[Ffmpeg-devel-irc] ffmpeg.log.20120630

[burek]

[00:09] <intracube> ok, one last question about x264
[00:10] <JEEB> sure
[00:10] <intracube> If i set a fairly high CRF value like 16, brighter parts of the image look fine - right down to reproducing the noise from the original footage. But the darker parts of the image suffer from macroblocking/colour banding :/
[00:10] <intracube> is there a way to re-allocate bits away from the highlights and towards the shadows?
[00:10] <intracube> I've heard that this is a known issue with x264
[00:10] <JEEB> uhh, 16 being "fairly high"? :D
[00:11] <JEEB> that's usually quite the low end of CRF
[00:11] <intracube> well, it reproduces the film grain/noise in the highlights, but not elsewhere
[00:11] <JEEB> also, how old of an x264 are you using, just per chance?
[00:12] <intracube> I'm using ffmpeg 0.11.1
[00:12] <JEEB> yes, but what about libx264/x264 version? API ver + revision
[00:12] <intracube> wouldn't x264 be compiled into ffmpeg?
[00:12] <intracube> ah hang on
[00:12] <intracube> checking
[00:13] <JEEB> if it's statically linked, yes -- but you still should know which version is linked there
[00:13] <JEEB> also, yeah -- it often does mostly put its work onto stuff that is generally seen. Some of that kind of stuff happens with badly calibrated screens that then end up showing stuff that shouldn't be visible.
[00:13] <JEEB> also I'd be interested in which preset you are using
[00:14] <intracube> hmm, according to "rpm -qa | grep -i x264" I have libx264-122-0, libx264-125-0, libx264-120-0
[00:15] <JEEB> which are all "relatively new", but whatever is that one that's linked to your ffmpeg :D
[00:15] <intracube> ldd says libx264.so.125
[00:15] <JEEB> okies
[00:15] <JEEB> now what kind of preset and friends are you using?
[00:16] <intracube> I did the original encode a while back, but I think it was 'slow'. I've still got the source here, so I'll have another go
[00:16] <intracube> what CRF value would you recommend?
[00:16] <intracube> 5-10?
[00:16] <JEEB> what for? :D
[00:17] <JEEB> for general usage unless your source is special in some way (there are such) I wouldn't usually go far lower than 18
[00:17] <intracube> fairly static content, computer generated graphics a small amount of added noise/grain
[00:17] <JEEB> but it really depends on what is it that you're trying to do
[00:19] <intracube> my source is high quality - a png sequence rendered from Blender
[00:19] <JEEB> you could try playing around with aq-mode (1/2), aq-strength and psy-rd as well as the crf
[00:19] <intracube> I'll try CRF 10 with the slow preset
[00:20] <JEEB> and yeah, if you have no file size limit and/or the source doesn't really use much bit rate even with lower crf values, you can just keep lowering it
[00:21] <intracube> JEEB: I'll take a look at those other params
[00:24] <JEEB> there's also an old patch for D_S's film grain optimization which is what then later became --psy-rd. psy-rd is more general and probably more useful, but some people ended up liking fgo as it happened to be more "friendly" towards animated content f.ex. -- now third parties keep that patch up-to-date :)
[00:24] <JEEB> also, as someone who actually liked to play around with settings I must confess that in many cases after just reseting my settings back to defaults (preset + tune) my results often were better off ^^;
[00:25] <JEEB> also I guess most of those settings I mentioned have to be accessed via the x264opts setting in ffmpeg :)
[00:25] <intracube> I'll have a play and report back :)
[00:26] <llogan> there is also the -tune option to monkey with
[00:27] <JEEB> also, if it's just one part that needs a lower crf value, you can always use zones to lower the used crf value in a spot :)
[00:52] <intracube> setting aq-strength=2 has given the biggest improvement so far
[00:53] <intracube> I'll have to see if it has a negative impact on different types of content
[00:54] <intracube> I wound't normally advocate upscaling video, but it's probably beneficial when uploading to youtube
[00:55] <JEEB> yeah, to get to ~720p for example -- there are certain resizing types you can use to get the most of various content for that
[00:58] <intracube> the lower resolutions on YT seem particularly bit-starved...
[00:58] <intracube> at 480p, this video looks very average: http://www.youtube.com/watch?v=E98iTbFqRuM
[00:59] <intracube> switching to 1080p makes a *massive* difference - not just in sharpness, but in the reproduction of the film grain
[01:01] <intracube> llogan: where can I find more info on the -tune option?
