[Ffmpeg-devel-irc] ffmpeg.log.20121128

[burek]

[00:14] <bakers> I have a video with two audio streams, and a subtitle track. I want to strip out everything except the video and the first audio track
[00:14] <bakers> I'm trying to do -map 0:0 -map 0:1 but I get errors
[00:14] <bakers> http://www.fpaste.org/H8zR/
[00:14] <bakers> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[00:15] <JEEB> libvo_aacenc doesn't support >2ch afaik, and is a crappy encoder
[00:15] <bakers> Oh the source is 5.1
[00:15] <bakers> I didn't even notice
[00:15] <JEEB> it seems to have picked the right tracks tho
[00:15] <bakers> what's the "best" aac encoder then
[00:15] <JEEB> fdk-aac is the best you can link to ffmpeg
[00:16] <JEEB> (too bad you can't distro it because of fraunhofer being dumb with their license)
[00:16] <JEEB> (they ended up with something very close to GPL, but then added a line about you not being able to take money from the piece of software, which GPL lets you do -> nonfree)
[00:16] <JEEB> but the input is already AAC it seems
[00:17] <JEEB> you could just do -c:a copy ?
[00:17] <bakers> true
[00:17] <bakers> ok
[00:17] <bakers> I didn't notice the 5.1
[00:17] <JEEB> heck, you could do c:v copy and c:a copy
[00:17] <JEEB> because you're not setting any settings :)
[00:17] <bakers> That was just a test to see if I mapping was correct
[00:17] <bakers> I need to make this smaller
[00:17] <JEEB> k
[00:17] <bakers> is -acodec aac
[00:17] <bakers> bad?
[00:17] <bakers> it's still "Experimental"
[00:18] <JEEB> yes, it's better than vo_aacenc but still needs at least ~192kbps or so
[00:18] <JEEB> better to just install fdk-aac if you need AAC
[00:18] <JEEB> another alternative is to build aotuv's vorbis
[00:18] <bakers> I'm running a "static" version that doesn't seem to have that
[00:19] <JEEB> well fdk-aac can't be redistributed
[00:19] <JEEB> as it's "nonfree"
[00:19] <bakers> k
[00:19] <JEEB> aotuv's vorbis can be, but I'm not sure if the guy who builds the static linux builds with aotuv or official
[00:20] <JEEB> official is still somewhat worse off than aotuv
[00:21] <bakers> This is x264 so that's also non-free
[00:21] <JEEB> no, x264 is GPL
[00:21] <bakers> oh ya?
[00:21] <JEEB> yes
[00:21] <JEEB> really
[00:22] <JEEB> you /can/ license x264 commercially if you wish, but by default it is GPL. Nonfree means something is not GPL-compatible.
[00:22] <JEEB> (or LGPL)
[00:26] <bakers> what's the status on libfaac?
[00:26] <JEEB> that was found to be using the reference source code
[00:27] <JEEB> -> nonfree
[00:27] <JEEB> same for libaacplus
[00:27] <bakers> are there any "free" aac encoders?
[00:27] <JEEB> vo_aacenc is based on other reference code, but properly licensed as Apache v2
[00:27] <JEEB> but it's crappier than the libavcodec ffaac one
[00:27] <JEEB> (aka -c:a aac)
[00:28] <JEEB> fdk-aac is actually is open source and properly licensed, but the license is not GPL compatible -> nonfree in ffmpeg
[00:28] <JEEB> which is a real shame because unlike the other nonfree things, it actually is properly licensed :<
[00:28] <bakers> What exactly does -strict experimental do
[00:29] <bakers> just says use these "not fully tested things"
[00:29] <JEEB> it just enables stuff that is labled "experimental"
[00:29] <JEEB> the internal AAC encoder is OK if you use bit rates of 192kbps+
[00:29] <bakers> and the -acodec aac is still experimental?
[00:29] <bakers> ah ok
[00:29] <JEEB> yes
[00:29] <bakers> got it
[00:29] <JEEB> it would be better if someone finished it :D
[00:29] <JEEB> too bad no-one did
[00:30] <JEEB> it currently has features turned off in the code because they just don't work
[00:30] <bakers> Awesome, thanks for the explanation
[00:30] <JEEB> the case of fdk-aac is really, really sad tho :<
[00:31] <JEEB> brought in by android just like vo_aacenc
[00:31] <JEEB> -> "lol we made the license not compatible with GPL"
[00:31] <JEEB> of course if you compile yourself that's not a problem
[00:31] <JEEB> or possibly if you only compile LGPL things, but I think ffmpeg doesn't even let you build it with fdk even in that case :)
[00:31] <JEEB> --enable-nonfree is global
[00:39] <bakers> What was brought in by android?
