretch the audio stream?
[02:43] <llrcombs> and that ffmpeg can convert it to a different codec in an MKA while keeping the audio long, because MKAs have that header, but in any other encapsulation format, it jumps to the real duration
[02:43] <llrcombs> burek: so it syncs with the video
[02:43] <burek> no
[02:43] <burek> vlc does that by itself
[02:43] <burek> or, should i say
[02:43] <burek> it doesn't exactly do that, out of the box
[02:43] <llrcombs> it'd need to know how how long it's supposed to be, though
[02:43] <burek> it syncs the time stamps of audio/video
[02:44] <burek> like any other player
[02:44] <llrcombs> MKV/MKA definitely has a timescale header option
[02:44] <burek> but it doesn't just stretch the audio to fit video, because they are of the different size
[02:44] <llrcombs> meant for syncing up a video and audio track that are different lengths, but should be the same
[02:45] <burek> well, if mka is something new, maybe no proper support has been written for it, yet?
[02:45] <llrcombs> MKV/MKA aren't new
[02:45] <burek> why don't you just use audacity or some other audio editor
[02:45] <burek> and stretch your audio manually
[02:45] <burek> and finish the task
[02:46] <llrcombs> do you know how to do that in audacity, or shall I google?
[02:46] <burek> dadada, doo you know how to use pipes? mkfifo and stuff
[02:47] <burek> llrcombs, I think it's not a rocket science :)
[02:47] <llrcombs> I just don't use audacity much
[02:47] <burek> should be some option like stretch
[02:47] <dadada> burek: yeah, mkfifo not often, but pipes I like
[02:47] <burek> or so
[02:47] <llrcombs> looks like google's got me covered on that
[02:47] <burek> llrcombs, use your favorite audio editor then :)
[02:47] <llrcombs> "change tempo"
[02:47] <burek> dadada, try this:
[02:47] <llrcombs> burek: I don't edit audio much in general :P
[02:47] <burek> mkfifo out.wav /dev/dsp
[02:47] <llrcombs> but I've got it, thanks
[02:47] <burek> ffmpeg -i file.mp3 out.wav
[02:47] <burek> :)
[02:48] <burek> llrcombs, ok :) :beer: :)
[02:49] <burek> dadada, or ln -s /dev/dsp out.wav
[02:50] <llrcombs> oh wow, Audacity 2.0
[02:50] <llrcombs> or ffplay
[02:50] <llrcombs> oh, you don't have that
[02:50] <llrcombs> nevarmind
[02:51] <dadada> burek: hmm, ffmpeg is missing libavfilter ...
[02:51] <dadada> damn, that package is probably not well maintained
[02:51] <burek> dadada :/
[02:52] <burek> try checking out what console players are available for linux
[02:52] <burek> and try them one by one..
[02:52] <burek> llrcombs, I think it might be possible to stretch/shrink streams with ffmpeg and -vf option
[02:52] <burek> using PTS something
[02:53] <burek> but I need to check it out
[02:55] <dadada> burek: I got ffmpeg running
[02:56] <burek> weee :)
[02:59] <burek> llrcombs, try reading this: http://ffmpeg.org/ffmpeg.html#setpts
[02:59] <burek> I know someone was talking about it
[02:59] <burek> some time ago here
[02:59] <burek> on this channel
[02:59] <llrcombs> burek: reading...
[02:59] <dadada> burek: if I direct a wav file to /dev/dsp its pure garbage sound
[02:59] <burek> about using setpts filter to speed up/slow down streams
[02:59] <dadada> like white snow tv
[02:59] <burek> dadada, try typing ffmpeg -formats
[02:59] <rogelio> hola
[02:59] <dadada> /dev/dsp probablly expects something like pcm
[02:59] <rogelio> hello
[02:59] <burek> you are looking for pcm something
[02:59] <burek> yes
[03:00] <llrcombs> that's for video, though
[03:00] <rogelio> does anyone know how to crop borders with ffmpeg?