[01:02] <intracube> llogan: nevermind, it was on the link JEEB posted earlier
[01:03] <JEEB> :)
[01:36] <intracube> "-crf 16 -preset slow -tune film -x264opts aq-strength=1.5" seems a good compromise
[01:37] <intracube> thanks for the help guys
[01:37] <JEEB> no problemo
[01:44] <BigToe> hi. can anyone help me? i want to remux a blu ray m2ts file using ffmpeg into a mkv container. but i am confused as to how to get the subtitle track? 0:0 is video, 0:1 is audio, and 0:6 is the subtitle track. so far i have ffmpeg -i 00000.m2ts -vcodec copy -map 0:0 -map 0:1 -map 0:6 -acodec libfaac -ac 6 -ab 640k output.mkv
[01:46] <BigToe> burek: you there bro?
[01:48] <BigToe> cbreak: you around feller?
[01:52] <jzmer> does ffmpeg use libaacplus to decode aac audio?
[01:53] <JEEB> there's various stuff for various types of AAC
[01:54] <JEEB> there's the most stable but nonfree (lib)faac for LC AAC, the GPL-compatible (lib)opencore(?) LC AAC encoder, and the experimental libavcodec's internal AAC encoder
[01:54] <JEEB> then there's the nonfree (lib)aacplus for HE-AAC
[01:55] <jzmer> but does libfaac does only encoding?
[01:55] <JEEB> oooh
[01:55] <JEEB> you were asking decoding
[01:55] <JEEB> misread
[01:55] <JEEB> sorry
[01:55] <JEEB> decoding is done by the libavcodec decoder in most/all cases
[01:55] <JEEB> which is internal
[01:55] <jzmer> its ok. all of us misread sometimes
[01:55] <jzmer> erh, "most cases", and there could be exceptions?
[01:56] <BigToe> JEEB: any idea for help with my question?
[01:56] <JEEB> there are a few kinds of AAC that the libavcodec aac decoder can't handle
[01:56] <JEEB> but you should just get an error with those if you happened to stumble upon such
[01:56] <JEEB> let's just say that unless you get REALLY unlucky you won't hit anything
[01:56] <JEEB> and I think the general HE-AAC types are just fine
[01:57] <JEEB> (of course I'm not sure and thus my selection of wording "most/all cases")
[01:57] <llogan> piping from an external decoder to ffmpeg is always an option too
[01:57] <jzmer> thats very helpful. thank you, JEEB.
[01:58] <llogan> BigToe: use a pastebin site to show your ffmpeg command and the complete console output
[01:59] <JEEB> I'm not sure if ffmpeg could handle picture/text based blu-ray subtitles
[01:59] <llogan> burek should make flogger squawk that with a !pastebin
[01:59] <JEEB> but subtitles are -scodec/-c:s methinks
[02:01] <BigToe> llogan: http://pastebin.com/kKA752mi
[02:01] <JEEB> Stream #0:6 -> #0:2 (pgssub -> ass) ....yeah, not gonna happen unless ffmpeg has an OCR system built in :D
[02:02] <JEEB> -scodec copy
[02:03] <BigToe> JEEB: when i do -scodec copy there's no subtitle track in the file
[02:04] <JEEB> hm?
[02:04] <BigToe> JEEB: i used mediainfo and there's only two tracks. audio and video
[02:05] <JEEB> also you don't want to be re-encoding that DTS-HD MA track with libavcodec, as libavcodec can only handle the lossy core
[02:06] <JEEB> regarding the subtitles I don't know, were there any subtitles in that part of time? Also I'm not sure of what things ffmpeg can do with pgs subtitles
[02:07] <BigToe> JEEB: i have surround sound headphones that only work with dolby digital 5.1. i wanted to convert it so i could have surround sound
[02:07] <BigToe> JEEB: is there a better way of doing that? i'm kind of new to this stuff
[02:08] <JEEB> for DTS-HD MA the only way to decode it is to use arcsoft's decoder binary on windows with something
[02:08] <JEEB> eac3to is probably the most popular thing for that
[02:08] <JEEB> (to get the full lossless stream decoded)
[02:08] <BigToe> ah. so the command i used won't make it dolby surround sound any way?
[02:09] <JEEB> hm?
[02:09] <JEEB> no, it will encode AC3 of course
[02:09] <JEEB> I'm just saying that it will only decode the lossy core of the DTS-HD MA track
[02:09] <JEEB> which is a normal DTS track IIRC
[02:09] <JEEB> also that transport stream already has an AC3 track
[02:10] <JEEB> is that not good?
[02:10] <BigToe> oh okay. i think i understand
[02:10] <BigToe> JEEB: i think it's another language
[02:10] <JEEB> ah
[02:11] <BigToe> JEEB: that part of the film i tested didn't have subs. i'll try the full thing
[02:13] <JEEB> I wonder if someone'll implement DTS-HD MA fully some day
[02:13] <JEEB> in libavcodec
[02:14] <JEEB> there's a specification for it now available, but it's basically a book that says "please consult our licensing for a copy of the reference implementation" in many things :)
[02:14] <JEEB> s/things/places/
[03:00] <BigToe> JEEB: subs still don't work :'( i found the subs on the internet. is there a way i can add them to the ffmpeg syntax? like could i use -i 2wice or is there a different way?