[00:39] <bakers> fdk-aac and what?
[00:40] <JEEB> vo_aacenc
[00:40] <chrisballinger> we are having trouble using the built in aac encoder because it expects floats
[00:41] <JEEB> well it isn't exactly good for low bitrate targets either
[00:41] <chrisballinger> we are going for 64kbps mono
[00:42] <JEEB> well, everything good is pretty much nonfree so you'll probably just have to talk on the -devel channel and see if you can get someone looking over you making a non-float version of/fixing the internal
[00:42] <JEEB> unless you don't distribute any binaries :)
[00:43] <chrisballinger> :(
[00:43] <chrisballinger> it used to accept floats
[00:43] <chrisballinger> er i mean ints
[00:43] <chrisballinger> 16-bit
[00:43] <JEEB> you should be able to convert within ffmpeg's libraries
[00:44] <JEEB> I thought you meant you can't have float math
[00:44] <JEEB> lol
[00:44] <JEEB> libavresample or libswresample
[00:44] <JEEB> one of these
[00:44] <chrisballinger> we are getting a crash using swr_convert
[00:44] <ubitux> how to reproduce?
[00:44] <JEEB> get a backtrace and poke -devel / the trac bug tracker
[00:45] <chrisballinger> this is JNI so it's hard to debug / get a trace
[00:45] <bakers> man that "aac" encoder is crap...
[00:45] <bakers> even at 128k
[00:45] <JEEB> yes
[00:45] <JEEB> which is why I said 192k
[00:45] <JEEB> +
[00:45] <bakers> I know I know
[00:45] <bakers> This is only for testing
[00:45] <bakers> and even for testing it's crap
[00:46] <chrisballinger> why can't we distribute binaries that use fdk-aac again?
[00:46] <JEEB> because it's not GPL-compatible
[00:46] <chrisballinger> not even LGPL compatible?
[00:46] <bakers> any chance on getting that changed?
[00:46] <JEEB> it could be LGPL compatible, but the nonfree limitation is global
[00:46] <JEEB> so derp
[00:47] <JEEB> bakers, find someone who wants to read the spec and is interested in psychoacoustics
[00:47] <chrisballinger> That seems rather silly
[00:47] <JEEB> well ask someone related to the project about that...
[00:48] <JEEB> might either get an answer that it's not OK, or that it's too hard to implement in the current configure way, or that you'll just get it within some hours
[00:48] <JEEB> bakers, and someone who wants to work on the encoder
[00:48] <JEEB> it's bad because a lot of features are turned off as they don't work f.ex.
[00:48] <JEEB> (it's better than the WMA2 encoder tho, lol)
[00:48] <JEEB> also did you downmix to 2ch?
[00:49] <JEEB> 128k and 5.1 is WHOOPS
[00:49] <bakers> libvo_aacenc is decent
[00:49] <bakers> just can't convert 5.1 to 2.0
[00:50] <JEEB> ...so you didn't even downmix when testing the internal one?
[00:50] <JEEB> because IIRC in a Japanese test the internal one was pretty much on the level of the vo_aacenc with 2ch
[00:50] <JEEB> not sure what bit rate it was, but it really sounds that you didn't downmix
[00:50] <bakers> how do I tell it to downmix?
[00:50] <JEEB> ...
[00:51] <bakers> I probably didn't
[00:51] <bakers> -ac 2?
[00:51] <bakers> I'll try that
[00:52] <JEEB> yeah, that is the old way IIRC, not sure if there's a new way to call the swresample based audio thingy
[00:52] <JEEB> the internals probably work on the same swresample base
[00:52] <bakers> Or I could just a clip that's not 5.1 to start with :)
[00:53] <JEEB> anyways, yeah -- -ac 2 makes sure you only have 2 channels in the output
[00:53] <JEEB> yeah, seems like the correct way still :)
[00:57] <bakers> Oh hey I meant to ask
[00:57] <bakers> is ffprobe -print_format json supposed to output anything
[00:57] <JEEB> never used that, so no idea :3
[00:57] <ubitux> along with -show_streams/-show_format/etc yes
[00:58] <joecool> seems like ffmpeg in the past 2 days broke spek, any major changes that would cause it to throw "unsupported file format"?
