[Ffmpeg-devel-irc] ffmpeg.log.20131008

[burek]

[04:09] <allengreen> what's the difference between libfdk_aac and faac? which one is better?
[04:11] <relaxed> libfdk_aac
[04:22] <allengreen> relaxed, sounds like faac is awesome too.
[04:30] <relaxed> allengreen: libfdk_aac is supposed to be much better.
[04:30] <llogan> allengreen: https://trac.ffmpeg.org/wiki/AACEncodingGuide
[04:38] <allengreen> thanks, fellows.
[08:09] <StamEhad> Hi! I have a muxed UDP MPEG-2 multi-program transport stream. How can I use ffmpeg to dump this stream to a .ts file without any transcoding?
[08:20] <luoluoluo> hi, noob question, what does v20544 for ffmpeg mean? what version corresponds to v20544?
[08:21] <sacarasc> luoluoluo: Where did you get that from?
[08:21] <luoluoluo> a private project that uses ffmpeg code
[08:22] <JEEB> I don't think I've seen ffmpeg itself prefix the revision with "v"
[08:22] <sacarasc> No.
[08:22] <JEEB> but if that is indeed a commit number, it is rather old
[08:22] <JEEB> like, years and years old :P
[08:22] <sacarasc> Yeah, it's up in the 50ks now, isn't it?
[08:23] <JEEB> 56k on my sep 15 build I'm using atm
[08:23] <sacarasc> Or is it even higher?
[08:23] <luoluoluo> Thanks guys.  So it's a commit number?
[08:24] <JEEB> most likely at least
[08:24] <luoluoluo> OK, I will try to search the git log for it.
[08:24] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=1f20782c0446ed373d6f915a263756916303cbb4
[08:25] <JEEB> "only" four years old
[08:25] <luoluoluo> thanks, how do you get this?  I searched for a while and cannot find it.  you are so cool
[08:26] <JEEB> I could have searched in the git log from the command line, but in this case I was lazy and just searched the commits for that number via that webui :P
[08:27] <JEEB> anyways, not sure you'll gain anything from getting such an old revision
[08:28] <luoluoluo> ok.  Since this project use it, I have to use it...
[08:28] <JEEB> ugh
[08:28] <JEEB> good luck with that :P
[08:29] <luoluoluo> So the most recent version from it's committing date should be the version corresponding to it, right?
[08:29] <JEEB> what?
[08:29] <JEEB> no
[08:29] <JEEB> also if this svn revision was used then it was used, not any version :P
[08:30] <luoluoluo> Actually I am trying to build it for android ndk, so I am trying to search like "ffmpeg x.x.x android.mk"
[08:30] <luoluoluo> that's why I wanna know the public version corresponding to 20544
[08:30] <JEEB> good luck with that :P
[08:31] <luoluoluo> OK... Thanks JEEB for the hint
[08:32] <JEEB> it's not a version, it's a random point of time as far as I can see. So it's that exact commit. And you'll surely have "extra fun" trying to compile such an old version for NDK or anything else like that
[08:33] <luoluoluo> yeah, "extra fun" indeed.  Caused much head ache to me....
[08:35] <luoluoluo> do you search from here? http://git.videolan.org/?p=ffmpeg.git;a=summary
[09:00] <StamEhad_> Is it possible to directly dump a media stream to a file without transcoding?
[09:26] <relaxed> StamEhad_: ffmpeg -i input -map 0 -c copy output
[09:33] <StamEhad_> @relaxed: thanks! I'll try that.
[11:50] <saschagehlich> hey, whenever I try to compile a video using libvpx as the video codec, I get a floating point exception (Ubuntu 13.04, installed via apt-get): https://gist.github.com/saschagehlich/04f0c9e8814977c3a4ce
[11:53] <relaxed> saschagehlich: you need to compile a more recent version of ffmpeg, or use http://johnvansickle.com/ffmpeg/
[11:58] <saschagehlich> thanks relaxed
[13:47] <WhiteNight> Hi, how can I decode a single h264 frame in my c++ program?
[14:14] <zap0> WhiteNight, are you a unix beardo, basement dweller into nerd BDSM?   if so, try linking in all same libXXXXX  that ffmpeg does.