[03:05] <llogan> BigToe: yes, you can specify as many inputs as you like
[03:08] <BigToe> llogan: so would the final command look like this? "ffmpeg -i 00000.m2ts -i subs.srt -vcodec copy -map 0:0 -map 0:1 -acodec ac3 -ac 6 -ab 640k -scodec copy output.mkv"?
[03:08] <BigToe> the first source file also has subtitle tracks. so do i need to map for the second source file?
[03:08] <BigToe> i just want the second source subs
[03:08] <BigToe> not the first
[03:11] <llogan> BigToe: yes, refer to the documentation on -map
[03:12] <llogan> but it's probably -map 1:0
[03:13] <BigToe> llogan: yep it is. thanks :)
[08:50] <marcosmlopes> hello
[08:50] <marcosmlopes> could someone help me?
[08:54] <marcosmlopes> i want to implement an streaming server in node.js and i need to know how ffmpeg publish video through http. Can someone point me and article ou documentation about this? Thanks.
[09:13] <marcosmlopes> could someone help me?
[13:14] <raptor67682>     ffplay http://www.healthradio.net/player/healthradio/playlist.pls
[13:14] <raptor67682>  ffplay cannot play URL from the internet pitty
[13:14] <raptor67682> is it a bug?
[13:14] <microchip_> it's a feature! :P
[13:15] <raptor67682> ffplay -vn     "mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf"
[13:15] <raptor67682> can ffmpeg play any URL streams for the internet ?
[13:15] <raptor67682> (debian stable squeeze)
[13:15] <raptor67682> or could it be capable to play anything in a soon future
[13:16] <raptor67682> replace ffplay by mplayer and youwill see that itis possibleto play an url stream using linux ;)
[13:31] <raptor67682> has  ffmpeg been tested before made available for debian stable?
[13:58] <burek> raptor67682, ff* does not support playlists (yet)
[14:00] <burek> raptor67682, try ffplay mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf -vn
[14:00] <burek> I told you yesterday that -vn is an output option
[14:01] <raptor67682>    not working  mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf: no such file or directory
[14:01] <raptor67682> ffplay says :
[14:01] <raptor67682>   mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf: no such file or directory
[14:01] <burek> also debian stable is always behind with the freshness of any package, due to that "stable" thing
[14:01] <raptor67682>     ffplay "mms://msnbc.wmod.llnwd.net/a275/e1/video/100/vh.asf" -vn
[14:01] <raptor67682>    not working
[14:01] <burek> so you'll never have up-to-date things like that
[14:01] <burek> your best bet is to compile from source
[14:01] <raptor67682> burek: are u using SID ?
[14:01] <burek> most probably you have something that is 1-2 years old
[14:02] <raptor67682> burek:  are you sure that it will work in the next versions of debian?
[14:02] <burek> what do you mean if it will work? of course it will
[14:02] <burek> updating distro won't touch your own compiled stuff
[14:02] <raptor67682>  sox cannot play url streams without being choppy, ffmpeg neither, mplayer= not installed, and mpg321 cannot play -vn else than mp3
[14:03] <burek> and if something still goes wrong, just recompile it again and it should be fine
[14:03] <raptor67682> burek: I have to use stable debian. bosses say that stable is to be used, because we dont mess at work
[14:03] <burek> also
[14:03] <burek> raptor67682, can you please use pastebin.com, to show your command line and its output?
[14:04] <burek> no, I'm not using SID
[14:04] <burek> that's too crazy to use in production area
[14:04] <burek> (at least for me)
[14:04] <burek> raptor67682, I also use stable debians everywhere
[14:04] <burek> I just compile things I need to be up-to-date
[14:04] <burek> the rest, I get with apt-get
[14:05] <raptor67682>   http://pastebin.ca/2165822
[14:05] <burek> uh...
[14:05] <burek> it's no wonder nothing works...
[14:05] <burek> version SVN...
[14:05] <burek> svn has been abandoned a long time ago in favor of git
[14:06] <burek> can you try a quick test?
[14:06] <raptor67682>    apt-cache search ffmpeg  reports ffmpeg - audio/video encoder, streaming server & audio/video file converter
[14:06] <burek> download the latest available
[14:06] <burek> and just run it
[14:06] <raptor67682> dpkg -l | grep ffmpeg
[14:06] <raptor67682> ii  ffmpeg                               4:0.5.4-1                          multimedia player, server and encoder
[14:06] <burek> to see if it will work
[14:06] <burek> but I doubt it will since it uses network
[14:06] <burek> and static version does not support a lot of network features
[14:06] <burek> anyway, your ffmpeg is too old :(
[14:07] <burek> you'll have to compile it yourself
[14:07] <burek> or use debian-multimedia repository
[14:09] <raptor67682> let me check ...
