[Ffmpeg-devel-irc] ffmpeg.log.20111208

[burek]

[00:11] <alyawn> so for map_channel to work, I need to specify -map_channel -1 for each channel I don't want, right?
[00:20] <ubitux> alyawn: no no
[00:20] <ubitux> -map_channel 0.0.0 -map_channel 0.0.1
[00:20] <ubitux> this will create a stream with 2 channels
[00:20] <ubitux> if you add a few -map_channel -1, it will add muted channels
[00:21] <alyawn> ah ok
[00:21] <ubitux> as said in the doc, the number of -map_channel option you specify will be the number of output channels in the stream
[00:21] <alyawn> ok... that makes more sense
[00:21] <alyawn> thanks for the help
[00:22] <ubitux> np :)
[00:46] <freeminds> can I force ffmpeg to output the convert status in a different way?
[00:48] <_klk_> hi all
[00:48] <_klk_> i'm running into link errors when using statically bit libav* libraries from ffmpeg 0.8.5
[00:48] <_klk_> *statically built
[00:49] <_klk_> here's the error: http://pastebin.com/wXE9A9wM
[00:50] <_klk_> here's the configuration: configuration: --enable-static --disable-shared --enable-gpl --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --prefix=../..
[00:51] <_klk_> i am including -lz in my list of libraries along with the libav*
[00:51] <_klk_> does anyone know why this is happening?
[00:59] <osearth> _klk_: well,  first your mom and dad met
[01:00] <_klk_> osearth:  what happened next???
[01:00] <osearth> i forget your mom got me drunk
[01:01] <_klk_> :( did she take all your money
[01:01] <osearth> no i was clear i needed cash up front
[01:01] <osearth> ';]
[01:01] <_klk_> oh lame i fixed this problem & just had to put the failing libraries AFTER the -lav* in my flags to gcc
[01:02] <osearth> never seen that error
[01:02] <osearth> nice i knew needed include
[01:15] <walisser> hi
[02:08] <StijnH> Hello. I see that FFmpeg is released as GPL and LGPL. If I make a program that uses FFmpeg, can the program itself be released under the MIT license, or does it need to have the same license?
[02:17] <pasteeater> freeminds: what do you mean?
[03:10] <freeminds> pasteeater, fixed already thx anyway
[03:10] <freeminds> I meant: FFmpeg outputs the convert status by updating the last line
[03:11] <freeminds> But for my application it would have been better if it would just print a new line
[03:37] <jbwiv__> guys, I have a video (info here: http://pastie.org/2983851). from the way I read this output, the current bitrate is 147 kb/s. I'm trying to reduce the file size, so I've tried various things like "ffmpeg -b 24k -i UI_Overview.mp4 newvid.mp4",  and I've varied the bitrate setting, but *anything* I do results in a video significantly larger. can anyone tell me what I should do to reduce it?
[03:38] <sacarasc> You're doing it wrong!
[03:38] <sacarasc> You put the bitrate as an input option, not an output option.
[03:38] <sacarasc> ffmpeg [input options] -i input [output options] output
[03:39] <jbwiv__> sacarasc: aha...that explains it. thanks ;-)
[03:41] <jbwiv__> sacarasc: actually, even this creates a bigger file "ffmpeg -i UI_Overview.mp4 -b 16k UI.mp4"
[03:42] <sacarasc> Pastebin the complete output.
[03:42] <jbwiv__> in my example above, I typed it in wrong. I've really been putting it after the input
[03:42] <jbwiv__> ok, one sec
[03:44] <jbwiv__>  sacarasc: I killed it when it got to 30M (the original file is 28M), but is this enough output? If not, I'll let it run all the way: http://pastie.org/2983876
[03:47] Action: sacarasc shrugs. 
[03:47] <sacarasc> Though you're using Libav, which is a fork and has a different channel. #libav I think.
[03:48] <jbwiv__> sacarasc: should I be using something else? i just apt-get'd the ffmpeg package
[03:49] <sacarasc> Ubuntu ships Libav rather than ffmpeg. There's a bunch of politics and I don't know which is best or even if there are many differences between them.
[03:50] <jbwiv__> sacarasc: gotcha. ok, I'll go try libav. thanks much
[03:51] <jbwiv__> guys, I have a video (info here: http://pastie.org/2983851). I'm trying to shrink it on ubuntu with ffmpeg, which appears to use libav. from the way I read this output, the current bitrate is 147 kb/s. I'm trying to reduce the file size, so I've tried various things like "ffmpeg -i UI_Overview.mp4 -b 16k newvid.mp4", and I've varied the bitrate setting, but *anything* I do results in a video significantly larger. can anyone tell me 
[03:51] <jbwiv__> what I should do to reduce it?
[03:51] <jbwiv__> woops
[03:52] <jbwiv__> sheesh, sorry, irc party foul
[04:38] <jeremyS> Can someone give me a command line option for the current git master version ffmpeg which will use an audio filter from libavfilter?
[04:40] <jeremyS> I have tried -af, -audiofilter and -filter:a
[04:51] <relaxed> jeremyS: -vf
[04:52] <jeremyS> for audio filters, or video filters?
[04:52] <relaxed> I believe it's for all filters
[04:52] <jeremyS> ohhhh.