[14:14] <zap0> WhiteNight, if you shower daily.. and sunlight doesn't scare you..  i'd just invoke ffmpeg and let it do all the work.
[14:32] <durandal_1707> zap0: what a nonsense
[14:35] <Woopa> Hey all. I need some help with ffmpeg. I'm trying to use -filter_complex but all i get is "Unrecognized option 'filter_complex'". This is my ffmpeg output so you can see which version Im using: http://pastebin.com/0RhmUZzf
[14:38] <durandal_1707> Woopa: you need something more recent than 0.10
[14:38] <Woopa> Ok. I have to compile then if I'm not mistaken?
[14:39] <durandal_1707> not at all
[14:39] <durandal_1707> !downloads
[14:39] <Woopa> Ok?
[14:42] <Woopa> durandal_1707: Hmm. Ok.
[14:45] <Woopa> durandal_1707: Crazy that the ppa they recommend for Ubuntu is that outdated.
[14:50] <Woopa> now i get "Unknown input format: 'x11grab'" when im running the downloaded binary. Damn.
[14:54] <durandal_1707> it is easy to compile ffmpeg yourself
[14:55] <Mavrik> hmm, yeah, the static builds don't have X11 support
[14:55] <Woopa> oh ok
[14:56] <Mavrik> so I'm afraid you'll have to build ffmpeg yourself if you need X11 support
[14:57] <Woopa> Compile it is then. :)
[15:02] <durandal_1707> what about this one: http://ffmpeg.gusari.org/static/?
[15:03] <Woopa> I'l try
[15:10] <Woopa> Can't even extract the tar. xD And some builds are 0B in size.
[15:10] <Woopa> Compiling now, hoping that will work.
[15:19] <Woopa> nope still the same version. Wierd.
[15:21] <durandal_1707> Woopa: what does not work?
[15:21] <durandal_1707> did you compiled it yourself?
[15:22] <Woopa> I followed this guide: http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[15:22] <durandal_1707> and you instealled it?
[15:22] <Woopa> sudo make install?
[15:23] <durandal_1707> are you sure you enabled x11grab?
[15:23] <durandal_1707> --enable-x11grab --enable-gpl
[15:23] <Woopa> yep
[15:24] <Woopa> this "ffmpeg 2>&1 | head -n1" tells me that I still use the old version.
[15:24] <mittens> if you didn't pass --prefix to ./configure, its probably installed in /usr/local
[15:25] <Woopa> aha
[15:27] <Woopa> It says "./configure --prefix="$HOME/ffmpeg_build"" in the guide
[15:28] <Woopa> Is that wrong?
[15:30] <Woopa> mittens :)
[15:36] <Woopa> I got a folder called "ffmpeg_build".
[15:36] <Woopa> in ~/
[15:39] <Mavrik> then your compiled ffmpeg is in ffmpeg_build/bin
[15:39] <Woopa> oh ok
[15:40] <Woopa> which step should i alter to install it more proper? In bin, i guess.
[15:41] <Mavrik> first uninstall the old one
[15:41] <Mavrik> and if you don't pass --prefix to configure, it'll install into /usr/local which is in Ubuntu default search path
[15:42] <Woopa> So I'l just run ./configure?
[16:04] <Woopa> sort of got it now.
[16:04] <avih> hi, could i use ffmpeg to create a virtual encoded file with random access from an arbitrary source to then be served over http (e.g. to mplayer or a flash player in a browser or webm in a browser). essentially, trying to create encoded chunks of fixed size which could be concatenated into a playable stream (even if not fully valid) , and with random access without transoding the entire video prio to the access point (but some prior decoding is acceptable).
[16:04] <avih> it doesn't have to support many clients, and assuming the cpu is enough to encode at least 5x real time at my selected parameters. could this be achieved?
[16:04] <Woopa> But i need to compile it with pulse. "--enable-pulse"?