[14:10] <raptor67682>        it is installed : deb http://www.debian-multimedia.org squeeze main
[14:10] <raptor67682>   the sources.list has it
[14:10] <raptor67682>   so you mean that it is not using the debian-multimedia repositories?
[14:12] <burek> I don't know, really
[14:12] <burek> If I were you, I would compile it
[14:13] <raptor67682> well, it is possible to compile, and use it into my ~/ directory in a given directory
[14:13] <burek> use --prefix
[14:13] <burek> although, why would you do that anyway
[14:14] <burek> it will go into /usr/local instead of the root
[14:14] <raptor67682>   well, I could try to re-update ffmpeg using the debian-multimedia  ... well, it would be still debian stable in some ways
[14:14] <burek> take a look at it this way
[14:14] <burek> package maintainers have a lot of packages to update all the time
[14:14] <burek> that takes time
[14:15] <burek> which means they can't possibly keep up-to-date with all the recent changes that ffmpeg developers do
[14:15] <raptor67682> burek: I wished that it would be more automatic to update/packages
[14:15] <burek> so if you use prebuilt packages you will always lack behind
[14:15] <raptor67682> deb making takes lot of time
[14:15] <burek> well it is actually
[14:15] <burek> look, try just to build it once
[14:15] <burek> next time you need to update
[14:16] <burek> go to your source code directory
[14:16] <burek> type git pull
[14:16] <burek> and recompile again and thats it
[14:16] <burek> once in a month would be enough
[14:16] <raptor67682> burek: I have broken the sound ... I changed some para in the modprobe.d ... got to fix before
[14:17] <burek> raptor67682, also, by compiling your own ffmpeg, you make sure that all the optimizations, that your own hardware supports, will be used
[14:17] <burek> thus you'll get the most out of your machine
[14:18] <raptor67682> certailny the  debian-multimedia way could be much faster... Just need to listen music at work
[14:19] <raptor67682> and... sometimes soccer ... soon the final
[15:49] <mparodi> Hello
[15:51] <mparodi> how do you convert a AVCHD (MTS file) to a "normal/common" format?
[15:51] <mparodi> I'm having problems watching it directly from this file (the audio is not sync)
[15:52] <mparodi> there're thousand of options, I don't know what to use o.o
[16:25] <mparodi> any idea?
[16:32] <mparodi> `md, ods15, saste, ubitux, uau: any idea?
[17:02] <jesk>  is it of any use to get near to a "constant bit rate(cbr)", especially in network streaming?
[17:04] <Mavrik> not unless you want to ruin your video quality :)
[17:04] <mparodi> what video filter should I use to remove the interlacing?
[17:04] <Mavrik> mparodi: yadif
[17:04] <mparodi> -deinterlace -vf yadif ?
[17:04] <Mavrik> no
[17:05] <Mavrik> -vf yadif
[17:05] <Mavrik> :)
[17:06] <mparodi> ffmpeg -i inputfile.mts -vcodec copy -vf yadif -b:v 3000k -acodec libmp3lame -b:a 128k -pass 2 outputfile.avi
[17:06] <mparodi> does it make sense?
[17:06] <Mavrik> no
[17:06] <Mavrik> because you're just copying the video
[17:07] <mparodi> ahm
[17:07] <Mavrik> video filters won't get applied
[17:07] <mparodi> I want a mp4 or an avi
[17:07] <mparodi> what is the correct vcodec?
[17:07] <Mavrik> don't know
[17:07] <Mavrik> which video codec do you want?
[17:07] <Mavrik> mp4 and avi are containers.
[17:08] <mparodi> I don't care that much, I just want to convert this mts to something more "common" :|
[17:08] <Mavrik> use libx264 then
[17:08] <Mavrik> if you want to pack stuff into mp4
[17:08] <mparodi> ffmpeg -i inputfile.mts -vcodec libx264 -vf yadif -b:v 3000k -acodec libmp3lame -b:a 128k -pass 2 outputfile.mp4
[17:08] <mparodi> ?
[17:09] <mparodi> trying
[17:09] <mparodi> [libx264 @ 0x1e8bda0] ratecontrol_init: can't open stats file
[17:09] <mparodi> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[17:10] <jesk> Mavrik: i read somewhere that some decoders aren't capable of decoding at vbr
[17:10] <mparodi> I really don't know, it has like 2 thousand options and parameters!