[04:52] <jeremyS> Mencoder uses -vf =video filters -af = audio filters, so I assumed vf would be only for video filters.
[04:53] <jeremyS> hmmm... that doesn't work with the volume plugin
[04:54] <jeremyS> It doesn't give an error, but it doesn't change the volume.
[04:57] <jeremyS> Also, the documentation (doc/ffplay.html) says "A filtergraph can be represented using a textual representation, which is recognized by the -vf and -af options of the ff* tools, and by the avfilter_graph_parse() function defined in ‘libavfilter/avfiltergraph.h’. "
[04:58] <jeremyS> but using  -af volume=0.1 with ffmpeg  give the error "Unrecognized option 'af' Failed to set value 'volume=0.1' for option 'af'"
[05:07] <calc84> hello
[05:08] <calc84> I can't figure this out, how can I force ffmpeg to not use the audio interleave preload duration in AVI output?
[05:10] <relaxed> jeremyS: it's -vf
[05:10] <calc84> actually, never mind, I don't even need to do that :/
[05:10] <calc84> I'm trying to figure out exactly what is causing some of my generated video files not to show up on my Nintendo 3DS
[05:11] <calc84> and -vf isn't very specific at all anyway
[05:13] <calc84> What bugs me is that I make 3 videos using the exact same options and only two show up on the 3DS
[05:14] <calc84> and the one that didn't show up had an interleave preload duration, whereas the others did not
[05:14] <relaxed> calc84: maybe it's the frame size or the par isn't 1:1
[05:14] <calc84> par?
[05:14] <relaxed> pixel aspect ratio
[05:14] <calc84> well, all 3 output videos were from the same source video
[05:14] <relaxed> put ffmpeg -i $file on pastebin for all 3
[05:15] <calc84> I split it up into sections of 9:59 becasue that's the maximum size that shows up
[05:17] <calc84> http://pastebin.com/GqmeL2fk
[05:17] <calc84> VID_0001 is the one that doesn't work, VID_0002 and VID_0003 do work
[05:18] <relaxed> does it not support h264/aac in mp4?
[05:18] <calc84> nope, just mjpeg/adpcm
[05:18] <calc84> because that's the output of the video recorder
[05:19] <calc84> it wasn't meant for loading custom videos, but of course I'm going to do it anyway :P
[05:21] <calc84> I don't think interleave preload duration actually has anything to do with it, because the videos I actually recorded on the 3DS have that as non-zero
[05:21] <relaxed> can you stick up a sample that was recorded using the 3DS?
[05:22] <burek> calc84, try setting the duration to 9:50 for the 1st video
[05:22] <calc84> okay
[05:27] <jeremyS> relaxed: so are you telling me that the volume filter is broken?
[05:28] <calc84> the 9:50 version doesn't show up either
[05:29] <burek> http://ffmpeg.org/ffmpeg.html#volume
[05:29] <burek> calc84, I was thinking, maybe the file size is what makes your 3DS not to play the file, not the file duration
[05:30] <calc84> I don't think so, I think I had one >100MB play before
[05:31] <calc84> hmmm
[05:31] <calc84> [avi @ 0000000001D3D280] parser not found for codec adpcm_ima_wav, packets or times may be invalid.
[05:31] <burek> calc84, ffmpeg -codecs | grep  adpcm_ima_wav
[05:32] <calc84> I know I have it, it says DEA D
[05:32] <burek> jeremyS
[05:32] <burek> can you please use pastebin.com, to show your command line and its output?
[05:33] <burek> calc84, ok :) then back to the lab :)
[05:33] <calc84> that was from when I did the pastebin though
[05:33] <calc84> I'll see if anything comes up when actually encoding
[05:33] <relaxed> jeremyS: No. I don't get any errors when using it.
[05:33] <burek> what cmd did you use to create those 3 files
[05:34] <calc84> ffmpeg -i "tmp.avi" -async 1 -vcodec copy -acodec adpcm_ima_wav -ss 00:00:00 -t 00:09:50 "VID_0001.AVI"
[05:34] <calc84> earlier I did this:
[05:34] <calc84> ffmpeg -i %input% -vf scale=320:240 -aspect 4:3 -r 20 -vcodec mjpeg -qscale %quality% -acodec pcm_s16le -ar 44100 -ac 2 "tmp.avi"
[05:34] <burek> I think you don't need -aspect along with -vf scale
[05:34] <burek> it's redundant
[05:35] <calc84> yeah, it was left over from somebody else's thing :P
[05:36] <burek> calc84, can you show the cmd+output for the 1st file
[05:36] <relaxed> I would use mkvmerge --split duration to cut the source into 9:50 pieces. Then have ffmpeg encode each one.
[05:36] <burek> using pastebin of course
[05:36] <jeremyS> burek: http://pastebin.com/cGi09z8B
[05:37] <calc84> I'll show it for the whole batch if you want
[05:37] <burek> jeremyS, instead of -vf put -af ?
[05:37] <jeremyS> I already posted the output to that above in irc
[05:38] <jeremyS> It gives an error instead of just silently not working
[05:38] <burek> can you please use pastebin.com, to show your command line and its output?