[16:05] <avih> and i could use extra tools, e.g. to pad each chunk with 0 at the end, etc
[16:06] <avih> the part with which i look for help is the chunks with random acess using ffmpeg
[16:09] <avih> also, i don't mind if it's a theoretically non-seekable container as long as the player could still somehow seek it
[16:20] <Mavrik> avih, it looks like you need MPEG2-TS stream :P
[16:20] <Mavrik> doing broadcasting? :)
[16:24] <durandal_1707> Woopa: it will not enable pulse unless you use that flag
[16:38] <avih> Mavrik: yeah, already started playing with just that :) thanks
[16:39] <avih> Mavrik: can i create fixed-size chunks which could be concatenated into a playable and seekable stream? (Assuming i create a virtual file on the server on the fly according to range requests by the client)
[16:41] <avih> Mavrik: not broadcasting. more of a media server of my local videos over hard network (socks, limited bacdwidth)
[16:42] <avih> bandwidth*
[16:42] <avih> Mavrik: and just for fun ;)
[16:42] <Mavrik> yes
[16:42] <Mavrik> M2TS is built for broadcasting, so you can cut it
[16:43] <Mavrik> of course you need to cut on 188B packet boundaries
[16:43] <Mavrik> and it helps if you cut on I-frames with PSS/SPS packets for H.264 streams
[16:44] <avih> i'm flexible with the formats etc, as long as i can play a transoded on-the-fly and seek over a network
[16:44] <Mavrik> seeking& is a problem
[16:45] <avih> yeah
[16:45] <Mavrik> you'll need a proper streaming server for that
[16:45] <Mavrik> and of course a client that knows how to do it
[16:45] <Mavrik> I had very good experience with Wowza and RTMP protocol for that
[16:45] <avih> which is why i need to create a virtual file, so the server can present it as if it knows all the data in advance, but encode the chinks on the fly
[16:45] <Mavrik> playing static mp4 files
[16:46] <avih> rtmp might be hard to play over socks
[16:46] <avih> but i don't mind an off the shelf solution. but so far i couldn't find one
[16:46] <avih> (tried ps3 media server, lix media, and others)
[16:47] <Mavrik> RTMPT works over socks IIRC
[16:47] <avih> vlc kinda works, but its ui is terrible, and seek can't be done from the client (but it's workable with a remove desktop)
[16:47] <avih> remote*
[16:48] <avih> and creating such server would be fun ;)
[16:48] <bencoh> its a bit of work
[16:48] <avih> it is.
[16:49] <avih> but depending on possibilities , it might be easier for a subset of the overall functions of such media server. right now i'm focusing on a virtual served transcoded-on-the-fly video
[16:49] <bencoh> nginx-rtmp supports seeking under certain conditions
[16:49] <Mavrik> transocidng on the fly with seeking is not really trivial
[16:49] <avih> right
[16:49] <Mavrik> doable, but you'll have to learn alot :)
[16:50] <avih> heh
[16:51] <avih> i was thinking of, e.g. if using a 100K/sec overall, then encode arbitrary 5s chunks, then pad them with 0 to some bigger size which would hold all such 5s transcodes with tight cbr
[16:52] <avih> so then if the client asks for bytes 1M-2M, these would be chunks 2,3 (0 based), which all have fixed size
[16:53] <Mavrik> that works
[16:53] <avih> but for this i'll need a format+container which i could pad and concatenate, and that it stays playable with whatever player (preferably mplayer or vlc, for instance)
[16:54] <Mavrik> well, I already told you which one that is
[16:54] <avih> yeah
[16:54] <Mavrik> MPEG2-ts was made for your use case
[16:54] <Mavrik> it supports padding as well
[16:54] <Mavrik> (not padding by zeros, but there are padding packets in the standard that are listened to by most clients)
[16:54] <Mavrik> of course, not all players support TS, but alot of them do
[16:54] <avih> so how would i encode the 5s chunk which starts at 100s at the source with mpeg2-ts?
[16:54] <avih> (and which codecs would fit into that container?)
