[Ffmpeg-devel-irc] ffmpeg.log.20121205

[burek]

[00:01] <llogan> teratorn: what do you mean by "synthesizes"?
[00:01] <teratorn> llogan: creates audio samples in code
[00:03] <Nick_S> can i put webm in ps container?
[00:04] <llogan> webm is a container
[00:05] <llogan> do you mean VP8 video?
[00:05] <norbert_> I asked in #linux if anyone knows a Linux distro that comes with an ffmpeg that has x.264 and the answer someone gave me is "None."
[00:05] <saste> teratorn, what's wrong with doc/examples/muxing.c ?
[00:05] <norbert_> if this is true, why is there no Linux distro with an ffmpeg that has x.264 support
[00:05] <norbert_> is that a licensing issue?
[00:05] <llogan> it is not true.
[00:06] <teratorn> saste: oh, did I miss the one I needed to not miss? :(
[00:06] <norbert_> I think it would be nice to have a distro that has a big button "Install x.264 everywhere"
[00:06] <norbert_> instead of wasting hours and hours just to get support for it in crazy nerdy ways
[00:06] <brx_> it has me stumped llogan, have been tinkering all day with this
[00:07] <llogan> norbert_: for example arch linux has ffmpeg that supports encoding with libx264. is that what you mean?
[00:07] <norbert_> llogan: yes, that "-vcodec libx264" works
[00:08] <norbert_> arch linux huh, hum... I may switch to that one then
[00:08] <llogan> what are you using now?
[00:08] <saste> teratorn, also check the resampling_audio example in the ffmpeg-devel archive, will be possibly pushed soon
[00:08] <norbert_> Debian stable
[00:08] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuideHardy
[00:08] <llogan> should get you most of the way there
[00:09] <teratorn> saste: thanks!
[00:09] <codecowboy> Hi. I need to extract two audio streams from a mov file to one file but I always just get one channel if I do it with just input and output file params. Can anyone tell me what I'm doing wrong? Its a Skype call which has been recorded to .mov without video. Command line output is here - https://gist.github.com/d9e380986c5431e6d7a8 - OSX Mountain Lion
[00:09] <norbert_> llogan: unfortunately, that guide didn't work at all
[00:09] <llogan> how, specifically?
[00:09] <norbert_> llogan: even with debian-multimedia in apt sources lots of important stuff is missing
[00:10] <norbert_> tried the Ubuntu guide and tried apt-getting stuff, but I apt tells me that packages "libfaac-dev", "libgpac-dev" and "libmp3lame-dev" are unavailable
[00:10] <codecowboy> the file seems to contain two stereo audio streams which I need to extract and bounce down to one audio file. mp3 preferred
[00:10] <norbert_> and then more problems, and so on
[00:10] <llogan> norbert_: ask yourself if you even need something before attempting to install it
[00:11] <llogan> "libgpac-dev" is useless unless you want to use x264 directly and have it output to mp4 container
[00:11] <norbert_> that's what I want
[00:11] <norbert_> actually, what I want is to record (a section of) my desktop
[00:11] <norbert_> I always used this for that:
[00:11] <norbert_> ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 25 -s 100x100 -i :0.0+100,100 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 output.mkv
[00:11] <norbert_> and that worked fine, but now libx264 no longer works
[00:12] <teratorn> saste: hmm well muxing.c can't encode to mkv or webm :/
[00:12] <teratorn> saste: sample format s16 not supported :(
[00:12] <teratorn> which is true - webm/vorbis requires FLTP format samples
[00:12] <llogan> norbert_: you're everywhere yet nowhere. you should have stated that from the beginning, and explain what you mean by "no longer works".
[00:14] <norbert_> it doesn't know the libx264 codec
[00:14] <saste> teratorn, that needs conversion
[00:14] <codecowboy> anyone ?? bueller ??