[03:00] <burek> dadada, so just add -f pcm_something out.wav
[03:00] <rogelio> i tried using -croptop -cropleft
[03:00] <rogelio> but it aint include in my windows build
[03:00] <burek> rogelio, http://ffmpeg.org/ffmpeg.html#crop
[03:00] <dadada> burek: I se no pcm there
[03:00] <burek> cropleft is deprecated long time ago
[03:01] <burek> dadada, you should see at least 10 of them
[03:01] <dadada> burek: I see 0
[03:02] <dadada> burek: http://fpaste.org/71UX/
[03:02] <rogelio> burek i have already tried that one, but it said: "Failed to set value '100:100:250:250' for option 'crop')
[03:02] <burek> rogelio,
[03:02] <burek> can you please use pastebin.com, to show your command line and its output?
[03:02] <rogelio> yes
[03:03] <dadada> dadada: sorry for not using pastebin and using my distros version of the same thing :-) kidding
[03:04] <burek> dadada, I think that build of ffmpeg is useless to you
[03:04] <burek> :/
[03:04] <rogelio> burek, here http://pastebin.com/A04qiuJ0
[03:04] <dadada> burek: what about "alsa" format?
[03:05] <burek> hm
[03:05] <burek> you can try it
[03:05] <burek> but I'm not sure if it will work
[03:05] <dadada> yeah, I tried it with -acodec alsa , says not supported
[03:05] <burek> anyway, sox, I think, has an option of creating a virtual audio device
[03:05] <burek> so you can mix several outputs into an input device
[03:06] <burek> and stuff like that
[03:06] <burek> so you might be able maybe to exploit that? :)
[03:06] <burek> rogelio,
[03:06] <burek> crop is not an option for ffmpeg, its a video filter
[03:06] <burek> so you need to do something like this
[03:07] <burek> ffmpeg ... -vf 'crop=...'
[03:07] <dadada> burek: sox is slow as hell.... or did you mean rogeleo?
[03:07] <burek> rogelio, if you read http://ffmpeg.org/ffmpeg.html#crop you should see there are examples at the bottom
[03:08] <dadada> also strange thing, I converted ogg to wav with ffmpeg but it plays slooow, whereis until now every wav I tried on the router played normal
[03:08] <burek> dadada, sox was for you, but let me check are there any other console players for linux
[03:08] <burek> that you can use
[03:08] <burek> can you play that wav on your computer
[03:08] <burek> to see if it's damaged
[03:11] <rogelio> burek thanks it worked
[03:12] <burek> dadada, http://moc.daper.net/
[03:12] <burek> rogelio, :beer: :)
[03:13] <dadada> burek: http://downloads.openwrt.org/snapshots/trunk/ar71xx/packages/ the repository I can use is pretty short :-)
[03:13] <dadada> maybe I could try gst, but maybe not simple to setup I don't know, has so many darn packages
[03:13] <burek> dadada, I hope you are not chatting over telnet? :)
[03:14] <burek> you're timing out frequently :)
[03:14] <burek> http://ffmpeg.gusari.org/irclogs/ffmpeg.log.20120408
[03:14] <dadada> no, it's just my wifi that makes probs
[03:15] <dadada> thanks, that logs doesn't show the timeouts though
[03:15] <burek> it shows just chat text that google can index :)
[03:16] <burek> and it's real time
[03:16] <dadada> in other words, this is totally not private
[03:16] <burek> for archive, you can check ffmpeg's mailing lists
[03:16] <burek> yes, it's not :)
[03:16] <dadada> heh
[03:16] <dadada> no problem
[03:16] <burek> it says in the topic too
[03:17] <burek> my friend would use to say: if you want privacy, just stay at home :)
[03:17] <burek> but we are already at our homes, so.. :)
[03:17] <dadada> makes sense, I also log most of my irc stuff, after all it's a significant part of my life ...
[03:17] <dadada> that says something about my life omg
[03:17] <dadada> heh
[03:18] <burek> :)
[03:18] <burek> that's ok :) we still love you :)))
[03:18] <dadada> :P
[03:18] <burek> but you should practice /join #real_life from time to time ^^
[03:19] <dadada> I just checked, nobody in that channel
[03:19] <burek> :O it's time to wake up then, Neo :)
[03:20] <dadada> rather to sleep, I'm a night-owl the last couple of weeks, I should try to normalize my rhythm so that I could function in day-life, but it's already too late at 03:20
[03:21] <burek> same here :)
[03:22] <burek> but I need to do some more things, so I'm not gonna sleep yet
[03:23] <dadada> I don't need to do things, but I've anxities about the next day so I rather stay in "this day" and try to avoid tranfering into the next day-day or something like that... of course you can always find things to do on the command line
[03:24] <burek> you should read more on astral projections :) that's a cool way to spend a night, while sleeping :)
[03:24] <burek> you soon realize you have so much spare time :)
[03:24] <dadada> do you do that?