[17:10] <jesk> Mavrik: but I have no idea about that
[17:12] <jesk> mparodi, did -pass 1 worked in the first place?
[17:13] <mparodi> no
[17:13] <mparodi> output http://paste.kde.org/510332 jesk
[17:14] <mparodi> specially this part: [buffer @ 0x13d6760] Changing frame properties on the fly is not supported.
[17:14] <mparodi>     Last message repeated 19 times
[17:18] <mparodi> suppose you need to convert a mts file to a mp4 (inside you can use whatever you like if it's common enough and if the quality is not that bad)
[17:18] <mparodi> what would you execute? jesk
[17:23] <jesk> how about just that:: ffmpeg -i source.mts -c:v h264 -c:a libfaac new.mp4
[17:24] <mparodi> Unknown encoder 'h264'
[17:25] <jesk> libx264
[17:26] <mparodi> [buffer @ 0x8fb3e0] Changing frame properties on the fly is not supported.
[17:26] <mparodi> same
[17:26] <mparodi> :|
[17:28] <jesk> ffmpeg version?
[17:29] <mparodi> ffmpeg version 0.10.3
[17:29] <jesk> considered a new one?
[17:29] <mparodi> is it the problem?
[17:29] <mparodi>  built on May  6 2012
[17:29] <mparodi> it's the ffmpeg in the repository
[17:29] <mparodi> (Debian testing)
[17:29] <jesk> i dont know, but building new one is maybe the easiest way to exclude another problemcause
[17:30] <jesk> takes you 30 minutes
[17:30] <mparodi> ok, I'll try now with handbrake as someone suggested me in #debian
[17:31] <jesk> there could be two reasons, either your .mts file is the reason or the ffmpeg version
[17:31] <mparodi> and then if it doesn't work I'll try to upgrade
[17:32] <Steve__> hey guys. have an application that's currently windows only. it uses directshow or windows media foundation to decode video which is then rendered in a 3d engine. looking at alternatives & don't know much about ffmpeg yet. can it decode into directx hardware surface ? (directx9 or dx11 ?)
[17:34] <raouk> hey im on windows using a 20120629 static snapshot, and im trying to convert a tif sequence to a movie.  the output of mpeg looks fine, but the video file just grey. any ideas?
[17:36] <Mavrik> Steve__: you'll have to copy decoded frames to the hardware surface
[17:37] <Steve__> Mavrik: that's what I thought you were going to say :( that means win media foundation is the most optimal solution at the moment as it hardware decodes straight into a directx surface.
[17:38] <Mavrik> Steve__: of course, ffmpeg is a CPU based encoder
[17:38] <Mavrik> Steve__: you can't expect CPU decoded frames to be in VRAM :)
[17:38] <Steve__> understood. just figuring out the state of play.
[17:38] <Steve__> unfortunately win media foundation currently appears to have very limited rtsp support.
[17:38] <Mavrik> Steve__: there is an api to pack demuxed data into PureVideo format to be shipped to GPU for processing
[17:38] <Mavrik> but that won't be cross-platform portable
[17:39] <Steve__> would be if it was written in opencl
[17:39] <Mavrik> since you'll have to implement PureVideo (or w/e) on Windows, VA-API/VDPAU on linux etc.
[17:41] <Steve__> & purevideo is nvida only ?
[17:42] <Mavrik> er, yeah, I meant the Windows HW video decoding API
[17:42] <Mavrik> whatever WMF is using
[17:43] <Steve__> so possibly use ffmpeg to access the rtsp stream then use wmf to harware decode.
[17:44] <Steve__> it's a crazy plan. I like it :)
[17:57] <Steve__> thanks Mavrik. think the relevant interface is DXVA2 but need to do some research.
[17:57] <Steve__> http://msdn.microsoft.com/en-us/library/windows/desktop/cc307941%28v=vs.85%29.aspx
[17:59] <naquad> i have 3 web cams that i want to broadcast as 1 single thread with 3 regions displaying video. can i somehow do that with ffmpeg?
[18:00] <burek> naquad, usb webcams?
[18:01] <naquad> burek, they're ethernet
[18:01] <burek> ip cams
[18:01] <naquad> they broadcast via http to local network
[18:01] <naquad> yeah
[18:01] <burek> yes you can
[18:01] <naquad> is there any tutorial on this? i've never tried to compose image with ffmpeg :( i don't know even where to start
[18:01] <burek> ffmpeg -i cam1 -i cam2 -i cam3 -vf overlay=... -f mpegts udp://server:port
[18:02] <burek> oh, well, it will take you at least an hour
[18:02] <naquad> burek, thank you, went reading
[18:03] <burek> :beer: :)
[18:04] <Steve__> Mavrik: did a bit of googling. sounds like there's already dxva2 ffmpeg support around. no idea what the state of play is so more googling ...