[00:14] <saste> feel free to post a patch, or file a feature request so you can track it
[00:15] <klaxa> codecowboy: use ffmpeg to extract the audio tracks, then use sox to mix them i guess?
[00:15] <codecowboy> klaxa, sox?
[00:16] <codecowboy> i am wearing socks if that helps :-/
[00:16] <klaxa> 3 steps, step 1: extract first audio track, step 2: extract second audio track, step 3: mix both tracks with sox
[00:16] <klaxa> http://sox.sourceforge.net/
[00:16] <llogan> you don't need sox
[00:16] <klaxa> you don't?
[00:16] <codecowboy> no one liner with ffmpeg then?
[00:16] <klaxa> ffmpeg does mixing?
[00:16] <klaxa> oh audio filters?
[00:19] <norbert_> llogan / cbsrobot: anyways, thanks for giving suggestions, I think I may have found a solution
[00:23] <codecowboy> llogan ??
[00:31] <cbsrobot> codecowboy: 2 stereo -> 1 stereo ?
[00:32] <codecowboy> cbsrobot yes thats what i need i think, just don't know how&.
[00:33] <cbsrobot> well try first: stream 0 to an mp3
[00:33] <cbsrobot> a
[00:33] <cbsrobot> nd then stream 1 to an mp3
[00:35] <cbsrobot> or why not copy them to an m4a ?
[00:35] <codecowboy> thats what klaxa was suggesting i think but llogan seemed to think it could be done without three operations
[00:36] <codecowboy> cbsrobot don't mind doing that as long as it will play on an android phone
[00:36] <cbsrobot> well are you sure on stream 1 (the 2nd) is any sound ?
[00:37] <teratorn> saste: I think I may post a patch for muxing.c - it just infinitely loops until you ^C it - then aborts before writing the trailer :(
[00:37] <codecowboy> cbsrobot must be as i get only get one side of the conversation (its a Skype call) if i try a simpler ffmpeg conversion
[00:38] <saste> teratorn, what command are you running?
[00:38] <teratorn> saste: hmm?
[00:39] <codecowboy> cbsrobot m4a = same problem
[00:39] <saste> teratorn, what output are you requesting?
[00:39] <teratorn> saste mpeg2
[00:39] <saste> teratorn, what's the name of the output file?
[00:40] <teratorn> saste: foo.mpeg2
[00:40] <codecowboy> cbsrobot is this syntax correct? ffmpeg -i 2012-11-24.mov -map 0:0 -map 0:1 -c copy test.m4a
[00:40] <cbsrobot> well I'm not sure an m4a can hold multiple streams
[00:40] <saste> teratorn,  VBV buffer size not set, muxing may fail ?
[00:41] <cbsrobot> what you want is to mix them together
[00:41] <teratorn> saste: is that a warning you see? I don't see that
[00:41] <teratorn> oh
[00:41] <codecowboy> cbsrobot indeed i do, do you know how perchance?
[00:41] <teratorn> I do see it sorry
[00:41] <teratorn> saste: yeah - i have no idea what format/container it actually works with
[00:41] <cbsrobot> codecowboy: did you read the manual ?
[00:42] <codecowboy> cvs yeah like i have time for that, thats why i'm on irc
[00:42] <teratorn> i wanted to use mkv, but it fails because the default audio codec won't work with s16 samples - which is another reason to work on a patch :)
[00:42] <saste> teratorn, it is mpeg format, mpeg2video, mp2 audio
[00:42] <saste> and yes i confirmed that's buggy with mpeg
[00:42] <saste> i'll have a look at it soon
[00:42] <teratorn> cool :) the examples have always needed a lot of TLC
[00:43] <saste> teratorn, TLC?
