[Ffmpeg-devel-irc] ffmpeg.log.20130209

[burek]

[00:34] <stephanedev> hi everybody. i have issues when streaming a video file using rtp and playing it with ffplay. it works fine when the 2 hosts are on a LAN, but the video is very jerky and most frames are dropped when doing it over the internet
[00:35] <stephanedev> the file contains a vp8 video stream 640*360, 25 fps, 500 kb/s
[00:36] <stephanedev> ffplay is passed in the sdp file made of the ffmpeg output in the terminal
[00:37] <stephanedev> command is: ffmpeg -re -i big_buck_bunny.webm -vcodec copy -an -f rtp rtp://<target_ip>:50004
[00:39] <stephanedev> in the ffplay output, there are a lot of messages like:
[00:39] <stephanedev> [vp8 @ 026bc380] RTP: missed 3 packets
[00:39] <stephanedev> [sdp @ 026a3e00] Missed unknown data, sequence broken
[00:41] <stephanedev> any idea as to what may be wrong? am i missing some command-line option to ffplay?
[00:42] <stephanedev> is it expected to receive the sdp error about a broken sequence?
[00:49] <Mavrik_> stephanedev: well, obviously it's telling you that you're missing packets
[00:49] <Mavrik_> either your bandwidth isn't large enough
[00:49] <Mavrik_> or you're having ordering problems
[00:59] <stephanedev> Mavrik_: thanks for the answer
[01:00] <stephanedev> bandwidth shouldn't be an issue. downloading and video streaming using skype works just fine between the 2 endpoints
[03:58] <portal`> hey guys - http://pastebin.com/RzV4yvgA - trying to transcode HTTP Live Stream to WMV, it cuts off after about 9-20 seconds.
[04:52] <st000> hello guys is anyone of you familiar with the problem that a from ffmpeg created (avi) file has the right aspect ratio in ffplay, but not in vlc or windows media player?
[05:24] <Sam__> Guys i am doing simple flv transcoding using libx264+ffmpeg using single thread, when i see CPU usage using top i could see it cross 112%
[05:24] <Sam__> and can see 4 instances of ffmpeg
[05:25] <Sam__> 214358 root      20   0  300m  23m 4788 R 90.3  0.5   0:07.17 ffmpeg 214361 root      20   0  300m  23m 4788 S  5.0  0.5   0:00.35 ffmpeg 214359 root      20   0  300m  23m 4788 S  4.4  0.5   0:00.34 ffmpeg 214360 root      20   0  300m  23m 4788 S  4.4  0.5   0:00.35 ffmpeg
[05:26] <Sam__> http://pastebin.com/TX5GqjWz
[05:27] <st000> llogan: what do you mean with !pb ?
[05:29] <llogan> st000: that is a shortcut to tell the #ffmpeg channel bot, fflogger, to tell you to, "please use a pastebin site..."
[05:30] <Sam__> any suggestions please
[05:30] <llogan> you could always...
[05:31] <st000> llogan okay^^
[05:32] <st000> llogan: ehm here my pastbin http://pastebin.com/yHve1nzZ
[05:35] <llogan> st000: not that it really matters, but your input is already 704 wide, so why add scale?
[05:36] <Sam__> http://pastebin.com/fhqMPuRz
[05:36] <Sam__> @ fflogger
[05:36] <Sam__> @fflogger http://pastebin.com/fhqMPuRz
[05:36] <elkng> anyone around ?
[05:37] <llogan> fflogger is a bot, not a person
[05:37] <st000> llogan: it was a try
[05:37] <Sam__> and CPU usage in Pasebin http://webchat.freenode.net/
[05:37] <Sam__> http://pastebin.com/TX5GqjWz
[05:38] <st000> even if i let it go it will not change anything
[05:39] <llogan> what does vlc or WMP think the aspect ratio is?
[05:39] <st000> 4/3
[05:39] <st000> or 1/1
[05:40] <st000> i think it is the last thing
[05:40] <llogan> do you have a sample?
[05:41] <st000> yes
[05:41] <Sam__> @<fflogger> any suggestions?
[05:42] <st000> it is crud it anything between 5:4 and 4:3
[05:45] <klaxa> Sam__: suggestion 1) update your ffmpeg 0.10 is pretty damn old
[05:45] <klaxa> suggestion 2) add: -threads 1
[05:45] <Sam__> -threads 1 is already add in commandline
[05:46] <klaxa> probably add it before -x264ops
[05:46] <klaxa> *x264opts
[05:46] <klaxa> also i'm not sure that's supported anymore
[05:46] <klaxa> i could be wrong though
[05:47] <Sam__> no effect adding it before x264opts as well
[05:49] <klaxa> mmh dunno
[05:49] <Sam__> @ <klaxa> I disagree with you here about options as these all are latest options for both
[05:50] <Sam__> ffmpeg and libx264
[05:50] <klaxa> your ffmpeg version is 0.10
[05:50] <klaxa> that's, if i may say so, old as fuck
[05:52] <Sam__> ok
[05:54] <klaxa> grab the latest ffmpeg and take a look at this http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide
[06:02] <gudy> Hi, is it valid for a GOP to not start with an I frame, but with B frames which reference a future I frame. The I frame is first in the decode sequence, but not first in terms of presentation time ?