[03:25] <burek> :)
[03:25] <dadada> seriously?
[03:26] <burek> it's kind offtopic, but :) let me just say that it doesn't take any of your "real" time :)
[03:26] <burek> you would spend it in sleep anyway :)
[03:26] <burek> so, you can try if you like :)
[07:55] <_SKiTZO> ive tried using -loop_input and -loop in combination with -t to have my audio input repeated, however i just get the input once and then silence.
[07:55] <_SKiTZO> is it even possible to loop audio? it works great with video
[08:13] <Khmar> Hello everyone, I'm trying to create a screencast using ffmpeg and I get messages like "[mpeg @ 0x7ca590]buffer underflow i=0 bufi=337262 size=341185". At the end, when I hit q, it seems to have recorded everything but there are lots of video frames skipped resulting in a choppy video. I tried passing -bufsize but then hitting `q` to exit the program doesn't work at all.
[08:17] <Khmar> In case it's relevant, this is my command line - "ffmpeg -f x11grab -s 1280x800  -r 50 -i :0.0 -f alsa -i pulse -sameq ./emacsmovies.org/screencast.mpg"
[08:39] <RPG-Master> I'm not sure if this is outside of y'all's expertice, but I keep getting this error from PS3 Media Server:
[08:39] <RPG-Master> INFO  2012-04-08 00:33:02.299 [New I/O server worker #1-15] Starting transcode/remux of 02 - Breathe.flac
[08:39] <RPG-Master> INFO  2012-04-08 00:33:02.832 [ffmpeg-20] Process ffmpeg has a return code of 1! Maybe an error occurred... check the log file
[08:53] <brimestone_home> hey guys, how do i use ffmpeg to convert a progressive 59.94fps to 29.976fps
[08:53] <brimestone_home> so technically, creating a slowmo effects
[10:48] <burek> RPG-Master,
[10:48] <burek> can you please use pastebin.com, to show your command line and its output?
[10:48] <burek> or the entire log file
[15:35] <Guest33361> hi. I want to create thumbnails from videos with ffmpeg, but at least for AVI files it seems that ffmpeg needs to read the whole file
[15:36] <Guest33361> is there a way to generate thumbnails fast, without ffmpeg needing to read the whole file?
[15:36] <Guest33361> I've seen that in ffplay, there is a "seek by bytes" option
[15:37] <Guest33361> I couldn't understand how it works, but may be seeking by bytes instead of time could do the job
[15:37] <Guest33361> is there a way to do this in ffmpeg?
[15:39] <Guest33361> no way?
[16:15] <zap0> thumbnailer, there are many methods, each file type is different.  a video is an abstract concept.  accept it.
[16:15] <thumbnailer> .F
[16:15] <thumbnailer> ok
[16:16] <thumbnailer> but the fact that I don't need very exact seeking may help
[16:16] <thumbnailer> I just want to create some thumbnails from a movie
[16:16] <thumbnailer> and I don't need to seek to an exact position or key frame
[16:17] <thumbnailer> theoretically, a rough seek in a file
[16:17] <thumbnailer> and then searching nearby for keyframe should do the job
[16:17] <thumbnailer> isn't it?
[16:22] <burek> thumbnailer,
[16:22] <burek> which command line did you use so far
[16:22] <thumbnailer> I've tried ffmpegthumbnailer in the first place
[16:23] <burek> did you try ffmpeg in the first place
[16:23] <thumbnailer> yes
[16:23] <thumbnailer> for example
[16:23] <burek> can you please use pastebin.com, to show your command line and its output?