[00:43] <codecowboy> cbsrobot i googled about a bit, went down a few dead ends, ended up here
[00:43] <cbsrobot> hehe
[00:43] <teratorn> tender love 'n care
[00:43] <codecowboy> cbsrobot lost will to live along the way
[00:44] <codecowboy> cbsrobot i just really need to know if its even possible to do it with ffmpeg, whether i can do it with -map args or if i need to do sth else
[00:44] <teratorn> saste: but somehow - despite all warnings - mplayer actually plays the resulting .mpeg file
[00:44] <codecowboy> i'd rather not read the whole manual to find out that it can't be done
[00:45] <cbsrobot> hehe
[00:45] <cbsrobot> I geuss the fastest is to extract the streams and amix them together
[00:48] <cbsrobot> try: ffmpeg -i 2012-11-24.mov -map 0:0 -c pcm_s16le test.wav && ffmpeg -i 2012-11-24.mov -map 0:1 -c pcm_s16le test2.wav && ffmpeg -i test.wav -i test2.wav -filter_complex amix -libmp3lame -b:a 64k output.mp3
[00:48] <saste> teratorn, takes a while, but the output is playable by ffplay
[00:48] <saste> anyway the buffer overflow spam is fishy
[00:52] <teratorn> hah! now my sound generator works perfectly using an 'ogg' container :(((
[00:52] <teratorn> and now it works with webm container too :(
[00:57] <llogan> codecowboy: ffmpeg -i 2012-11-24.mov -filter_complex "[0:0] [0:1] amerge" -ac 2 -c:a libmp3lame -q:a 4 out.mp3
[00:57] <llogan> untested
[00:59] <codecowboy> llogan thanks. segfaulted :(
[01:00] <codecowboy> but thanks for taking the time. will try again tomorrow
[01:01] <cbsrobot> did he even try my command line ?
[01:01] <cbsrobot> well nevermind
[01:01] <llogan> i was also going to suggest a recent static build. http://www.evermeet.cx/ffmpeg/snapshots/
[01:02] <llogan> brx_: did it work?
[01:09] <TheJK> In getting the images out of a live stream -- how do I force it to write to the same filename?  (Preferably write it to file.jpg.new, and then rename it as file.jpg) --> I currently have: ffmpeg -i 'udp://x.x.x.x@y.y.y.y:zzzzz?reuse=1' -r 1/2 -f image2 some%3d.jpg
[01:09] <iksik2> -y
[01:10] <cbsrobot> llogan: isnt amerge more for mono + mono ->  stereo ?
[01:10] <cbsrobot> or similar
[01:10] <TheJK> thanks iksik2
[01:14] <llogan> cbsrobot: i think it's more flexible than that
[01:20] <llogan> iksik2: you never included your console output as asked by fflogger bot
[01:30] <TheJK> @iksik2 -- tried the "-y" to override the same "jpeg" output... but getting "Could not get frame filename from pattern" -- thoughts?  I had this: ffmpeg -i 'udp://x.x.x.x@y.y.y.y:zzzz?reuse=1' -r 1/2 -y -f image2 some.jpg
[01:39] <smj> why does input '-f alsa -i pulse' even though PulseAudio's internal samplerate is different?
[01:39] <smj> why does input '-f alsa -i pulse' have sample rate of 48000Hz even though PulseAudio's internal samplerate is different?
[02:54] <ekristen> i keep getting "requested output format arc is not a suitable output format", I'm configured with --enable-libfaac, can anyonee help?
[03:23] <Alix___> Hello. I'm trying to convert a .mov (h.264/AAC) to a .swf. So i'm trying this : ffmpeg -i in.mov -acodec mp3 -ar 44100 -vcodec flv -f swf out3.swf
[03:23] <Alix___> But there is no sound when I play it from the browser
[03:24] <Alix___> the sound is working with vlc :/
[03:24] <Alix___> I don't see what i am missing
[03:46] <juanmabc> browser does not support mp3?
[03:47] <Alix___> It's chrome and it's working well on a simple .mp3
[03:47] <Alix___> and i tyed on different computer
[03:49] <sacarasc> Can MP3 go in SWF?