[16:55] <Mavrik> pretty much all MPEG codecs work
[16:55] <Mavrik> DVB-T e.g. currently puts either H.264/AAC or MPEG2/MP2 into it for broadcasting
[16:55] <Mavrik> I suggest you stick with H.264/AAC for compatibility and efficiency
[16:55] <avih> right
[16:55] <avih> yeah, sounds reasonable
[16:56] <avih> and can i transcode an arbitrary location at the source with relatively fast seek using ffmpeg? (preferably better seek than O(N), like O(log N) would be nice)
[16:57] <avih> (and yeah, padding a non-0 pattern would be fine)
[16:57] <Mavrik> seeking in ffmpeg is pretty fast always if GOP size is reasonable
[16:58] <Mavrik> you'll probably just have to pad chunks manually, not sure if ffmpeg supports that
[16:58] <avih> well, i don't have control over the sources, as they're very varied
[16:58] <avih> i can pad manually, np
[16:58] <avih> well, with some code :)
[16:59] <avih> Mavrik: so for seek, should i use -ss on input? or output? the docs say the output is more accurate but slower. i guess i need accurate if i don't want duplicate frames etc?
[17:00] <Mavrik> avih, if you put it on output ffmpeg will decode whole file until -ss
[17:00] <Mavrik> if you put it in input it'll jump to approximate position, backtrack to first I-frame it can find and decode up until the exact point
[17:00] <Mavrik> if you have a reasonable GOP (which you should), there's little reason of using -ss in output
[17:02] <avih> right. sounds reasonable
[17:02] <avih> so it would be roughly O(log n) + O (M) where N is the seek position and M is GOP size, right?
[17:02] <Mavrik> mhm
[17:02] <Mavrik> for most files it's a very small value
[17:03] <avih> so if the disk IO is relatively fast, can i expect seek time of under 2s for, say, 1G source file of 2 hours?
[17:03] <Mavrik> mhm
[17:03] <Mavrik> for non-broken files it's very fast
[17:03] <avih> just to get a feel of the seek time numbers
[17:04] <Mavrik> especially if they have constant framerate
[17:04] <avih> yeah, i think most files would have constant frame rate... i _think_
[17:06] <avih> do i want mpegts or mpegtsraw?
[17:15] <avih> Mavrik: so, this kinda works (plain concat): ffmpeg  -ss 0  -i source.mp4 -t 5 -vcodec libx264 -preset faster -vf scale=480:320:-1 -acodec libmp3lame -ab 32k -b:v 268k -f mpegts  chunk0.mp4
[17:16] <avih> i made 3 chunks for 0s, 5s, 10s, concatenated them (no padding), and it plays, but for some reason in double frame rate..
[17:16] <avih> does this seem like a reasonable start?
[17:17] <avih> also, the player (mplayer) seem to get confused with the video length
[17:17] <Mavrik> use .ts extension for starters
[17:17] <Mavrik> everything else looks ok
[17:18] <Mavrik> yes of course, video length won't be valid :)
[17:18] <Mavrik> you probably have to set constant framerate with -r
[17:18] <avih> hmm.. will try both. this is fun :p
[17:19] <avih> do i want mpegts or mpegtsraw?
[17:20] <Mavrik> mpegts, the other thing is for internal use
[17:20] <Mavrik> also, use h264_mp4toannexb filter
[17:20] <Mavrik> some mp4 files will require it
[17:20] <Mavrik> (it's a bitstream filter)
[17:20] <avih> for the source file?
[17:21] <Mavrik> when encoding the TS
[17:21] <avih> does it effect decoding in ffmpeg? or encoding?
[17:32] <bencoh> bitstream encoding/muxing
[17:36] <avih> hmm.. i'm getting some duplicate audio chunks there, and also the video gets stuck at some stage (trying concat of 6 chunks).
[17:36] <avih> can i use -r to set the output to use the input frame rate?
[17:43] <avih> tried another video. similar symptoms: faster than expected frame rate and audio is not in sync, and also with duplicate audio segments.
[17:43] <avih> i'm using this command to encode a single chunk:
[17:44] <avih> ffmpeg  -ss 3000 -r 23.976 -i source.mp4  -t 5 -vcodec libx264 -preset superfast -vf scale=480:320:-1 -acodec libmp3lame -ab 32k -b:v 268k -r 23.976 -f mpegts -bsf:v h264_mp4toannexb chunk0.ts
[17:45] <avih> (symptoms are when playing concatenated few chunks, each encoded as above)
[17:46] <avih> Mavrik: with the above command, can i expect the audio to align decently?