[18:05] <burek> raouk, can you please use pastebin.com, to show your command line and its output?
[18:06] <raouk> burek, http://pastebin.com/uDA7jrEn
[18:08] <Steve__> Mavrik: apparently libavcodec contains support for dxva ... continues researching ...
[18:08] <burek> "the output of mpeg looks fine, but the video file just grey. any ideas?"
[18:08] <burek> what does that mean raouk :)
[18:09] <burek> Steve__, Mavrik is not online
[18:09] <raouk> burek, ffmpeg doesnt appear to have errored judging from its output.  a video file is created, 16 seconds long (aboutright), but all the frames are grey
[18:11] <Orbixx> I keep getting "Error while opening encoder for output stream #x:x - maybe incorrect parameters such as bit_rate, rate, width or height" whenever attempting conversions.
[18:12] <burek> Orbixx, can you please use pastebin.com, to show your command line and its output?
[18:12] <burek> well raouk is there a specific reason
[18:12] <burek> why do you want it to be mpg?
[18:12] <Steve__> cheers. talking to myself now anyway - looks like I have what I need.  thanks guys :)
[18:12] <raouk> burek, nope. i just want a video from this sequence
[18:13] <burek> raouk, for example, try this: ffmpeg.exe -i img%05d.tif -vcodec libx264 out.mp4
[18:13] <raouk> burek, same results
[18:14] <burek> can't be
[18:14] <burek> mp4 not mpg
[18:14] <raouk> but it is mp4 and it is the same result
[18:14] <burek> that's strange
[18:14] <Orbixx> burek: http://pastebin.com/nzpca1h4
[18:15] <burek> are your tiff images ok raouk ?
[18:15] <burek> i.e. are they gray too?
[18:15] <raouk> burek, yup. they are fine.
[18:16] <burek> Orbixx, your audio is 48KHz
[18:16] <burek> flv does not support mp3 of 48 KHZ
[18:16] <burek> try this
[18:16] <burek> fmpeg -i clip.mkv -ar 44100 clip.flv
[18:16] <burek> also, are you trying to reconvert your audio/video or just remux to flv?
[18:16] <burek> raouk, I don't know then...
[18:17] <burek> it seems ffmpeg reads your tiff images as gray pictures..
[18:17] <mparodi> have anyone tried HandBrake?
[18:17] <Orbixx> burek: For some reason, Youtube is failing to convert my video file, so I'm simply trying to get it into a format that it's able to process. I'm not particularly familiar with A/V and codecs, sorry.
[18:18] <mparodi> there're many deinterlace options, Fast, Slow, Slower, Bob, Custom
[18:18] <burek> Orbixx, I see.. Well, format (container) is just an envelope to audio/video streams, encoded with different codecs
[18:18] <mparodi> but the output looks the same
[18:18] <Orbixx> burek: Also, that suggestion produces the same error.
[18:18] <mparodi> what's the difference?
[18:18] <Orbixx> burek: Ah, I know about containers.
[18:18] <burek> so, my question was, if you want to just change the container and keep your original audio/video streams (without re-encoding them)
[18:18] <burek> or do you want to re-encode everything completely
[18:18] <Orbixx> burek: Presumably Youtube can handle containers fine, so it's having a problem with the actual codec used.
[18:18] <burek> and possibly loose quality and time
[18:18] <Orbixx> Re-encode.
[18:19] <burek> mparodi, ask on the handbrake support channel, not here
[18:19] <burek> Orbixx, try this
[18:19] <mparodi> maybe you knew since it uses ffmpeg behind the scene
[18:20] <burek> fmpeg -i clip.mkv -vcodec libx264 -acodec aac -strict experimental -ar 44100 -ab 128k -ac 2 clip.flv
[18:20] <burek> mparodi, still, ask on their support channel
[18:20] <mparodi> (I did)
[18:20] <mparodi> thanks
[18:20] <burek> why in particular do you use handbrake?
[18:20] <burek> i.e. what does it do, that ffmpeg can't do?
[18:21] <burek> raouk, try converting your images to bmp maybe
[18:21] <mparodi> in fact, nothing... the point is that I was having problem to convert a mts file to mp4
[18:21] <mparodi> I prefer to use ffmpeg, of course
[18:21] <burek> ffmpeg -i input.mts -c copy output.mp4
[18:22] <Orbixx> burek: No stream processing output, produces a 0-byte file.
[18:22] <Orbixx> No error
[18:22] <Orbixx> Just quits after it appears to begin processing, but does nothing
[18:22] <burek> Orbixx, can you please use pastebin.com, to show your command line and its output?