[03:49] <Alix___> http://helpx.adobe.com/flash/kb/supported-codecs-flash-player.html
[03:49] <Alix___> yes i think
[06:27] <escortkeel> hello! I'm trying to stream rtsp at the moment, but when I run ffmpeg like this:
[06:27] <escortkeel> ffmpeg -i movie.mp4 -f rtsp -an rtsp://localhost:554/yay.sdp
[06:28] <escortkeel> i get this (last line):
[06:28] <escortkeel> Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input
[06:28] <escortkeel> would anyone have any ideas? thanks in advance :D
[08:55] <Zeeflo> does anyone here know about movie files and meta data?
[09:07] <sacarasc> Zeeflo: I dare say a bunch of people do, but they'd also respond better to direct questions.
[09:28] <theos> hmm
[10:12] <Zeeflo> Well, its kind of a twoface question.
[10:13] <Zeeflo> I use flowplayer to stream my mp4's from an amazon s3 account
[10:13] <Zeeflo> when I use flowplayer, the stream has to cache for an eternity.
[10:13] <Zeeflo> When I use other players, such as videojs the stream loads instantly.
[10:14] <Zeeflo> Now, i've read somewhere that flowplayer has to read some meta info before it can start the stream/video.
[10:14] <Zeeflo> What if this meta info is in the end of the video file, then it has to precache the whole video before it starts streaming?
[10:15] <Zeeflo> so, how can I add this meta data to the beginning of the video file? Is it done with FFmpeg
[10:15] <Zeeflo> ?
[10:15] <Zeeflo> on the other hand, i dont get it.. Cause other players dont have to cache for ages before they start playing my stream/video
[11:16] <Nick-S> concating
[11:16] <Nick-S> is a bliss
[11:16] <Nick-S> but for webm you say i need mkvmerge, eh?
[11:16] <Nick-S> that sucks, i like ffmpeg i don't want to have a tool for it
[11:45] <theos> hey Nick-S
[11:45] <theos> JEEB you around?
[11:55] <theos> i am getting this error ERROR: libx264 not found. trying to configure :s
[11:56] <theos> nvm its not installed :s
[12:03] <theos> haha ERROR: libx264 must be installed and version must be >= 0.118
[12:05] <burek> theos you have 2 libx264 installed
[12:05] <theos> burek i do?
[12:06] <burek> probably one compiled yourself and one installed through the use of package manager
[12:06] <theos> i dont remember compiling my own. just installed libx264-dev
[12:07] <theos> the package is libx264-85
[12:07] <burek> oh i see
[12:07] <burek> well, you need to compile your own :)
[12:07] <theos> lol
[12:08] <burek> it's easy really
[12:08] <burek> let me give you a link
[12:08] <burek> http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20quickly%20compile%20FFmpeg%20with%20libx264%20(x264%2C%20H.264)
[12:09] <theos> thanks
[12:09] <burek> :beer: :)
[12:10] <theos> no beer left :/ cashews?
[12:11] Action: burek sets mode: +beer theos
[12:11] <burek> there can be no beer shortage
[12:11] <theos> :D
[12:15] <burek> theos, git url was incorrect for x264
[12:16] <burek> i just updated that article
[12:16] <theos> libx268 source is 70MB...
[12:16] <burek> should be git://git.videolan.org/x264.git
[12:16] <burek> no, it's ffmpeg
[12:16] <burek> my mistake :(
[12:16] <burek> you can rm -rf that
[12:16] <burek> and git clone git://git.videolan.org/x264.git
[12:16] <theos> yea my bad. didnt read the link text :<
[12:17] <burek> just refresh the page, should be showing the correct url now :S
[12:17] <burek> sorry..
[12:17] <theos> no problemo :)
[12:20] <theos> Found yasm 0.8.0.2194 Minimum version is yasm-1.0.0 If you really want to compile without asm, configure with --disable-asm.