[00:58] <ubitux> what's spek?
[00:58] <bakers> ubitux: when I run that I get the standard ffmpeg output to stderr and { } to stdout
[00:58] <joecool> spectral analyser
[00:58] <ubitux> bakers: what command?
[00:58] <bakers> ubitux: http://www.fpaste.org/ze77/
[00:59] <ubitux> 00:57:55 <@ubitux> along with -show_streams/-show_format/etc yes
[00:59] <bakers> OH!
[00:59] <bakers> Got it
[00:59] <bakers> I expected there to be SOME default
[00:59] <ubitux> joecool: i don't know what you are talking about
[01:00] <ubitux> please provide a sample/command line/...
[01:05] <joecool> ubitux: it's graphical, https://xwaretech.info/spek.jpg
[01:05] <joecool> normally that would generate a spectrogram, it doesn't
[01:06] <llogan> JEEB: were you referring to this? http://d.hatena.ne.jp/kamedo2/20120729
[01:06] <llogan> i never did try with -cutoff
[01:06] <JEEB> llogan, yeah
[01:08] <ubitux> joecool: i don't know how that work
[01:09] <ubitux> it can be a badly compiled ffmpeg, using libs or the tool, can even not be ffmpeg, etc
[01:10] <joecool> ubitux: it does not bundle ffmpeg, it was working fine until about 2 days ago, i build ffmpeg nightly
[01:11] <JEEB> if you build it, you could start going backwards in revisions
[01:11] <JEEB> and see which one was the one that borked something for you
[01:11] <joecool> i can bisect it, but it takes time
[01:11] <joecool> i was hoping someone would say "oh yeah we did this, they need to fix their shit" or something to that effect
[01:11] <ubitux> use ccache, build only the minimum, bisect with 2 days
[01:11] <ubitux> should be pretty fast
[01:12] <JEEB> yup, but since no-one else has an idea how to replicate it or what's the problem, you can't really get an answer like that :P
[01:12] <JEEB> so bisect it is
[01:12] <ubitux> if you can reproduce with ffmpeg directly that could help
[01:15] <funyun> hi. can ffmpeg repair a corrupt .mov file?
[01:16] <JEEB> if you have no index in it, nope.avi
[01:17] <funyun> JEEB: how can i check?
[01:18] <JEEB> if you do ffmpeg -i derp.mov and it tells you it can't find something, that means it can't be automatically opened with a generic demuxer
[01:19] <funyun> JEEB: http://pastebin.com/0ir23Mez
[01:20] <JEEB> well, at least it seems to have an index
[01:20] <JEEB> also man that is old
[01:20] <JEEB> build a newer one
[01:20] <funyun> JEEB: you mean ffmpeg?
[01:20] <JEEB> also when did clang get to 4.0 o_O
[01:20] <JEEB> funyun, yes
[01:21] <Plorkyeran> apple clang
[01:21] <JEEB> oh
[01:21] <Plorkyeran> completely different set of version numbers from normal clang
[01:21] <JEEB> they have a separate versioning system?
[01:21] <JEEB> ah
[01:21] <JEEB> funyun, ok your problem wasn't the one most people have with mov/mp4 etc. files
[01:21] <Plorkyeran> there was a really confusing point where they almost but didn't quite line up
[01:21] <JEEB> what's the problem, or is it that other timestamp funnyness?
[01:22] <funyun> JEEB: it plays fine in quicktime. but it won't play in vlc or mplayer and most importantly, i can't edit the video in final cut pro
[01:23] <JEEB> first build current ffmpeg from the git repo, secondly try ffmpeg -i input.mov -map 0 -c:a copy -c:v copy out.mp4 or so
[01:23] <JEEB> and see if it can copy the streams
[01:23] <JEEB> although that might or might not affect this if this is the problem "Seems stream 0 codec frame rate differs from container frame rate: 1200.00 (1200/1) -> 24.00 (24/1)"
[01:24] <JEEB> (the video track and the container have different maximum timescales)
[01:24] <JEEB> aanyways, I really need sleep now o/.