[16:23] <thumbnailer> I didn't have any problems with extracting or thumbnailing
[16:23] <thumbnailer> but it was too slow
[16:24] <burek> I understand
[16:24] <burek> can you show your command
[16:24] <thumbnailer> because my files are on network location
[16:24] <thumbnailer> yes
[16:24] <thumbnailer> sure
[16:24] <thumbnailer> just a  moment
[16:27] <thumbnailer> ffmpeg -ss 100 -vframes 1 -s 320x240 -i video.mp4 thumb.jpg
[16:27] <thumbnailer> and I want to use it on a HTTP url
[16:27] <thumbnailer> like
[16:27] <thumbnailer> ffmpeg -ss 100 -vframes 1 -s 320x240 -i http://host.com/video.mp4 thumb.jpg
[16:27] <burek> ok, so
[16:28] <burek> if you try to do it on a format, that saves info about streams inside in the front of the file
[16:28] <burek> then you are ok
[16:28] <burek> otherwise it will be fun :)
[16:28] <thumbnailer> but it seems that ffmpeg reads the file until it reaches the 100 second reaches
[16:28] <burek> mp4 has the ability to save global headers
[16:28] <thumbnailer> so, what that it means?
[16:28] <burek> atoom structures
[16:28] <burek> and similar at the front of the file
[16:29] <burek> so that you retrieve that first
[16:29] <relaxed> thumbnailer: That command is wrong. It should be ffmpeg -ss 100 -i http://host.com/video.mp4 -vframes 1 -s 320x240 thumb.jpg
[16:29] <burek> that tells you which frames are keyframes
[16:29] <thumbnailer> sorry, actually I've tested it with AVI
[16:29] <burek> so you can seek it, etc, without downloading everything
[16:30] <burek> and yes, relaxed is right, your command would read only 1 frame from the input
[16:30] <burek> instead of generating 1 frame at the output
[16:30] <burek> the order of options does matter
[16:30] <thumbnailer> oh
[16:30] <thumbnailer> i didn't knew this
[16:30] <relaxed> and it defines the input as 320x240
[16:30] <relaxed> instead of scaling to 320x240
[16:32] <thumbnailer> I removed -s
[16:33] <burek> http://ffmpeg.org/ffmpeg.html#image2-2
[16:34] <burek> try ffmpeg -i http://host.com/video.mp4 -ss 100 -vframes 1 -f image2 -s 320x240 thumb.jpg
[16:36] <thumbnailer> you mean: ffmpeg -ss 100 -i http://host.com/video.mp4 -vframes 1 -f image2 -s 320x240 thumb.jpg
[16:36] <burek> no
[16:36] <burek> try what I offered
[16:36] <thumbnailer> tried both
[16:37] <burek> and?
[16:37] <relaxed> "-ss 100" before the input seeks to the position; after it decodes to the position.
[16:37] <thumbnailer> the one you said needs to get the whole file
[16:37] <thumbnailer> it seems
[16:38] <burek> if you set -ss before -i then ffmpeg needs to get meta data from the file, in order to know where are keyframes to be able to properly seek
[16:38] <thumbnailer> when putting "-ss 100" before the input, it says: Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
[16:38] <burek> if you put it after -i then ffmpeg will decode stream as it downloads it and will use that info to seek to the desired position
[16:38] <burek> can you please use pastebin.com, to show your command line and its output?
[16:39] <thumbnailer> yes, but pastebin.com is not accessible here
[16:39] <thumbnailer> this one is ok?
[16:39] <thumbnailer> http://www.heypasteit.com/
[16:39] <burek> yes
[16:39] <thumbnailer> but it's blocked here
[16:39] <thumbnailer> you know?
[16:39] <burek> ok, use that one
[16:39] <thumbnailer> :-(
[16:39] <burek> btw google indexes pastebin.com in the matter of seconds
[16:40] <burek> it's the only site that I know of that has such a privilege
[16:40] <thumbnailer> google even started to index what you see
[16:40] <thumbnailer> :-)
[16:40] <thumbnailer> with the latest internet-connected glasses
[16:42] <thumbnailer> it doesn't exit
[16:42] <thumbnailer> because the file is large
[16:42] <thumbnailer> I have to press q to exit
[16:42] <thumbnailer> but if I wait, it will do the job
[16:42] <thumbnailer> I press q and send the result
[16:43] <burek> then use -ss before -i
[16:43] <burek> whatever does the work done
[16:43] <burek> gets*
[16:43] <thumbnailer> ok
[16:44] <burek> anyway, if it works with -ss before -i does it do what you wanted?