[12:20] <theos> now i should compile yasm too? :)
[12:21] <burek> :))))))))
[12:21] <theos> ffmpeg doesnt like me :(
[12:22] <burek> well, disable asm would really cripple your x264
[12:22] <burek> git clone git://github.com/yasm/yasm.git
[12:22] <theos> yea
[12:22] <theos> huhuhu
[12:23] <theos> btw why do i need x264 to compress mp4?
[12:23] <JEEBsv> because you pretty much want H.264 in mp4, not mpeg-4 part 2
[12:23] <burek> mp4 is a format, no?
[12:23] <burek> or not
[12:23] <JEEBsv> and there is no better video encoder available atm than libx264
[12:24] <theos> JEEBsv!
[12:24] <SirDarius> hi ! kudos at libavformat and all the devs, i'm developing a program to do a looping 'rate_emulated' streaming of a playlist of videos files to RTMP, and it works so great that i just felt the urge to share my joy here :P
[12:25] <JEEBsv> SirDarius: do I parse a slight sense of sarcasm in thy words?
[12:25] <SirDarius> not at all, really
[12:25] <JEEBsv> :D
[12:25] <JEEBsv> great
[12:25] <SirDarius> playing is smooth, cpu usage is near zero
[12:25] <JEEBsv> because libav* can be a PITA, but when they work they work
[12:26] <SirDarius> took me some time to get started, had to plunge into ffmpeg.c, but in the end it's worth the effort :)
[12:30] <Mavrik> uhm... any idea where AnnexB standard for H.264 is defined?
[12:31] <JEEBsv> in the specification of H.264
[12:31] <JEEBsv> http://www.itu.int/ITU-T/recommendations/rec.aspx?rec=11466
[12:31] <Mavrik> doh.
[12:31] <Mavrik> JEEBsv: thanks.
[12:31] <Mavrik> :)
[12:32] <JEEBsv> Annex B indeed means, Annex B of the specification :)
[12:32] <JEEBsv> methinks
[12:33] <JEEBsv> yup
[12:34] <JEEBsv> Annex B: Byte Stream Format
[12:34] <theos> h264 is same as libx264?
[12:34] <JEEBsv> no
[12:34] <JEEBsv> H.264 is a standard
[12:34] <JEEBsv> libx264 is an encoder that implements encoding for that standard
[12:34] <JEEBsv> in other words, libx264 makes streams that are H.264, yes
[12:35] <JEEBsv> but H.264 does not equal libx246
[12:35] <JEEBsv> *libx264
[12:35] <theos> ok
[12:35] <Mavrik> yeah, for some reason I got sidetracked when I read "H.264 itself doesn't define bytestream syntax"
[12:36] <theos> why do i need libfdk_aac?
[12:36] <Mavrik> to encode AAC?
[12:36] <Mavrik> (that is - audio)
[12:36] <theos> i already have libfaac
[12:36] <JEEBsv> if you are going to be creating mp4 files, the audio you want most probably is AAC -- and fdk-aac is the best AAC encoder
[12:36] <JEEBsv> faac is much worse
[12:36] <theos> :;/
[12:37] <JEEBsv> naturally you can still use faac if you really want, I don't /really/ care
[12:37] <theos> cant i have the original audio?
[12:37] <JEEBsv> yes, if you use -c:a copy
[12:37] <theos> thanks
[12:37] <Mavrik> hmm, did anyone do a comparison libFAAC/libvo-aacenc/libfdk_aac lately? something I can throw at people? :)
[12:38] <JEEBsv> vo_aacenc sucks and only supports stereo
[12:38] <JEEBsv> faac was never great
[12:38] <JEEBsv> fdk is fraunhofer's and has won several tests, although this probably isn't exactly the same thing
[12:39] <JEEBsv> there was some japanese test between things lately'ish
[12:39] <JEEBsv> which happened to even tell that internal AAC encoder > vo_aacenc
[12:39] <JEEBsv> lol
[12:39] <Mavrik> JEEBsv: yeah, I'm interested in finding a rather comprehensive test... usually just saying "that encoder sucks" doesn't sell to my customers well :D
[12:40] <JEEBsv> (aka no wonder Google went for fraunhofer from vo_aacenc)
[12:40] <Mavrik> mhm, vo_aac was pretty terrible :)
[12:40] <JEEBsv> yeah
[13:25] <david_> Hi everyone, I have a problem with recording audio from both screen and mic anyone got some time to help me?