[01:24] <funyun> alright. thanks anyway
[01:26] <cbsrobot> funyun: it could also be the audio ...
[01:26] <funyun> cbsrobot: in fcp, the video is corrupted
[01:26] <funyun> i get a green screen
[01:26] <cbsrobot> ah ok
[01:27] <funyun> even tho when i play that same time with quicktime, it plays fine
[01:28] <funyun> a little back story if it might help. i recorded this video on my iphone and it recorded until the phone ran out of battery
[01:28] <funyun> so maybe it just didn't save correctly?
[01:30] <cbsrobot> funyun: maybe it wont play it because of vvs
[01:30] <funyun> vvs?
[01:30] <cbsrobot> http://www.youtube.com/watch?v=Bt9zSfinwFA
[01:32] <funyun> cbsrobot: lol
[01:32] <funyun> i hate vvs as well
[01:32] <funyun> so that's not the problem
[01:33] <cbsrobot> ok - lucky you
[01:33] <cbsrobot> try what jeeb said
[02:16] <joules> hi, what is the ffmpeg equivalent to mp=mcdeint:0:0:10 and mp=pp:hb/vb/dr ?
[02:18] <joules> theres no mcdeint so i can forget about that...
[03:06] <joules> hi, i'm trying to seek to a position in this vob but only -sb works -ss gives me EOF @ 100secs
[03:08] <joules> damn -ss is not working for me. any suggestions?
[04:30] <joecool> ubitux: working on bisecting it now, i'm at this commit and it works https://github.com/FFmpeg/FFmpeg/commit/a13148f63351db5f8283a23de9a22e940d29d8cd
[05:31] <joecool> ok i think i got it
[05:32] <joecool> https://github.com/FFmpeg/FFmpeg/commit/59b68ee8870ad7b831321ff39638660ac59e4a18 << this commit breaks spek (just won't show graphs at all, claims unsupported format on anything)
[05:32] <joecool> https://github.com/FFmpeg/FFmpeg/commit/a13148f63351db5f8283a23de9a22e940d29d8cd << this commit works just fine, last working commit
[05:40] <joecool> actually that commit half-works, still going back to see when it 100% works
[05:55] <joecool> https://github.com/FFmpeg/FFmpeg/commit/86270236d5ef97263858a576fab2c366fb2aa76a << fully works on this
[07:38] <JEEB> joecool, if it starts failing on the commit where certain decoders moved to planar audio input (multiple pointers and multiple buffers instead of everything being interleaved into a single buffer) then whatever is utilizing it probably still tries to get interleaved?
[07:39] <JEEB> which it can't
[07:39] <JEEB> if you use the libraries you would have to use libswresample or libavresample to interleave it to the format you need
[07:40] <joecool> JEEB: dunno enough about it, it generates visual spectrographs, not sure how it does it
[07:41] <JEEB> it probably assumes that everything outputs interleaved
[07:41] <JEEB> which is how it was before that change
[07:41] <JEEB> now I think all audio decoders output planar
[07:42] <JEEB> and if you don't like planar audio, you have to interleave it with, say, libswresample or libavresample
[07:42] <JEEB> that's it
[07:43] <JEEB> and "half-works" probably means that some formats were planar, some interleaved at that point
[07:44] <JEEB> so you're pretty much getting <joecool> i was hoping someone would say "oh yeah we did this, they need to fix their shit" or something to that effect
[07:45] <joecool> JEEB: cool, i'll see if i can get in contact with the developer
[07:45] <joecool> thanks
[07:45] <JEEB> np
[10:39] <bizulk> Hi ! I would like to port my app on the new ffmpeg API. But I do not understand the options params with avcodec_open2. May I put NULL for options, if no how can I fill it before calling options.
[10:42] <JEEB> http://ffmpeg.org/doxygen/trunk/group__lavc__core.html#g11f785a188d7d9df71621001465b0f1d
[10:43] <JEEB> even has an example
[10:43] <JEEB> :)
[10:46] <bizulk> Yes thanks. But on this exemple we set arbitrary options. But if there a not particular options ?