[05:39] <jeremyS> Okay
[05:39] <jeremyS> http://pastebin.com/EbnTAGTi
[05:40] <calc84> Here, I accidentally the whole thing: http://pastebin.com/FzcEJHxN
[05:40] <calc84> First is tmp.avi, then are VID_0001 to VID_0003
[05:42] <calc84> it's also complaining about pcm_s16le, even though that is also DEA D
[05:44] <burek> jeremyS, try af=volume=0.1:1
[05:45] <burek> wait sorry
[05:45] <jeremyS> ?
[05:46] <burek> calc84, Invalid and inefficient vfw-avi packed B frames detec
[05:46] <burek> ted
[05:46] <burek> you have an error in the input stream
[05:46] <burek> try using -ss (seek) to skip a part of a video
[05:46] <burek> ffmpeg -ss 00:00:05 -i ...
[05:46] <calc84> it was a problem with the input video the entire time? >_<
[05:46] <burek> i guess
[05:46] <burek> jeremyS, just a sec
[05:47] <calc84> yeah, I guess I should have tried some other videos too
[05:48] <calc84> hmm, putting in another video gives the same message D:
[05:48] <calc84> I'll try starting later in the video like you said
[05:50] <burek> calc84, I think the error is at the very beginning of the file
[05:50] <burek> that's why I suggested to use seek
[05:50] <burek> to skip it
[05:50] <burek> jeremyS, did you compile the ffmpeg from latest git?
[05:50] <calc84> starting at 5 seconds gives the same thing
[05:51] <jeremyS> yes
[05:51] <jeremyS> I just git cloned a few seconds ago
[05:51] <jeremyS> *minutes
[05:51] <calc84> oh wait
[05:52] <calc84> I was starting at 5 seconds in the wrong one
[05:52] <burek> jeremyS, you might have just found a bug :)
[05:52] <burek> can you try with 5 instead of 0.1
[05:53] <jeremyS> sure
[05:53] <jeremyS> I think I already tried it with 10 with no effect
[05:53] <jeremyS> but I'll check
[05:53] <calc84> okay, even doing that in the right one doesn't work
[05:54] <jeremyS> burk:yeah, no effect
[05:54] <calc84> though, come to think of it
[05:54] <burek> jeremyS, can you please create a bug report for that
[05:54] <calc84> wouldn't it detect that frame when seeking, anyway?
[05:54] <jeremyS> I was going to start development of LV2 plugin support for libavfilter...lol
[05:55] <burek> calc84, i guess it would, but i was hoping it would also skip it and not encode it into the output
[05:56] <calc84> well, that's possible too, I'll go through with the whole thing and try again
[06:00] <calc84> ah, it shows up now (I used an offset of 10 seconds, but less probably would have been fine too)
[06:03] <burek> :)
[06:04] <jeremyS> burek: alright, bug reported: https://ffmpeg.org/trac/ffmpeg/ticket/720
[06:04] <burek> i think -ss 00:00:00.040 will also do :)
[06:04] <burek> thanks a lot jeremyS :)
[06:05] <burek> btw jeremyS, if you need to do get the job done, without waiting for the fix, you can use one of the older gits or use a stable release 0.8.5 or something
[06:07] <burek> jeremyS, also, please add the pastebin output of using -af too
[06:07] <burek> so that devels can see you tried the correct option
[06:07] <jeremyS> burek: But that clearly says that the option is not supported
[06:07] <burek> exactly, because it should be
[06:07] <jeremyS> ooh
[06:07] <burek> it's in the docs http://ffmpeg.org/ffmpeg.html#volume
[06:08] <jeremyS> K
[06:08] <jeremyS> What is a version which has working libavfilter audio plugins?
[06:08] <relaxed> -filter:a doesn't work either
[06:08] <jeremyS> https://ffmpeg.org/trac/ffmpeg/ticket/720
[06:08] <jeremyS> yeah.
[06:09] <jeremyS> but it doesn't say it's not an option.
[06:11] <calc84> yep, 00:00:00.040 does indeed do it :D
[06:11] <burek> :beer: :)
[06:11] <burek> hmh, jeremyS, you can try apt-get install ffmpeg maybe
[06:11] <burek> just to get your job done
[06:11] <calc84> I was wondering why nobody on neogaf seemed to be having the same problem as me
[06:11] <burek> i mean, if you don't want to wait for the fix :)
[06:12] <jeremyS> burek: I believe I get libav ffmpeg
[06:12] <jeremyS> which has the plugins stripped
[06:12] <jeremyS> it only has the anull audio plugin
[06:13] <jeremyS> Well how long are we talking till it gets fixed? a few days, weeks, months?
[06:14] <burek> I really can't tell, because frankly I don't know
[06:14] <jeremyS> So *perhaps* the libav libavfilter is working, but I can't tell because the only plugin I can use is one which is supposed to do nothing to the audio.
[06:14] <relaxed> jeremyS: libav has '-vol volume'
[06:14] <jeremyS> Do you know a past version that should work?
[06:14] <burek> hmh
[06:14] <relaxed> or you could use sox
[06:15] <jeremyS> relaxed: I'm trying to get a reference environment for developing a libavfilter plugin
[06:15] <jeremyS> I don't actually care about changing the volume of a file
[06:15] <relaxed> then what do you care about?
[06:16] <jeremyS> Having an audio filter which works, so I can compare my new one to it.