[13:31] <burek> david_, did you check our wiki page?
[13:39] <david_> I have downloaded ffmpeg from git and done my own configure with some parameters. What parameters enables alsa and hw calls?
[13:41] <burek> alsa is auto-detected
[13:41] <burek> if you install libasound2-dev
[13:43] <david_> my packagemanager says I have libasound2 but I get the error: Unknown input format: 'alsa'
[13:44] <burek> prior to ./configure
[13:45] <burek> and also libasound2-dev is not same as libasound2
[13:45] <david_> Ok, the dev was not visible in my package manager
[13:51] <burek> david_, btw, you dont need to do "make" in order to see if alsa got compiled or not
[13:51] <burek> after your ./configure finishes, you should see a list of all the things that will get built
[13:51] <burek> as the last info there are input/output devices
[13:52] <burek> if alsa is not listed there (together with probably oss) then you know you did something wrong
[13:53] <david_> Ok, I got a 32-bit opensuse 12.2 and cant find libasound2-dev/devel :/
[13:53] <burek> packages list | grep libasound ?
[13:58] <david_> Im a 1 year linux newbie tried your command without the question mark and got nothing
[14:00] <david_> I have libasound2 version 1.0.25 but cant find a dev package for it:/
[14:03] <burek> i didnt give you a suse-specific command
[14:04] <burek> i gave you a pseudo command, because im not familiar with suse package manager
[14:04] <burek> in debian i would write: apt-cache search libasound | grep libasound
[14:06] <burek> david_: zypper search libasound
[14:07] <david_> thx:)
[14:07] <burek> http://lilypond.org/blog/janneke/openSUSE-HOWTO
[14:07] <david_> were looking through zypper help command
[14:08] <david_> getting back this : i | libasound2 | Advanced Linux Sound Architecture Library | package
[14:08] <burek> you'll probably end up with: zypper install libasound2-devel
[14:09] <david_> :( Package 'libasound2-devel' not found.
[14:10] <burek> type here: /join #opensuse
[14:10] <david_> ok jumped over there
[14:10] <burek> they will know better i believe :)
[14:11] <david_> hopefully:)
[14:13] <david_> Thx for your help burek
[14:15] <burek> :beer: :)
[14:15] Action: Zeeflo hands burek a cold heineken
[14:16] <burek> meow :)
[14:16] <Zeeflo> i could sure as shit use one too!
[14:16] <klaxa> mmh... yeah i could use a beer right now too...
[14:51] <alix_> Hello.
[14:51] <alix_> Anyone know how to convert to swf ? I have no sound in browser with -acodec mp3 -ar 44100
[16:48] <SirDarius> seems I was being a little bit too optimistic earlier
[16:49] <SirDarius> some MP4 files make my program panic, I'm getting weird DTS values
[16:52] <SirDarius> I'm probably doing something very wrong when playing with pts/dts
[16:53] <SirDarius> I must admit that they confuse me a bit, especially the whole rational manipulation with the time_base
[16:59] <brx_> llogan i managed to get some good results yestereday by replacing -sameq with -q:v 2 as you sggested
[16:59] <brx_> so thanks for that (sry i saw missed your message to me yesterday)
[17:40] <SirDarius> hmm so, what is the difference between an AVStream's time_base, and an AVStream's codec's time_base ?