[10:46] <JEEB> the decoding example just passes NULL to that just fine
[10:46] <JEEB> it seems
[10:46] <JEEB> http://ffmpeg.org/doxygen/trunk/doc_2examples_2decoding__encoding_8c-example.html
[10:49] <bizulk> that's what I expected. I shall take some time to read these relevant short example. ffplay confuses me.
[10:50] <bizulk> JEEB: thks
[10:50] <JEEB> ffplay is a "barely working example" :) for example, it fails if the video is not 4:2:0 YCbCr
[11:13] <Mavrik> hmm, why does filtering with avfilter change PTS of my frames?
[12:32] <JEEBsv> Mavrik: you might want to detail on that some more :D
[12:32] <JEEBsv> which filter, how etc.
[12:32] <Mavrik> yeaah, basically I'm doing yadif,scale on video and aresample,aconvert on audio
[12:33] <Mavrik> and I'm seeing the PTS change on both accounts to... something
[12:33] <Mavrik> and I can't seem to find any documentation/mention of filters adjusting PTS of frames
[12:33] <Mavrik> which (obviously) causes me some sync problems
[12:34] <Mavrik> I can gladly provide more information... but I have to know what :D
[12:34] <JEEBsv> well, yadif can change the pts naturally, as it can bob
[12:34] <JEEBsv> scale shouldn't
[12:34] <JEEBsv> audio naturally could change PTS with resampling
[12:34] <Mavrik> mhm
[12:34] <Mavrik> [2012-11-28 12:34:28,735][TRACE][../src/transcode/streams/VideoStreamProcessor.cpp:219] [VideoStreamProcessor] - Frame [720x576] Timestamp 7779355407
[12:35] <Mavrik> [2012-11-28 12:34:28,737][TRACE][../src/transcode/streams/VideoStreamProcessor.cpp:237] [VideoStreamProcessor] - [F] Frame [640x360] Timestamp 15558703614
[12:35] <Mavrik> this is what happens - the change is pretty big (not just adjustment for deinterlace, resample)
[12:35] <JEEBsv> make sure you give out the timescale too
[12:35] <Mavrik> a.
[12:35] <JEEBsv> PTS is a x/y value after all
[12:35] <Mavrik> I only set time_base on the buffersrc, not src
[12:35] <Mavrik> *not sink
[12:36] <Mavrik> this is how I setup the filter chain: http://pastebin.com/dxVPe3j1
[12:36] <JEEBsv> I have absolutely no idea what you're doing and what you're not doing, but I would check what timescales you are getting on all sides, there are also functions to then match PTS to a certain timescale
[12:37] <Mavrik> JEEBsv: as far as I know the frame object have no timescale property attached, only PTS. Where can I check that?
[12:38] <Mavrik> as I said, I have no idea how timing is handled within libavfilter and I can't seem to find any documentation on that
[12:39] <JEEBsv> and I have only used libavcodec from the inside, aka implemented an encoder. I only know limited amount of stuff on the outside APIs
[12:40] <JEEBsv> all I can say is check your timescale and that you're handling both the X and the Y of your PTS correctly
[12:40] <Mavrik> mhm, I'll go bug on libav-users or something
[12:40] <JEEBsv> and not taking an X value and putting it into another timescale
[12:40] <JEEBsv> which naturally will end up with a/v desynch
[12:41] <Mavrik> JEEBsv: yeah, I'm handling those correctly outside the filter chain, but as I said, my PTS gets chained while frames are being filtered and I can't seem to find the timebase to which it gets converted
[12:41] <JEEBsv> because that's what it sounds like
[12:41] <Mavrik> *changed
[12:41] <Mavrik> anyway, thanks :)
[12:41] <Mavrik> I'm gonna continue to go through the source to find out what the heck is going on
[12:50] <Mavrik> ok got it, I need to rescale timebase from output buffer sink filter's timebase to the codec's timebase
[13:30] <brontosaurusrex> any clues what quantel is using as a master codec?
[13:30] <brontosaurusrex> is that some i-frames type of system?
[13:30] <brontosaurusrex> i know, iam offtopic
[14:03] <samek_> Hi, can anyone help me debug why  rtmp streaming is not stable ? It stalls after sometime or it issues FCunpublish. Source of the video is Live rtp multicats.
[14:33] <samek_> Is it possible to debug only output RTPM packet communication ?