[17:46] <Mavrik> there's no reason for audio to be out of sync, neither for video to run too fast
[17:46] <Mavrik> unless your player is broken
[17:47] <avih> i can try another player, but usually mplayer is pretty dependable...
[17:49] <avih> vlc doesn't play with fast framerate. but it looks as if the chunks are not fully accurate.
[17:50] <avih> (just tried also with pcm audio)
[17:51] <avih> i could use avisynth as source. it creates an uncompressed virtual file, so maybe it'll be easier for ffmpeg to cut the chunks correctly..
[18:55] <durandal_1707> teratorn: but you cant use amovie inside -vf ....
[18:55] <teratorn> durandal_1707: oh dear
[18:56] <teratorn> OK time to -filter_complex
[18:56] <durandal_1707> teratorn: you can use -af
[18:56] <durandal_1707> -af amovie ....
[18:56] <teratorn> durandal_1707: ah
[18:56] <durandal_1707> that should work
[18:58] <teratorn> durandal_1707: I get no audio output stream at all...
[19:00] <teratorn> ffmpeg -f lavfi -i color=size=1280x720 -vf '...' -af 'amovie=source1.mp4'
[19:01] <teratorn> it runs but just doesn't produce an audio stream
[19:05] <teratorn> durandal_1707: any clue?
[19:17] <durandal_1707> teratorn: what version you use?
[19:17] <teratorn> durandal_1707: latest
[19:17] <teratorn> durandal_1707: I figured it out one way with just, -f lavfi -i amovie=source1.mp4
[19:18] <teratorn> I would prefer to just acodec copy the audio, but meh
[19:18] <durandal_1707> -c:a copy
[19:20] <durandal_1707> but, yes, i think you need for copy to have audio in -i INPUt
[19:22] <durandal_1707> but using filter_complex should be much friendlier
[19:23] <durandal_1707> you specify all inputs with '-i inputX'
[19:24] <teratorn> yeah...
[19:24] <teratorn> durandal_1707: anyway I got it sorted. thanks!
[19:33] <l1x> hi
[19:33] <l1x> http://pastebin.com/3itcBs4m
[19:34] <l1x> can somebody tell me how to instruct ffmpeg to set the pix_fmt to the correct value instead of -1?
[19:35] <teratorn> durandal_1707: OK, problem... ffmpeg doesn't stop after exhausting the first movie... it just keeps pressing the last frame(s) forever and ever... any clue ? :-(
[19:38] <teratorn> durandal_1707: sure I will if adding -t <time> doesn't fix it :)
[19:39] <durandal_1707> -shortest
[19:41] <Hurons> Hey
[20:20] <Hurons> Anyone on that knows a bit about ffmpeg dshow for decklink cards? (HDYC uyvu422 support)?
[20:24] <newguy123> anyone able to help a new guy with a basic restream command?
[20:27] <llogan> newguy123: what are you trying to do?
[20:29] <newguy123> I am trying to transcode and it was locking up so I tried just to do a simple restream and it does the same thing    ffmpeg -re -f rtsp -rtsp_transport tcp -i rtsp://media-us-2.soundreach.net/s lcn_sports.sdp -vcodec copy -acodec copy -f rtsp rtsp://127.0.0.1/stream.sdp
[20:57] <Hurons> Trying to capture from a decklink card using fdshow under windows. But cant get passed this buffer full error. Anyone have a clue? See : http://pastebin.com/XCDKxTV0
[21:02] <plm> Hi all
[21:04] <plm> ffmpeg does it? Or are there a server that split the stream into two or more streams and sends one over each link and in other side gets stream pieaces and join again? I would like hav ea box with many 3g cards and split video over each one and other side join it and show..
[21:05] <avih> guys, i'd appreciate some help in forcing input video frame rate. for some reason, ffplay plays it correctly, but when i encode it with ffmpeg, i get twice as fast framerate (but audio plays normally, so audio comes out twice as long as the video)
[21:06] <avih> i tried -r for the input or output, and also tried -vf fps=fps=25 for output, and still the player (mplayer) plays with double frame rate
[21:06] <newguy123> http://pastie.org/8387463 <--- commands I have tried for restream
[21:12] <avih> llogan: my scenario is complex and involves more than one file. my specific question was about forcing framerate of the input and/or output video with ffmpeg.