[17:46] <SirDarius> ah, I think I got this ! the packet's dts/pts need to be in avstream->time_base format, which changes according to the format, so when remuxing from MP4 to FLV i need to convert from eg. 1/25000 of seconds to 1/1000 of seconds
[17:47] <SirDarius> FLV always uses milliseconds, that I know, but MP4 seems variable, am i wrong ?
[17:57] <JEEB> SirDarius, MP4 lets you set your X/Y of the PTS freely
[17:57] <JEEB> FLV is milliseconds, yes
[17:57] <JEEB> (well, not fully freely, but in general you can select your timebase and all)
[17:58] <SirDarius> so when remuxing, I need to rescale the PTS/DTS, right ?
[17:58] <david> Hi all, so how do I get both my mic audio and monitor audio recoded at the same time?
[18:47] <Tthread> hello
[18:48] <Tthread> could you help me? I have memory leak in libavfilter in chain that has only buffer_src and buffer_sink filters, is it well known issue?
[18:50] <burek> david, why do you need that?
[19:08] <jordan__> is it possible to change the decode resolution without modifying the source? say i have hd encoded, but i don't need that high res on decode, it is possible to change that?
[19:08] <JEEB> no
[19:08] <JEEB> or well
[19:09] <JEEB> depends on the format, but f.ex. H.264 doesn't support it
[19:09] <JEEB> MPEG-4 Part 2 had a lowres setting at one point, but it might have been removed because it didn't really speed things up
[19:10] <jordan__> hmm
[19:10] <jordan__> that sucks
[19:11] <JEEB> if you need more speed, you'll just have to re-encode with a lower res :)
[19:11] <jordan__> yup
[19:12] <JEEB> also if you're using libx264 for H.264 encoding
[19:12] <JEEB> there is -tune fastdecode
[19:13] <jordan__> it's funny because i can encode with two different res one high one low, get the same disk size out. both terrible qualty, but one seeks faster than the other
[19:13] <jordan__> so i've found that it's not quality but resolution that is the biggest factor in seek
[19:16] <jordan__> actually the higher res looks worse becuase it is such a low bitrate, why does it *have* to decode it to such a large resolution when the quality of it is so terrible anyway
[19:17] <jordan__> anyway i guess it is a dead end
[19:18] <jordan__> hey is it possible to set say some flags in the video file and not actually reencode
[19:18] <jordan__> so that it will decode to a lower res?
[19:18] <jordan__> so modify the source but not actually reencode
[19:21] <JEEB> <jordan__> it's funny because i can encode with two different res one high one low, get the same disk size out. both terrible qualty, but one seeks faster than the other <- sounds like you're using some encoder that uses a static default bit rate
[19:21] <JEEB> which could be very low
[19:22] <JEEB> or you just have a lol old ffmpeg
[19:22] <jordan__> no i meant to do that
[19:22] <JEEB> "lol old" at the moment is until around middle of 2011 at the moment
[19:22] <JEEB> ugh
[19:22] <JEEB> yes, sorry
[19:22] <JEEB> I'm tired as hell
[19:22] <JEEB> and only read that one line
[19:22] <JEEB> and missed the next one
[19:23] <jordan__> i was checking whether it was the res or bitrate that was cuasing the choppy seeks
[19:23] <jordan__> it was the resolution
[19:23] <JEEB> and yes, while bit rate /does/ have something to do with how fast you can decode (more data = more things to decode)
[19:23] <JEEB> the resolution means generally more
[19:24] <jordan__> yea, so there must be something in the video file that tells the decoder what res to decode to?
[19:24] <jordan__> what if you just alter that?