[14:34] <samek_> I have a problem with streaming from rtp multicast source to rtmp. ffmpeg connection normaly just hangs or it issues FCunpublish. I have a log with debug but's it's 400+Mb
[16:05] <samek_> Hi, can anyone help me debug rtmp publishing from an rtp multicast source ?
[16:06] <samek_> It stops after a a certain time (ffmpeg "hangs")
[16:58] <^BB^> question: i have an MKV file with a video and audio stream. the audio stream must have a delay, because when i demux the streams and remake an MKV without audio delay, the audio is out of synch (it runs about 3 seconds behind the video). so, how do i found out the audio delay? ffmpeg -i file.mkv doesn't show any delay.. :s
[16:59] <^BB^> mediainfo will say it's at 1mn 5sc, which to me sounds like a 655xx value (signed bit problem?)
[17:00] <^BB^> however, any media player will play it correctly, so there must be another way to obain the correct delay value
[17:02] <JEEB> <Mavrik> ok got it, I need to rescale timebase from output buffer sink filter's timebase to the codec's timebase <- called it :)
[17:02] <Mavrik> JEEB: yeah :D
[17:02] <Mavrik> had to read through ffplay.c to find out where time_base is stored
[18:46] <samek__> anyone here ever had ffmpeg hand during rtmp streaming ?
[18:47] <samek__> hand = hang
[18:51] <Laputa> hi, i'm wondering if ffmpeg support mp4 container ? i wanna wrap h.264 video and aac audio to a mp4 container.
[18:51] <sacarasc> Laputa: Yes.
[18:54] <Laputa> sacarasc: any url for that ? i converted some video to mp4, but the flash player could read the video's info untill the video is all loaded.
[18:54] <Laputa> sacarasc: i used to use mp4box to wrap the video again, i wonder if i can do this in ffmpeg
[18:55] <sacarasc> You'd have to run qtfaststart on the video, which comes with the ffmpeg source.
[19:03] <samek__> could someone suggest me on how to debug the rtmp session which hangs when streaming to FMS server. http://pastebin.com/jXzExPit
[19:25] <Laputa> sacarasc: thanks . i've tried that , and works great !
[20:06] <PMacDiggity> Hello, how can I identify in MacOS 10.8 what the correct name for an input device is for video capture (Thunderbolt BlackMagic Deign UltraStudio Express) for use with the '-i' flag?
[21:49] <zuloyd> hihi
[21:49] <zuloyd> (just once, sorry)
[21:50] <zuloyd> can someone please tell me what's the problem here? http://pastebin.com/Xfz7PPwV
[21:50] <zuloyd> i'm trying to generate a thumbnail from a video
[21:51] <zuloyd> is it because it's too short?
[21:54] <llogan> zuloyd: input duration is 3.9 seconds. your -ss is 4 seconds.
[21:54] <zuloyd> is it possible to extract a random frame?
[21:55] <zuloyd> or at least something from the middle of the film
[21:55] <zuloyd> i don't want it to be the first frame in every movie
[21:55] <zuloyd> that is, without knowing the duration
[22:02] <llogan> zuloyd: ignoring any scripting you might be able to also do it with the select filter, but i don't know
[22:02] <llogan> however your ffmpeg is too old
[22:03] <zuloyd> I know, but I have no chance of updating it since it's a managed server
[22:03] <zuloyd> and the hoster is very conservative
[22:03] <llogan> you can use a static build
[22:03] <llogan> http://ffmpeg.org/download.html#LinuxBuilds
[22:04] <zuloyd> okay, a related but different questions: Can I somehow use static libraries?
[22:04] <zuloyd> i.e. codecs
[22:04] <llogan> sorry, i don't understand the question
[22:04] <zuloyd> I need a video codec that's not installed on the system and I can't install it since I don't have root access
[22:05] <zuloyd> but is there a way to get this codec from somewhere, perhaps place it in the same directory as the static ffmpeg build and then use it?
[22:05] <llogan> use it with ffmpeg?
[22:05] <zuloyd> yes
[22:05] <llogan> it doesn't work like that
[22:06] <llogan> what format are you trying to encode or decode that you can't now?
[22:06] <zuloyd> trying to encode WebM
[22:06] <llogan> (better links for static builds)
[22:08] <llogan> the static builds both support libvorbis and libvpx which are the audio and video formats used in webm container
[22:12] Action: llogan needs to make a non root/su version of the ubuntu compile guide
[22:16] <llogan> ...and i should say "encoders" instead of "formats"
[22:41] <giany> hello
[22:42] <giany> anyone is able to play properly mp4->webm files  on firefox browser?