[06:16] <jeremyS> So I know any issues are on my end, and not a problem with ffmpeg
[06:17] <jeremyS> If I could find any filter, that would be fine, there's nothing special about the volume one, except I assumed it would be the simplest, so it would be most likely to work
[06:19] <burek> jeremyS, the volume filter has been added on 2011-11-05
[06:19] <burek> http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=volume
[06:20] <burek> and I don't see any updates for it, so I guess the bug exists since the beginning :/
[06:20] <jeremyS> bruek: lol... I assumed it would be the first one written.
[06:20] <burek> :)
[06:21] <jeremyS> So which one would you suggest I try?
[06:21] <burek> http://ffmpeg.org/ffmpeg.html#toc-Audio-Filters
[06:21] <burek> try the others
[06:21] <burek> and see if they can help
[06:21] <jeremyS> I tried earway
[06:21] <jeremyS> *earwax
[06:21] <jeremyS> didn't wokr
[06:21] <jeremyS> *work
[06:21] <calc84> bye, thanks for the help guys!
[06:21] <jeremyS> and pan didn't work
[06:21] <burek> calc84, :beer: o/
[06:21] <burek> none of them work?
[06:21] <jeremyS> I haven't tried them *all* but the ones I listed, yeah
[06:22] <jeremyS> ashowinfo did nothing
[06:22] <relaxed> aresample does nothing
[06:22] <jeremyS> hmmm... so I guess none of them are working?
[06:24] <jeremyS> yeah, and aconvert does nothing either
[06:24] <jeremyS> hmmm... scratch that! anull works!
[06:24] <jeremyS> It does exactly what it's supposed to!
[06:25] <burek> :))))
[06:25] <burek> jeremyS, how did you specify params for ashowinfo
[06:26] <jeremyS> I didn't give any params, the documentation didn't list any
[06:27] <burek> it might be that the cmd line parser is broken for -af
[06:27] <burek> i just tried all those filters
[06:27] <burek> all give the same error about unrecognized params
[06:27] <burek> *option
[06:27] <jeremyS> I would believe that
[06:31] <relaxed> doesn't -filter:a = -af
[06:31] <relaxed> they should get rid of -vf/-af and stick with -filter
[06:32] <Skaag> how do I create mp41 instead of mp42 when I transcode to mp4?
[06:33] <burek> relaxed, jeremyS  this actually works: ffmpeg -i 2011-10-31.mp4 -filter:a ashowinfo -f mp4 /dev/null
[06:34] <burek> or it doesn't :)))
[06:34] <burek> i mean, the command works, but i dont see any output of ashowinfo :)
[06:35] <jeremyS> oh yeah
[06:35] <jeremyS> lol
[06:35] <burek> Skaag, did you try -vcodec mpeg1video
[06:35] <Skaag> hm. no
[06:36] <burek> Skaag, try: ffmpeg -codecs | grep mpeg
[06:37] <Skaag> right now I use -vcodec libx264
[06:37] <Skaag> I'm targeting old iPhones 
[06:38] <burek> i think there is a preset for iphones
[06:38] <Skaag> for some reason it doesn't work for me& :-\
[06:38] <Skaag> this is what I use right now: http://pastebin.com/8i45L64d
[06:41] <burek> you should really use -preset and -tune instead all those x264 specific options
[06:41] <Skaag> I'm trying to find a guide on this
[06:42] <burek> also, try -f ipod
[06:42] <Skaag> I tried to google 'ffmpeg iPhone preset' but it gives me all kinds of weird pages
[06:44] <Skaag> that gives me: [ipod @ 0x33f0ce0] Warning, extension is not .m4a nor .m4v Quicktime/Ipod might not play the file
[06:44] <Skaag> I set the extension as .mp4
[06:44] <burek> hmh
[06:46] <Skaag> I know that page, it does not use presets like you say
[06:46] <Skaag> I read it earlier today
[06:49] <burek> well
[06:49] <burek> type x264 --help
[06:49] <burek> and read about tune and preset
[06:49] <burek> and then add it to ffmpeg like: ffmpeg -i ... -preset blah -tune blah -f mp4 ...
[06:54] <jeremyS> burek: do you know a version+plugin combination that works?
[06:55] <burek> plugin?
[07:20] <jeremyS> burek: filter
[07:24] <burek> jeremyS, no I don't, but you can try to search on git when was each filter added
[07:25] <jeremyS> k, thanks
[07:27] <Skaag> I think the source I'm given is broken to begin with
[07:28] <Skaag> burek: with the -vpre iphone preset, the iPhone gives me a "This movie could not be played" error...
[07:33] <burek> why -vpre
[07:33] <burek> use -preset
[07:33] <burek> vpre is deprecated
[07:44] <Skaag> oh I see
[07:45] <Skaag> it's what I find in the ffmpeg documentation& :-\
[08:22] <GordonFreeman> hi
[08:53] <Skaag_> I'm trying to achieve h264 (Constrained Baseline) but no matter what params I pass to ffmpeg it remains on (high)
[10:14] <bove> Do I need any extra arguments for multi threading a ProRes encode in 0.8.7?
[10:16] <kollapse> Hi. I have 2 files here - a reference file called X and another file called Y. How can I convert Y to have exactly the same characteristics as X (same resolution, audio/video codec, fps etc.) ?