[16:45] <burek> brb
[16:45] <thumbnailer> ok
[16:48] <thumbnailer> This is the first (ss after input)
[16:48] <thumbnailer> http://www.heypasteit.com/clip/0AA8
[16:48] <thumbnailer> and this is the second
[16:48] <thumbnailer> (ss before input))
[16:48] <thumbnailer> http://www.heypasteit.com/clip/0AA9
[17:01] <thumbnailer> :-)
[17:01] <thumbnailer> it seems that the problem is with the nerwork module
[17:01] <thumbnailer> because it does the job great on local files
[17:01] <thumbnailer> nearly instant
[17:31] <thumbnailer> are u there?
[17:37] <thumbnailer> what if i want to extract more frames?
[17:37] <thumbnailer> -vframes 2 ?
[18:32] <level09> why is my x264 video not running on html5 players ?
[18:51] <thumbnailer> I think even with -ss before input
[18:51] <thumbnailer> FFMPEG reads all the file
[18:51] <thumbnailer> :-(
[19:44] <cryptopsy> does anyone know how to keep the same bitrate on sox? sox is a tool that uses ffmpeg i think, that can be used to trim or split songs (at least that's what i'm using it for, amongst other features)
[20:38] <juanmabc> i guess you are alone at "man sox" and searching for rate, bit-rate
[21:56] <cryptopsy> juanmabc: it is mentioned only as a definition
[21:57] <cryptopsy> are you telling me it's impossible? give me a flying fucking break
[22:26] <relaxed> cryptopsy: If your sox has ffmpeg support then I would guess it could stream copy if all you're doing is cutting.
[22:27] <relaxed> meaning you wouldn't lose any quality.
[22:29] <relaxed> Keep in mind this isn't #sox and you have not really give any details about what format you're dealing with.
[22:31] <cryptopsy> there is no #sox
[22:31] <cryptopsy> ffmpeg and mplayer channels are like the video and sound editing communities on this network
[22:32] <relaxed> Which is why I answered your question.
[22:32] <cryptopsy> if sox cannot perform, then ffmpeg gains another user by figuring out how to trim audio
[22:33] <relaxed> sox is much better at trimming audio
[22:34] <relaxed> decode the audio to wav -> trim with sox and output wav -> compress wav to flac/alac via ffmpeg
[22:35] <cryptopsy> mm actually i think sox uses lame
[22:35] <relaxed> That method should be lossless.
[22:36] <relaxed> if sox has ffmpeg support it can use pretty much anything.
[22:38] <relaxed> So if your source is mp3, I assume trimming with sox would stream copy the section you want to the output. I'm not a 100% sure though.
[22:39] <relaxed> sox does have a mailing list.
[22:40] <cryptopsy> i'd rather use lame or ffmpeg if it's possible than email an abandoned software
[22:42] <relaxed> It's not abandoned. Use whatever you like.
[22:43] <cryptopsy> there is a fine line that is actually a really fat line which is drawn when determining if a software is abadoned. on myspace, girls will tilt their fat faces at an angle, to appear more desirable by the bourgeois. in software, mailing lists and websites are kept online, despite authors having no interest in fixing the issue.
[22:46] <highzeth> how one can claim a project being abandoned, when the last commit to it was less than a day ago, well, that brings a new definition to the word for me
[22:49] <relaxed> highzeth: Because there's no irc channel with people falling all over theirselves to help Mr. Piss Poor Attitude here.
[22:50] <cryptopsy> highzeth: people like to participate, despite no long-term goals by the original authors
[22:50] <cryptopsy> people write patches instead of starting their own forks under obscure names
[22:51] <cryptopsy> relaxed: IRC channels are not made for helping single people. we live in a society, where people help each other and in exchange you get something done for you
[22:51] <cryptopsy> that's what the smurfs was about, do you know the story?
[22:51] <highzeth> relaxed: hehe
[22:51] <cryptopsy> in the smurfs, doing a job meant getting 10 jobs done in return for you
[22:51] <cryptopsy> that's what communities are about
[22:52] <cryptopsy> irc channels are communities
[22:55] Action: relaxed had Smurfs bedspread and pillowcases as well as all the cups from Hardee's when he was a kid
[22:55] <relaxed> It was pretty smurfy.
[22:58] <relaxed> I also had a cat named Azrael.
[23:58] <cnx> i run ffmpeg -f mpeg -vc 1 -tvstd NTSC -i /dev/video0 -y test.mpg i get a few lines but  when i hit control c no file is created, why?
[00:00] --- Mon Apr  9 2012


More information about the Ffmpeg-devel-irc mailing list