[18:22] <mparodi> no, I need to apply -vf yadif so I can't copy the video, burek
[18:23] <burek> mparodi, can you give a pastebin of: ffmpeg -i input.mts
[18:23] <mparodi> ok
[18:23] <Orbixx> burek: http://pastebin.com/K3WjfGjd
[18:24] <raouk> burek, that would probably be best. i ran ffmpeg on a linux box and got tiff errors 'Invalid strip size/offset', even though they are visibly fine
[18:24] <mparodi> http://paste.kde.org/510380 burek
[18:25] <mparodi> ffmpeg -i OpenFING.mts -c:v libx264 -vf yadif -c:a libfaac new.mp4
[18:25] <mparodi> I tried with this ^
[18:26] <mparodi> this is the first error that appears: [buffer @ 0x14e5740] Changing frame properties on the fly is not supported.
[18:27] <burek> Orbixx, everything looks fine...
[18:28] <burek> I really can't tell there is any trouble there..
[18:28] <Orbixx> burek: Yeah :/ Bizarre
[18:28] <Orbixx> burek: It just stalls and exits
[18:28] <burek> since you compiled your ffmpeg
[18:28] <burek> can you make uninstall/distclean
[18:28] <burek> git pull
[18:28] <burek> make/make install
[18:28] <burek> ?
[18:28] <Orbixx> burek: I did not, I used somebody's x86_64 win32 nightly builds
[18:28] <burek> just to check if there was some bug or something
[18:28] <burek> oh
[18:28] <Orbixx> burek: I believe they're linked to from the ffmpeg site
[18:28] <burek> is there any newer version
[18:29] <Orbixx> Most likely
[18:29] <burek> can you try it
[18:29] <burek> mparodi, try this
[18:30] <burek> ffmpeg -i OpenFING.mts -vf yadif -vcodec libx264 -acodec aac -ac 2 -ar 44100 -ab 128k -strict experimental out.mp4
[18:31] <mparodi> same
[18:31] <burek> (I'm not sure if mp4 container can hold ac3 audio
[18:31] <burek> if it can, just use -acodec copy instead of "-acodec aac -ac 2 -ar 44100 -ab 128k -strict experimental"
[18:31] <mparodi> Incompatible sample format 's16' for codec 'aac', auto-selecting format 'flt'
[18:31] <burek> mparodi, can you please use pastebin.com, to show your command line and its output?
[18:31] <mparodi> http://paste.kde.org/510386
[18:31] <Orbixx> burek: Looks like it was a bug ^^
[18:31] <mparodi> yep, I was doing so :P
[18:31] <burek> Orbixx, cool :)
[18:32] <Orbixx> burek: Appreciate the guidance, thank you.
[18:33] <burek> :beer: :)
[18:35] <mparodi> burek, btw, if you have a better idea it's ok as well. I don't need it in mp4, I just need to apply deinterlace and convert it to a common format
[18:35] <burek> mparodi, "Changing frame properties on the fly is not supported."
[18:36] <burek> I think you need to update your ffmpeg to the latest version
[18:36] <burek> can you compile your ffmpeg?
[18:36] <burek> or use a static version
[18:36] <mparodi> I think so
[18:36] <burek> (just download and run, see if it works)
[18:36] <burek> it's quicker
[18:36] <mparodi> ok, and after that what should I use
[18:36] <mparodi> ?
[18:36] <burek> the same cmd line
[18:36] <mparodi> % ffmpeg -i OpenFING.mts -c:v libx264 -vf yadif -c:a libfaac new.mp4
[18:36] <burek> ffmpeg -i OpenFING.mts -vf yadif -vcodec libx264 -acodec aac -ac 2 -ar 44100 -ab 128k -strict experimental out.mp4
[18:36] <burek> or
[18:36] <mparodi> I guess
[18:36] <burek> ffmpeg -i OpenFING.mts -vf yadif -vcodec libx264 -acodec copy out.mp4
[18:37] <mparodi> ok ok
[18:37] <mparodi> thank you!
[18:37] <burek> if 2nd one works, then your audio will be left in the original form
[18:37] <burek> i.e. not re-encoded
[18:37] <burek> :beer: :)
[18:40] <mparodi> one last thing
[18:40] <mparodi> do you suggest aac over mp3?
[18:41] <burek> well, imho, aac+ is far better than mp3
[18:41] <mparodi> alright
[18:41] <burek> but aac not
[18:41] <burek> aac-LC
[18:41] <mparodi> u.u
[18:41] <burek> which is -acodec aac
[18:41] <burek> that's about the same as mp3
[18:41] <mparodi> ok
[18:41] <burek> -acodec libaacplus is AAC+
[18:41] <burek> but you have to compile it in your ffmpeg
[18:42] <mparodi> I'll use mp3 then
[18:42] <mparodi> I upgraded ffmpeg and it's working now :D
[18:42] <burek> ok :)
[18:42] <mparodi> % ffmpeg -i OpenFING.mts -c:v libx264 -vf yadif -c:a libmp3lame output.mp4
[18:42] <burek> cool :)
[18:43] <mparodi> in fact, I'm doing this because my video player wasn't applying a deinterlace filter automatically. I'm not quite sure how intelligent it is to remove it
[18:45] <burek> :)
[18:51] <mparodi> can ffmpeg convert a video to have the metadata at the beginning?