[21:14] <llogan> good luck
[21:16] <Hurons> Llogan, is using fdshow and/or windows with ffmpeg a strange thing? Cuz I've been pulling my hairs out so far. And stackoverflow, and the ffmpeg-user mailing dont seem to have any answers. Feels like I'm going about this the wrong way or something.
[21:17] <avih> llogan: https://pastebin.mozilla.org/3219172
[21:18] <avih> the source is an avisynth script, on windows.
[21:20] <avih> apparently the source is correctly recognized as 25fps, but when played with mplayer, the info it gives is that it's 50fps, so the video plays twice as fast and stops when the audio (which plays normally) is still playing
[21:25] <llogan> Hurons: you could ask on ffmpeg-user
[21:26] <Hurons> If you mean the mailing list, I have done so a few days ago. I will be a bit more patient, thanks.
[21:27] <llogan> oh, i see it. sorry, but i don't have an answer for you (i have no decklink experience)
[21:28] <Hurons> No worries. It's just frustrating since I have no audio/video/ffmpeg experience. So it's like a needle in a haystack atm.
[21:29] <llogan> avih: do other players also have trouble with it, or only mplayer?
[21:31] <avih> llogan: i just tested with vlc and directshow based (mpc) and both played correctly. however, my actual scenario is one which only mplayer can handle (of those players)
[21:32] <avih> so i need it to work with mplayer. also, appreciating mplayer as the mothre of all players, i want it to work properly with it regardless
[21:32] Action: llogan blames mplayer
[21:33] <avih> llogan: so, would you help me with the ffmpeg syntax to force input and/or output frame rate, please?
[21:33] <llogan> if it helps you can refer to the -r option and/or the fps video filter
[21:33] <llogan> http://ffmpeg.org/ffmpeg-filters.html#fps
[21:34] <llogan> but i don't know why mplayer is behaving like that
[21:34] <avih> as i said, i already tried -vf fps=fps=25 per the example from that section
[21:34] <llogan> the answer for when i don't know, however, is "try a newer (mplayer) build"
[21:34] <avih> but maybe i got the syntax wrong somehow
[21:35] <llogan> oh, i missed that
[21:35] <relaxed> does ffplay play it too fast?
[21:35] <avih> checking
[21:35] <llogan> you said "ffplay plays it correctly" so i assumed you meant the output
[21:36] <relaxed> If so, try mplayer -vfm ffmpeg input
[21:36] <avih> i meant for the input. but now i also tried the output, and it's also fine.
[21:36] <avih> (ffplay plays the output fine)
[21:37] <avih> but my ultimate usecase is ugly concatenation of ts files, which i expect to play decently. and neithre ffplay, not vls or mpc could play the concatenated file. mplayer can. but it has double frame rate (eith in a single chunk or with the concatenation)
[21:38] <llogan> is that why you are using avisynth, to concat?
[21:39] <avih> no, i use avisynth to get frame accurate and sample accurate chunks
[21:39] <avih> is i use -ss 1000 -i source.mp4 -t 10 and then anothre chunk which starts at 1010, and then concat the outputs, i get overlapping audio
[21:41] <avih> and the video is also messed up. the avisynth script (funnily, with ffmpegSource2), produces accurate chunks which after encoding concatenate to a decently continuous video and audio
[21:42] <relaxed> -ss after the input will be more accurate
[21:42] <relaxed> if all else fails try tsmuxer
[21:42] <avih> yeah, but i need random access to chunks, and -ss for the output will decode whatever happens before that timetamp, which i cannot do
[21:43] <avih> from performance point of view
[21:43] <avih> the avisynth source indexes the file once and caches the iframes index, so seeking is both fast and frame accurate
[21:44] <avih> (i'm trying to build a transcoding server which transcodes on the fly and on demmand, with seek support, by exposing a virtual transcoded file, whose chunks i generate on the fly)
[21:45] <durandal11707> avih: what ffmpeg version you use?