[10:27] <impulze> convert but texactly the same characteristics?
[10:27] <impulze> *but exactly
[10:27] <impulze> i can't follow
[10:27] <impulze> cp?
[10:35] <kollapse> impulze: Well, I want to transcode file Y to have the same resolution, codec, fps etc. of file X
[10:36] <kollapse> as file X*
[10:38] <kollapse> Look here: http://pastie.org/2985072 I have two files, as can be seen: MVI_5999.AVI and video.mp4 - I want to transcode video.mp4 so that it will be the same type of file as MVI_5999.AVI (Video: mjpeg, yuvj422p, 640x480, 20 tbr, 20 tbn, 20 tbc, Audio: ... etc.)
[10:43] <kollapse> Any ideas ?
[10:53] <bove> a "get settings from existing file" would be nice, but I don't think it exists today
[10:58] <kollapse> bove: Hmm, so I have to manually match the parameters ?
[10:59] <kollapse> The reason I need this is that I want to upload a video file to a camera for playback. But understandably, the camera only supports a very specific type of file.
[10:59] <kollapse> In this case, mjpeg / yuvj422p
[11:04] <bove> upload the result of 'ffmpeg -i filefromcamera' to pastebin.com and I'll try to help matching it
[11:05] <kollapse> bove: Well it's here, the first file (MVI_5999.AVI) - http://pastie.org/2985072
[11:37] <sybariten> bove: thats exactly a feature i've been thinking of, too! Surely this is a thing that must exist, somewhere in the vast video encoding business?
[11:38] <Peace-> hi there , i would like undestand this error  ffmpeg    -s $(xrandr  | awk '/, current /{print $8}')x$(xrandr  | awk '/, current /{gsub(/\,/,"");print $10}') -f x11grab -r 10  -i :0.0  -vcodec gif output.gif
[11:39] <Peace-> Could not write header for output file #0 (incorrect codec parameters ?)
[11:39] <Peace-> but i have this 
[11:39] <Peace->  DEV D  gif             GIF (Graphics Interchange Format)
[11:52] <Peace-> ok now it works
[11:52] <Peace-> but black gif :D
[11:53] <Peace-> ffmpeg    -s $(xrandr  | awk '/, current /{print $8}')x$(xrandr  | awk '/, current /{gsub(/\,/,"");print $10}') -f x11grab -r 10  -i :0.0  -pix_fmt rgb24 -y output.gif
[11:56] <bove> sybariten: We could get pretty far by just reading the Input and stream lines from the existing file
[11:57] <bove> but a lot of files would also need very specific muxer options to be compatible
[12:09] <Shimmy> Anyone knows why this project: http://code.google.com/p/amv-codec-tools isn't combined with the original FFmpeg development?
[12:09] <Shimmy> It allows for FFmpeg to support AMV files
[12:10] <Shimmy> the problem is I don't know how to compile it with the up-to-date FFmpeg version
[12:10] <Shimmy> You can see my question here as well: http://stackoverflow.com/questions/8429733/a-windows-compatible-c-ide-that-will-be-able-to-compile-ffmpeg
[12:16] <Peace-> hahahahahaha 
[12:16] <Peace-> it works!
[12:16] <Peace-> :D ty all anyway :P
[12:19] <kollapse__> bove: Any luck ? :)
[12:53] <shevy> Hi.
[12:53] <shevy> One can do a screencast with FFmpeg like so:
[12:53] <shevy>   ffmpeg -f x11grab -y -r 12 -s 800x600 -i :0.0+480,200 -vcodec ffv1 -sameq out.avi
[12:53] <shevy> Is there a way to do "stop after 30 seconds" ?
[12:53] <shevy> except hitting Ctrl-C. Reason is, I'd like to use this in a script.
[12:58] <Mavrik> shevy, -t 00:30:00 
[12:58] <shevy> oh
[12:59] <shevy> hmm that does not seem to want to work
[13:00] <Compn> shevy : you must be putting that command in the wrong order then
[13:00] <Compn> also its -t 30
[13:00] <Compn> for 30 seconds
[13:01] <Compn> -t 30:00 is 30 minutes i think
[13:01] Action: Compn wonders what Mavrik is doing
[13:01] <shevy> aaaaaah
[13:01] <shevy> yes
[13:01] <shevy> that was it :)
[13:01] <shevy> it works now, thanks Mavrik and Compn 
[13:01] <Compn> np
[13:03] <Mavrik> Compn, errr, yea
[13:03] <Mavrik> my mistake :)
[13:32] <Diogo> hi, this is not the right channel buy anyone work with erlyvideo?
[13:33] <Diogo> *but
[13:39] <raven> hi
[13:39] <raven> maxrate obviously does not work - any suggestions?
[14:23] <Bove> kollapse: What have you tried in making your file?