[18:51] <burek> mparodi, what metadata
[18:51] <mparodi> to stream the video, for example
[18:51] <burek> that's not metadata
[18:51] <burek> you mean like moov atom and stuff?
[18:51] <mparodi> when you want to play a video on internet?
[18:52] <burek> if your files are mp4, use qt-faststart
[18:52] <mparodi> like in youtube or wherever
[19:45] <mparodi> what if it's webm?
[20:20] <Lupu_> hello, does the FFserver has his own irc channel ?
[20:22] <burek> Lupu_, no, it's part of ff* family
[20:23] <burek> you can ask the question here
[20:23] <burek> mparodi, well, test it :)
[20:23] <Lupu_> ok, it is possible to restream a feed without re-encoding
[20:24] <burek> yes it is
[20:24] <Lupu_> sonthing lice -vcodec copy
[20:24] <burek> VideoCodec copy
[20:24] <marcosmlopes> Hi guys, can someone point me an resource explaining how ffmpeg publish video through http to ffserver? I want to implement my own media server using node.js. Thanks
[20:24] <burek> in ffserver.conf
[20:24] <burek> Lupu_, just bear in mind that specific codecs can go into certain containers.. i.e. not all codecs can go into all containers
[20:25] <Lupu_> the input stream is mpeg2 in a mpeg TS container
[20:25] <burek> marcosmlopes, ffmpeg feeds the ffserver through ffm format/container, like this: ffmpeg -i input.avi -f ffm http://localhost:8090/feed1.ffm
[20:25] <Lupu_> the ouput i want it to bee the same
[20:25] <burek> but you can omit -f ffm
[20:25] <burek> ffmpeg -i input.avi http://localhost:8090/feed1.ffm
[20:25] <burek> that'll do
[20:26] <burek> mpeg TS over http? hmm..
[20:26] <marcosmlopes> i see. Can i change the container to webm?
[20:28] <burek> Lupu_, try this: http://pastebin.com/6C39csgF
[20:28] <Lupu_> the input stream is also mpeg TS over http
[20:28] <Lupu_> thx, il wil test now
[20:28] <burek> marcosmlopes, yes you can, it's just a question if your audio/video stream (inside the container) is supported in webm
[20:28] <burek> or you might need to re-encode them into some other codec(s) that are supported by webm
[20:29] <marcosmlopes> burek: hmm, thanks!
[20:29] <burek> :beer: :)
[20:29] <marcosmlopes> :)
[20:29] <burek> Lupu_, ffmpeg -i http://server/input.ts http://localhost:8090/feed1.ffm
[20:29] <burek> Lupu_, it might be good to set -analyzeduration 10000000 before -i
[20:30] <burek> because ts might be sending delta frames for the time of default analyzeduration time period
[20:30] <burek> and key frame might not get in that time
[20:30] <burek> so you'll get an error about that
[20:31] <burek> increasing analyzeduration you tell ffmpeg to wait a little bit more (and buffer the stream) so it can recognize it finally
[20:31] <mparodi> <burek> mparodi, well, test it :) <-- I know it doesn't work, the question was.. is there an alternative?
[20:32] <burek> I really don't know.. try google mparodi
[20:32] <mparodi> I found unofficial projects that do the same for webm files, maybe I should use one of them
[20:38] <Lupu_> burek, i get an error message saying Unknown VideoCodec: copy
[20:38] <Lupu_> http://pastebin.com/yyM950eL
[20:43] <Lupu_> ffserver tries to identify the VideoCodec string using avcodec_find_encoder_by_name(str)
[20:44] <Lupu_> and i think "copy" is not o valid encoder
[20:46] <burek> hm.. it seems it can't do vcodec copy..
[20:47] <burek> can you check how does ffmpeg deal with -vcodec copy
[20:47] <burek> which "encoder" does it use
[20:47] <burek> and use the same one in ffserver.conf
[22:48] <brontosaurusrex> on debian, what should i do after getting "ERROR: libx264 not found"
[22:48] <brontosaurusrex> trying a git compile
[23:26] <eni> how to convert m3a to mp3 using ffmpeg?
[23:27] <sacarasc> ffmpeg -i foo.m3a foo.mp3
[23:31] <eni> sacarasc, thanks
[00:00] --- Sun Jul  1 2012