[21:45] <avih> from the past few days
[21:45] <avih> ffmpeg version N-56930-gee3d03b
[21:47] <avih> so i have two problems. only mplayer can play the concatenated mpeg2-ts chunks decently, and mplayer plays it with double frame rate whether it's concatenated or not.
[21:48] <avih> (eventually the chunks will be padded to a fixed size such that the virtual transcode is consistent and predictable, but for now i just play with the concat)
[21:55] <Hurons> I'm out, thanks for the conversation Llogan.
[22:01] <relaxed> avih: did you try mplayer -vfm ffmpeg input?
[22:01] <avih> relaxed: no. not sure what that means. is this an ffmpeg option? or an mplayer one?
[22:02] <relaxed> mplayer
[22:02] <avih> will try now
[22:02] <relaxed> if that doesn't work try "mplayer -demuxer lavfpref input"
[22:03] <relaxed> The former forces ffmpeg's decoders, the latter its demuxers
[22:04] <relaxed> try both for shits and giggles
[22:05] <avih> didn't work. same double frame rate
[22:05] <avih> gonna try the demux now
[22:06] <newguy123> does anyone know why this simple command doesn't work?   ffmpeg -re -f rtsp -rtsp_transport tcp -i rtsp://media-us-2.soundreach.net/s lcn_sports.sdp -vcodec copy -acodec copy -f rtsp rtsp://127.0.0.1/stream.sdp
[22:07] <relaxed> -simple
[22:07] <avih> relaxed: the -demuxer lavpref option doesn't play the file. it just says "end of file" and that's it
[22:07] <relaxed> newguy123: there's a space in the input name?
[22:08] <relaxed> if it belongs there use quotes
[22:08] <durandal11707> newguy123: what it does? and what it should instead do?
[22:08] <relaxed> avih: can you stick a small sample of the video up somewhere?
[22:08] <avih> relaxed: sure
[22:10] <avih> relaxed: i use this command to encode it, and it plays fine in vlc, mpc and ffplay, but not in mplayer: ffmpeg  -ss 1025 -i source.avs  -t 30 -vcodec libx264 -preset superfast -vf scale=480:320:-1 -acodec libmp3lame -ab 96k -b:v 268k -f mpegts -bsf:v h264_mp4toannexb 6.ts
[22:10] <avih> (uploading now)
[22:11] <avih> relaxed: http://dropcanvas.com/jig0z and ignore the incorrect aspect ratio
[22:13] <newguy123> for some reason ffmpeg seems to freeze after it displays the stream title audio and video codec
[22:13] <avih> relaxed: ultimately i'm trying to create a playable file by concatenating few of those, but only mplayer plays the concatenation, so i need to either fix the concatenation, or make mplayer play it properly (regardless of concatenation)
[22:14] <relaxed> avih: mplayer2 and mpv, both forks of mplayer, play it correctly
[22:14] <avih> relaxed: you mean mpc?
[22:15] <relaxed> the original mplayer may too...are you using the latest svn version?
[22:15] <avih> (didn't try with mplayer2 yet)
[22:15] <avih> quite recent. the one which comes with smplayer for windows from maybe a month or two ago
[22:15] <relaxed> no, I mean http://mpv.io
[22:15] <relaxed> I recommend mpv out of the three
[22:17] <avih> iirc mplayer2 couldn't be built for windows for some months now, and i've never tryed mpv. are there pre-built windows mpv binaries?
[22:18] <relaxed> http://mpv.srsfckn.biz/
[22:19] <hernantz> which could be the ffmpeg alternative to this command? avconv -i input.avi -c:v copy -c:a aac -strict experimental -b:a 13k output.mp4
[22:21] <relaxed> hernantz: they take the same options pretty much
[22:23] <relaxed> avih: how does mplayer fit into this puzzle?
[22:24] <avih> relaxed: mpv seems to work :) thanks! what do you mean about mplayer?
[22:24] <avih> (mpv also plays the concatenation flawlessly! :) )
[22:25] <relaxed> more work here is done
[22:25] <avih> relaxed: :) but what did you mean about mplayer?
[22:26] <relaxed> never mind, you were just curious why it didn't work with mplayer.