[14:32] <kollapse> Bove: Does this look correct- ffmpeg -i video.mp4 -vcodec mjpeg -s vga -r 20 -acodec pcm_u8 -ac 1 -ar 11024 -ab 88 OUT.AVI      Reference file is here - http://pastie.org/2985876
[14:33] <Bove> you probably need some bitrates in there
[14:33] <Bove> ffmpeg -i Y -vcodec mjpeg -pix_fmt yuvj422p -s 640x480 -r 20 -b 16000k -acodec pcm_u8 -ar 11024 -ac 1 -ab 88000
[14:36] <kollapse> Ran ffmpeg -i on the resulting file. Audio is a match, but it shows me bitrate: 5727 kb/s
[14:37] <kollapse> Even if -b 16000k
[14:37] <Bove> you can try to add -minrate 16000k -maxrate 16000k
[14:40] <kollapse> A constant bitrate of 16000k would probably increase file size, right ?
[14:41] <Ave> unless you are bandwidth limited/constrained, its better to just use constant quality setting
[14:42] <kollapse> Ave: I am limited by what the camera supports.
[14:42] <Bove> I would try with the file you made from my line
[14:46] <kollapse> Strange... camera says "Unrecognized format". The two files are pretty similar though - http://pastie.org/2985920
[14:48] <Bove> maybe you need a header? Or does the camera make thumbnail files for all videos? (Can't remember the extension)
[14:49] <kollapse> The camera does make a thumbnail file for videos. I used a thumbnail from another video. (it uses .THM - but are actually jpeg)
[14:50] <kollapse> Maybe it checks the metadata to see if the encoder is "CanonMVI06" ? And if not, it refuses to play ?
[14:50] <sybariten> Bove: well... yes, i think theres still quite a bit of discrepancy between what you read in the ffmpeg output, and what you would supply as encoding instructions
[14:51] <Bove> sybariten: Buffer sizes, rate limits etc. would probably be hard to read from an encoded file as well
[14:52] <kollapse> Is there any way to force modify the metadata ?
[14:52] <sybariten> Bove: yeah... but still
[14:52] <sybariten> A lot of questions on mailinglists etc maybe, could be avoided   :)
[14:53] <Bove> kollapse: Check out http://www.videotoolshed.com/product/42/qtchange
[14:54] <Bove> sybariten: It would probably be a lot easier for the devels
[14:54] <kollapse> Well that costs (1) and there's no Linux version (2)
[14:55] <Bove> kollapse: I was thinking more for the research phase. There is a demo version, but if you're linux only...
[14:56] <kollapse> Well I don't know why it won't run this video. It should have been fooled.
[14:57] <Bove> have you tried without audio? -an
[14:59] <Bove> kollapse: You might also try qtstreamize (don't know if that's built into the regular ffmpeg branch yet)
[15:00] <kollapse> Bove: That makes the video streamable ?
[15:04] <Bove> puts the header in front
[15:06] <kollapse> I tried using -metadata encoder="CanonMVI06" but it still sets Lavf53.5.0
[15:06] <kollapse> And -metadata creation_time="..." doesn't set.
[15:08] <Bove> I might need an excact size for the header. What if you try and compare 3 files for the camera and see if you can find the header size
[15:09] <kollapse> Bove: Oh, so using -metadata won't work?
[15:10] <Bove> It could be the camera expects n bytes of header before the actual stream begins
[15:10] <kollapse> How can I determine the header size ?
[15:13] <Bove> If the bitrate is cbr, you can compare duration vs. file size on 3 files
[15:16] <kollapse> I don't think it's cbr. This 6 second video has 12.5 mb. Calculation duration (seconds) * bitrate gives about 104 mb
[16:16] <relaxed> kollapse: run mediainfo on both files
[16:21] <OneFix_Work> Is there any hardware (besides a bigger CPU) that can be bought to speed up x264 encoding?
[16:21] <Compn> ask in #x264 
[16:22] <Compn> there are somethings, also there are hardware h264 encoders
[16:22] <Compn> but better ask them for input
[16:23] <OneFix_Work> Well, I know there is the acceleration that uses 3D cards, but I thought that was mainly for playback and not encoding...
[16:24] <Compn> there is some work being done to speed up encoding, not sure how far it has gotten
[16:24] <Compn> like i said, ask in the #x264 channel, they will know more
[16:25] <Mavrik> OneFix_Work, faster processor with more cores
[16:26] <OneFix_Work> Mavrik: Right, but that means at least 4 cores and faster than 2.5GHz which is usually really expensive
[16:27] <OneFix_Work> At this point, I would rather spend $200 to buy something that can offload the encoding process rather than spend $300 (minimal) for yet another CPU upgrade
[16:28] <Compn> there is a broadcom crystal hd h264 pcie card, but thats decoder only
[16:28] <Compn> i cant remember any encoders
[16:28] Action: Compn goes afk
[16:28] <Mavrik> OneFix_Work, well, 3.3GHz 2500 quad is about 160¬ last I checked
[16:28] <Mavrik> that'll help more than anything else
[16:28] <Mavrik> since x264 doesn't support any of other hardware accelerators
[16:29] <OneFix_Work> Mavrik: This is my home Linux box...it does ZoneMinder, Subsonic, and Plex
[16:29] <Mavrik> yes?
[16:29] <OneFix_Work> Mavrik: And occasionally I will rip a DVD or edit photos in DarkTable...
[16:31] <OneFix_Work> I already have a fast FreeNAS box (quad core, 8GB RAM) for storage
[16:33] <Mavrik> fine, but how is that connected to the x264 problem?