[22:51] <bakers> Is there  way to make ffprobe NOT output the compile information to stderr?
[22:52] <saste> bakers, 2&> /dev/null?
[22:52] <bakers> saste: Other than that :)
[22:52] <bakers> I think -loglevel quiet might work
[22:52] <saste> and does it work?
[22:57] <bakers> Testing now
[22:57] <bakers> Does apple trailers use ffmpeg to encode?
[22:57] <bakers> I see lavf in the encoding header section, isn't that ffmpeg
[22:59] <sacarasc> I would presume they use the QuickTime encoders...
[23:03] <bakers> If I'm joe-bob-tube... some small youtube startup, and I want to use (legally) ffmpeg to encode x264 and aac audio what would I need to license
[23:03] <bakers> and from whom?
[23:06] <burek> bakers, license applies to (re)distribution afaik
[23:06] <burek> not usage
[23:06] <bakers> so I can encode all I want
[23:06] <bakers> but if I have actual visitors is when you have to pay
[23:06] <burek> as long as nobody arrests you :)
[23:07] <burek> hm
[23:07] <burek> not that
[23:07] <burek> http://ffmpeg.org/legal.html
[23:07] <bakers> who wrote libx264
[23:07] <bakers> where did that come from?
[23:08] <burek> videolan
[23:08] <bakers> the VLC guys?
[23:09] <bakers> libx264 is a pretty solid encoder I'm impressed... considering we can't get a solid GPL compatible AAC encoder
[23:13] <samek__> could someone suggest me on how to debug the rtmp session which hangs when streaming to FMS server. http://pastebin.com/jXzExPit
[23:13] <samek__> I tried even with static build posted here on this channel moments ago .. it hangs.
[23:17] <burek> why do you use -re with live input
[23:18] <samek__> burek i've tried everything until now ..
[23:18] <samek__> with and without -re
[23:18] <samek__> I can create another without -re it will hang
[23:19] <burek> your "Stream mapping:" says you have 3 video + 3 audio encodings for the live stream
[23:19] <burek> are you sure your cpu/ram can handle all that in real time?
[23:20] <samek__> It should
[23:21] <samek__> dual Intel(R) Xeon(R) CPU           X5550  @ 2.67GHz
[23:21] <burek> did you check your cpu usage during your tests
[23:21] <burek> also -threads 0 is advised
[23:22] <samek__> i'm running it right now
[23:23] <samek__> top shows .. Cpu(s):  9.2%us,  0.3%sy, 36.2%ni, 54.0%id,  0.0%wa,  0.0%hi,  0.3%si,  0.0%st
[23:23] <burek> next thing, try removing rtmp outputs and use file outputs
[23:24] <burek> to see if it is the encoder(s) issue
[23:24] <samek__> ok I'll do it.
[23:27] <samek__> What if it's working on files ? how can we debug the rtmp session ?
[23:27] <burek> if it does
[23:28] <burek> then either rtmp muxer is buggy
[23:28] <burek> or your network is unstable
[23:28] <burek> or somebody cursed you :D
[23:29] <samek__> Since i belive i've tried with recorded file couple of days ago and it didn't go off on the same timestamp (when sending to rtmp)
[23:30] <samek__> so it's the other way arround what I'm doing right now not streaming to rtmp
[23:32] <samek__> It's still writing to files .. but than again somethimes it streams also for several minutes.
[23:36] <samek__> burek may i ask where are you from cause of the nick - It's a food.
[23:36] <samek__> I'm from slovenia btw.
[23:42] <burek> serbia :)
[23:42] <samek__> Cool
[23:42] <burek> :beer: :)
[23:44] <samek__> burek i guess it's not the encoder since it's still running. How could  i home in where the problem is network or muxer then ?
[23:45] <burek> hmh... try another test.. save your input in a file (1-2 minutes long) and then use that file instead of live input and use your rtmp outputs
[23:46] <burek> (don't forget -re)
[23:46] <burek> to see if the sync between input/outputs causes issues
[23:46] <samek__> Ok
[00:00] --- Thu Nov 29 2012