[22:26] <avih> i still am. do you know the answer?
[22:27] <relaxed> most likely there's a bug in the version you're using.
[22:27] <avih> though the concatenation of the mpeg2-ts encoded chunks doesn't seem to be a valid video file, which is suboptimal.
[22:27] <avih> tried 2 mplayer versions, few months apart
[22:27] <relaxed> are you using ffmpegconcat:
[22:28] <relaxed> ffmpeg's concat:
[22:28] <avih> mpv manages to work around it, but so far it's the only player capable of playing it
[22:28] <avih> not ffmpeg concat. i'm trying to ceate individual encoded chunks to be concatenated on the fly without ffmpeg
[22:29] <avih> but use ffmpeg to create those chunks
[22:29] <BoomerBile> what file do i need to include to use av_open_input_file
[22:31] <BoomerBile> ?
[22:36] <BoomerBile> alright forget that question, this is a linking error.. hmm
[22:44] <avih> relaxed: you got any idea how to encode chunks with ffmped which i could later concatenate into a valid file which most players could handle?
[23:15] <sprite`> question: i'm having some audio distortion/skipping issues throughout videos when i use libfaac with h264...doesn't seem to happen with libmp3lame but if i use that the audio doesnt play in our flash player so i'm at kind of a loss (input/output here: http://i.imgur.com/28647TJ.png ) ... any ideas why this may be happening?
[23:30] <sprite`> http://pastebin.com/dGQhPdga
[23:31] <llogan> sprite`: does the input file play fine in your player?
[23:32] <sprite`> yes
[23:32] <sprite`> no audio skipping
[23:33] <llogan> why do you not stream copy the audio instead of re-encoding? (-codec:a copy)
[23:33] <llogan> why use -async 1?
[23:34] <llogan> is there distortion with another AAC encoder? (-c:a aac -strict experimental)
[23:34] <sprite`> i do use copy right now thats the only way ive been able to do it...but some of our raw audio is quite large and im trying to support lower bandwidths
[23:34] <sprite`> async 1 i started doing because it was out of sync
[23:35] <sprite`> i also a lot of times get inputs that dont support the containers
[23:35] <sprite`> i havent tried that
[23:37] <sprite`> trying -c:a aac -strict experimental
[23:39] <sprite`> well the distortion seems to be gone with that haha
[23:39] <sprite`> so is my libfaac messed up
[23:43] <sprite`> plays in the player too
[23:44] <sprite`> well thanks for the help...any reason i shouldnt use this as opposed to libfaac?
[23:51] <rmelo> high guys
[23:51] <rmelo> hi
[23:51] <rmelo> lol
[23:51] <rmelo> sorry
[23:53] <rmelo> When i use the -ss xx:xx:xx -t xx:xx:xx the splitted video always starts at beggining of the movie
[23:53] <rmelo> any idea why?
[23:54] <llogan> sprite`: the quality isn't as good at the same bitrate. https://trac.ffmpeg.org/wiki/AACEncodingGuide
[23:54] <llogan> what fplash player are you using? what browser?
[23:55] <sprite`> it's custom made
[23:55] <sprite`> firefox
[23:55] <llogan> do other flash players have the same issue?
[23:56] <llogan> you can quickly try jw player via their wizard
[23:56] <sprite`> oh really
[23:56] <llogan> http://www.longtailvideo.com/jw-player/wizard/
[23:56] <sprite`> theirs plays the mp3 ones
[23:57] <llogan> i meant the aac stutter issue
[23:57] <llogan> s/stutter/distortion
[23:57] <sprite`> oh..yeah the stutter issue is the same even if i play it in vlc or in the browser itself lol
[23:57] <llogan> can you provide the input file?
[23:58] <rmelo> fflogger: sorry, here it comes
[23:58] <rmelo> http://pastie.org/8387921
[23:58] <llogan> sprite`: is it distorted in ffplay too?
[23:58] <sprite`> http://stream05.trillhd.com/trill/4636_480_90vl2ljf7t.mp4
[23:59] <llogan> rmelo: do you mean that ffmpeg seems to be ignoring your -ss?
[23:59] <rmelo> exactly
[00:00] --- Wed Oct  9 2013