[16:33] <OneFix_Work> Mavrik: Just saying that I've built a quad core system recently
[16:37] <OneFix_Work> Mavrik: Actually, for half the price of the 2500, you can get an AMD Quad-Core Phenom II X4 @3.6GHz
[16:42] <OneFix_Work> Mavrik: And I'm half thinking just trying to do transcoding using my FreeNAS box.
[16:44] <OneFix_Work> Another question I had, has anyone done a comparison of the libx264 presets?
[17:01] <OneFix_Work> I've seen the page that compares lossless, veryslow and veryfast ... but I haven't found anything that includes the newer ultra fast presetd
[17:10] <raven> hi
[17:10] <raven> ffmpeg/winff preset: how to limit mpeg4 super high quality preset to maximum 4mbit?
[17:14] <OneFix_Work> raven: -br 4000k
[17:16] <OneFix_Work> raven: Depneding on your version it might be just -b 4000k and you might need to add -bt 4000k
[17:17] <raven> OneFix_Work, i do not understand, why maxrate 3000k does not take effect - any suggestions?
[17:18] <OneFix_Work> raven: The -b or -br option is the average and -bt tells it how much it can change.
[17:19] <raven> OneFix_Work, not here - when i put anything into b it still runs above to double or more
[17:23] <calc84> hey, is it possible to change the sample rate in ffmpeg without resampling? I want to speed up the audio to double speed
[17:23] <calc84> I already got the video to double speed
[17:24] <gst-kaps> has anyone tried compiling ffmpeg.c on windows ?
[17:25] <calc84> I was thinking that using -async would make the sound go as fast as the video, but apparently not
[17:25] <raven> OneFix_Work, this is the winff preset for mpeg4 super high -crf 15.0 -vcodec libx264 -preset veryslow -acodec libfaac -ar 48000 -ab 192k -coder 1 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -me_method hex -subq 6 -me_range 16 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -threads 0
[17:27] <vcs> hi, I have a file with 4 video streams and 2 audio streams, is it possible using ffmpeg to split the files up into a file for each stream?
[17:27] <vcs> or using some other utlity that uses ffmpeg
[17:29] <alyawn> so the q=<num> on the output of ffmpeg, is that an average, or is it specific to the current frame
[17:38] <qxt> Was wondering if there was a way to import video uncompressed from a video camera that uses DAT. 
[17:40] <qxt> once edited Ill toss it into a mkv container and compress it with x264 using ffmpeg like usual.
[17:48] <kollapse> relaxed: Hello. You told me to run mediainfo on 2 files.
[17:48] <relaxed> yes, to see if it reveals any differences you may have missed
[17:50] <kollapse> relaxed: I see a difference. The camera file has "Writing application: CanonMVI06" and "Mastered date: Sun Aug 7 09:50:18 2011" while the ffmpeg file only has "Writing application: Lavf53.5.0"
[17:52] <teratorn> how would I use ffmpeg to copy the first frame of a video in to a new file?
[17:52] <teratorn> I just want a new video file with a single frame in it
[17:52] <teratorn> ah, -vframes
[17:54] <relaxed> kollapse: you could hexedit the video and replace Lavf53.5.0 with CanonMVI06.
[17:55] <calc84> so can I make audio double speed? (pitch increase and all)
[17:56] <kollapse> relaxed: Hmm, here's what I get - http://pastie.org/2986785 (could anything else in there be the cause?)
[17:59] <relaxed> kollapse: -b 16M
[18:01] <kollapse> relaxed: I've set that but ffmpeg still gives out a file with 5M bitrate
[18:01] <kollapse> 5626 kbps
[18:07] <qxt> How would I capture my dv1394 (firewire) video as raw video. I can do this in dvgrab but would like to use ffmpeg. This part I know "ffmpeg -f dv -i /dev/dv1394/0 "
[18:09] <qxt> or just uncompressed is a better terminology. 
[18:12] <Mista_D> Why encoding to 4 output files (1st of 2 passes) utilizes 2.5 CPU cores? No matter what the number of outputs, `-threads` or `-thread_type` are, the utilization is ~2.5 cores of 16 available. http://pastebin.com/m5ctKziU
[18:12] <calc84> I guess I'll be forced to use mencoder to speed up my videos :/
[18:15] <relaxed> Mista_D: before the input determines how many threads used for decoding.
[18:17] <cryptopsy> i'm having a hard time converting 400+ kb/s flac to 320kb/s mp3
[18:17] <cryptopsy> ffmpeg -i 111.flac -ab 320k -f mp3 test.mp3 produces 160kb/s bitrate mp3
[18:17] <relaxed> Mista_D: place "-threads 0" after the input and consider using a preset.
[18:17] <cryptopsy> and using -acodec copy produces a 400kb/s+ mp3 file that cannot be played
[18:18] <Mista_D> relaxed: I tried that already - no changes.
[18:18] <cryptopsy> i see [mp3 @ 0xc197e0] Unsupported sample rate.
[18:18] <cryptopsy> don't get why it's doing this
[18:18] <walisser> cryptopsy, what is the sample rate of the flac?, e.g. ffmpeg -i 111.flac
[18:18] <relaxed> cryptopsy: -aq 0 will give you highest quality vbr mp3.