[19:25] <jordan__> without a full re-encode
[19:42] <jordan__> there is something called transrating, that modifies the input stream but it does not allow to modify the resolution
[19:45] <rkrishna> Hi, is there a way to disable H.264 decoding in h264.c, I am just interested in extracting some metadata with out actually decoding
[19:57] <james3> hey guys
[19:57] <james3> what error is this: AVC: nal size
[20:05] <david> sry burek was AFK, the reason is that I want to record my voice at the same time as gamesounds
[20:09] <james3> what error is this: AVC: nal size
[20:09] <james3> hey david
[20:10] <david> hi james
[20:10] <ekristen> how would I convert a 1080p mpeg2 to a 1080p h264 mp4?
[20:11] <james3> with the command line or with programming ekristen?
[20:11] <llogan> ekristen: ffmpeg -i input -c:v libx264 output.mp4
[20:12] <ekristen> command line
[20:12] <james3> ^^
[20:12] <ekristen> llogan: really that simple?
[20:12] <james3> kk 1 sec
[20:12] <llogan> ekristen: it could be simpler: ffmpeg -i input output.mp4
[20:12] <ekristen> llogan: he made it on a single line
[20:12] <ekristen> lol
[20:13] <llogan> default is to use libx264 for mp4 container if it is supported (depending on your ffmpeg version)
[20:13] <ekristen> N-47441-g03847eb
[20:13] <llogan> otherwise 'mpeg4' is used
[20:13] <smj> is audio copied?
[20:14] <ekristen> Video: mpeg2video (Simple) ([2][0][0][0] / 0x0002), yuv420p, 12288x4096, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc -- well this one happens to be 420p it seems
[20:14] <llogan> smj: not by default
[20:14] <smj> what's with that resolution?
[20:14] <smj> uh, nevermind
[20:14] <james3> llogan: https://gist.github.com/4218615
[20:14] <llogan> 12228x4096!
[20:15] <james3> i don't think it's reading the frame but not sure why
[20:16] <ekristen> llogan: thats what ffmpeg is telling me
[20:16] <llogan> ffmpeg will use the input frame size for the output by default
[20:16] <ekristen> llogan: but I am pretty sure that is wrong
[20:17] <llogan> can you provide a sample?
[20:17] <ekristen> I can -- its just a mythtv recording of a 1080p stream from an hdhomerun box
[20:18] <ekristen> ah, I see the problem with that one --- the recording terminated early -- that might be why the details are messed up
[20:18] <llogan> smj: i should clarify. audio will be re-encoded by default, but not copied as in a stream copy ("copy and paste" from input to output).
[20:18] <james3> i thought you meant the code, nvm
[20:18] <llogan> james3: sorry, i'm not an API user
[20:19] <smj> rkrishna: I don't know about disabling H.264 decoding, but what kind of metadata are you looking for?
[20:24] <llogan> ekristen: see https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide for more info
[20:24] <ekristen> llogan: thanks
[20:25] <llogan> and let me know if anything is confusing or unclear so i can fix it
[20:25] <ekristen> llogan: will do -- I have to get another recording that is complete to test and then I'll run through it and follow back around with you if anything doesn't make sense
[20:26] <llogan> can ffplay playback the input that ffmpeg thinks is 12228x4096?
[21:56] <bhrobinson> can anyone help me with a missing x264 error
[21:58] <bhrobinson> trying to compile and it is telling me https://gist.github.com/557ea3c8172dfd52c173
[22:06] <JEEBsv> bhrobinson: pastebin config.log
[22:12] <bhrobinson> sorry, do you mean the config.err?
[22:13] <bhrobinson> https://gist.github.com/71f68823bdb65985291b
[22:23] <burek> bhrobinson http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20quickly%20compile%20FFmpeg%20with%20libx264%20(x264%2C%20H.264)
[22:24] <burek> add a closing bracket, if your irc client complains about unknown link
[22:24] <burek> )
[22:25] <burek> or just use this one: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20quickly%20compile%20FFmpeg%20with%20libx264%20(x264%2C%20H.264%29
[22:26] <bhrobinson> awesome!
[22:26] <bhrobinson> let me try that... you guys are great!
[00:00] --- Thu Dec  6 2012