[18:18] <cryptopsy> walisser: over 400 like i said
[18:19] <walisser> thats bitrate, not sample rate
[18:19] <cryptopsy> bitrate: 495 kb/s
[18:19] <cryptopsy> this is the only thing ffmpeg -i is showing
[18:20] <walisser> ok then specify the sample rate when transcoding, e.g. -ar 48k
[18:20] <cryptopsy> is the sample rate in hertz?
[18:20] <cryptopsy> 22050 Hz
[18:20] <walisser> yes
[18:20] <cryptopsy> what's the difference between the two?
[18:21] <walisser> bit rate is the compressed rate (streaming rate) of the file
[18:21] <walisser> sample rate is the playback rate of the audio, e.g. samples per second
[18:22] <cryptopsy> thanks
[18:37] <calc84> why DOESN'T ffmpeg support changing the sample rate without resampling?
[18:38] <Tjoppen> try -ar before -i
[20:34] <pasteeater> vcs: late reply, but the answer is "yes".
[20:34] <pasteeater> ffmpeg -i VTS_02_1.VOB -c copy -map 0:0 out0.mpg -c copy -map 0:9 out9.ac3 -c copy -map 0:11 out11.ac3
[20:34] <pasteeater> there might be a better way. i'm not totally used to the new map structure
[20:40] <vcs> hm ok, ill give it a shot thanks
[20:46] <Skaag> can ffmpeg be used somehow to output just specific information about a media file?
[20:46] <Skaag> such as its resolution?
[20:46] <walisser> skagg, probably... why not grep ffmpeg -i output
[20:47] <Skaag> I do that
[20:47] <Skaag> I was just wondering if it has something simpler
[20:48] <Skaag> I grep for "Stream" and "Video" first, then I locate (\d)x(\d)
[20:48] <walisser> in libavformat... when I call avformat_find_stream_info() how can I get back the frames/packets read during the probe?
[20:48] <Skaag> I just don't want it to break on me at some point
[20:49] <walisser> skaag, that is probably ok
[20:49] <Skaag> ok
[20:49] <walisser> you would have to use the C library (libavformat for example) to avoid that I think
[20:49] <Skaag> oh I see.
[20:50] <cbsrobot> Skaag: use ffprobe
[20:50] <cbsrobot> ffprobe -show_streams -output_format json input
[20:50] <cbsrobot> ^ might have some errors
[20:52] <Skaag> my version doesn't love the output_format option
[20:53] <walisser> the docs say "examined packets may be buffered for later processing", does that mean they may not be buffered as well?
[21:27] <stonie> hi
[21:27] <stonie> how may i encode in lame and pass options to not encode in joint stereo?
[21:28] <stonie> something like -lameopts ?
[21:30] <pasteeater> stonie: ffmpeg -i input -f wav - | lame -m s - output.mp3
[21:30] <stonie> theres also a videostream inside
[21:31] <pasteeater> and what do you want to do with this video stream?
[22:05] <vcs> pasteeater: any clue when the map stuff was changed?
[22:18] <alyawn> as ffmpeg runs, is the q= that's displayed an average or only the current frame?
[22:19] <cek> Hi. What video decoder supports FLV8 video? how is it named/
[22:19] <Skaag> It is named Zod, kneel before Zod!
[22:19] <cek> pSrZhb 
[22:19] <Skaag> just kidding, sorry, I don't know what it's called& :-)
[22:21] <cbsrobot> cek: ffmpeg -codecs
[22:22] <cbsrobot> check the vp* codecs
[22:22] <cek>  D V D  vp8             On2 VP8
[22:22] <cek> but it doesn't decode the swf i have, only getting audio
[22:22] <cek> and it used to work 2 years ago with old version
[22:23] <cbsrobot> pastebin the command + output
[22:23] <cek> hereyogo https://gist.github.com/92c4d4d00e5e3b18d11e
[22:26] <cbsrobot> well I only see audio
[22:26] <cbsrobot> but try with ffprobe
[22:28] <cbsrobot> but it doesnt look like the video is recognized 
[22:31] <cek> thats what im saying
[22:31] <cek> Hi. What video decoder supports FLV8 video? how is it named/
[22:31] <cek> the video plays in chrome
[22:35] <gfto> mediainfo reports for the audio "Delay relative to video: -146ms" Is there a way for this delay to be 0 (or at lease less that 40 ms) without recompressing? The log is at : http://pastebin.com/rAabQtJM   It is generated by ffmpeg but with avconv the result is the same.
[22:36] <gfto> ...and the sentense should start with : I've asked this a couple of days ago at ffmpeg-user but nobody seems to know, so here it is again.... I'm remuxing mpegts files into mpegps (-f vob) and the problem I'm having is that mediainfo ...
[22:57] <teratorn> anyone have a clue about this memory leak inside libavformat? valgrind output: http://codepad.org/wIgKypR6
[23:59] <gdoteof> can someone take a look here?  i am trying to turn this very large (680M) 3:24 clip exported from finalcut pro to something manageble
[23:59] <gdoteof> i am using some commands i got from a board http://pastebin.com/nMCzBg40
[23:59] <gdoteof> line one is the command with flags
[23:59] <gdoteof> the rest is the output from ffmpeg and the resulting errors
[00:00] --- Fri Dec  9 2011