[06:03] <gudy> (in h264 high profile)
[06:12] <st000> llogan: do you want to have the sample ?
[08:28] <jac1d> Hello all.  I have a number of PCM streams coming from FM tuners to a single x86 CPU and I want to encode them to AC3 stereo in real time.  What kind of CPU power would I need to do this?
[09:42] <int> hello guys, I want to compress video file a bit without a big quality loss, but I receive the followint error: http://paste2.org/p/2841453
[09:42] <int> could somebody help me?
[09:42] <int> maybe I should change codec?
[09:45] <relaxed> int: ffmpeg -i avn1.mkv -c:a libmp3lame -b:a 128k -c:v libx264 -preset veryslow -crf 18 output.mkv
[09:45] <sacarasc> Why are you using h263?
[09:45] <int> sacarasc: I thought it is the best for compression
[09:46] <int> relaxed: thank you very much
[09:46] <sacarasc> H264 is what you wanted, using libx264.
[09:46] <int> sacarasc: hm
[09:46] <relaxed> The source is h264, so I doubt you'll gain much more compression.
[09:46] <int> ok, I will try
[09:47] <int> does redusing resoultion help me?
[09:48] <sacarasc> Yes, it would.
[09:53] <int> -s 1024x570?
[10:17] <sacarasc> -vf scale=1024:570 I think is the proper way these days.
[10:20] <int> ok, thank you
[11:08] <creep> h
[11:08] <creep> another nice nuclear meltdown in result of stupidity from japan, have a good appetite for fish from sea  http://en.wikipedia.org/wiki/Lists_of_nuclear_disasters_and_radioactive_incidents
[11:10] <elkng> creep: when was the latest ? in fukushima ?
[11:10] <elkng> or there is new incident ?
[11:12] <creep> elkng<< about a year ago, you know, it taskes some time to share this unimportatnt information :)
[12:02] <sceriffo> hi , where can I find a code for generation extradata for AAC decoder ?
[12:03] <creep> elkng<< soon, chinese will start selling radioactive flashlights, then they will be able to get rid of their radioactive waste legally, no recharge, 100 years of light, cheap, made in china...
[12:13] <elkng> creep: "radioactive flashlights", its creepy
[12:29] <iive> creep: funny. but not really accurate.
[12:29] <iive> all flashlights are radioactive, because light is radioactive radiation :)
[12:35] <Mavrik> hmm, does anyone know what "afterburner" parameter on Fraunhofer AAC encoder actually does?
[12:36] <JEEB> seemingly optimizes more on quality than speed
[12:36] <JEEB> what it *actually* does tho is something you'll have to ask wbs
[12:38] <Mavrik> hmm
[12:38] <Fjorgynn> seems like I
[12:38] <Mavrik> basically I have a Motorola STB here which chokes as soon as I enable afterburner
[12:39] <Mavrik> JEEB, wbs?
[12:41] <JEEB> IIRC that's the IRC nickname for Martin Storsjö, who made fdk-aac from an Android thing to a library :P
[12:41] <JEEB> and also who pushes fixes towards back to Android
[12:41] <Mavrik> ah, ok :)
[13:05] <DeLoreann> Hi! I have a problem with ffmpeg. While recording rtmp stream from ustream, quality decreases during recording. Anyone? (sorry for my english ;)
[13:05] <DeLoreann> Here is a sample: http://bocik.pl/ss/player.php?v=njIdVI.flv
[13:18] <burek> ive just set up ffmpeg as a p2p voice tool :)
[13:18] <burek> btw, which audio codec is the most suitable for p2p voice chat
[13:18] <burek> (the least delay, speech codec)
[13:18] <JEEB> the one mumble uses, opus
[13:21] <burek> hmh, it doesnt produce any errors
[13:21] <burek> but sending it over udp with -f mpegts doesnt work not even in vlc
[13:21] <burek> i mean, i cant hear anything
[13:22] <JEEB> are you sure vlc is equipped with libopus decoding? are you sure its muxing into mpeg-ts is somehow supported by Various Things?
[13:22] <burek> oh
[13:22] <burek> how should i send it then?
[13:22] <burek> i can use ffplay
[13:23] <JEEB> to be honest I have no idea whatsoever, I think mumble just sends out a raw opus stream or whatever.
[13:23] <JEEB> not sure if ogg has a set up mapping for opus, matroska's one is still in the works
[13:25] <burek> ill try ogg too, thanks :)
[13:25] <JEEB> as the container, that is
[13:25] <JEEB> or just raw opus
[13:25] Action: JEEB shrugs
[13:26] <burek> how to specify -f raw
[13:28] <DeLoreann> Mavrik: http://pastebin.com/SSjM5Gx9
[13:28] <JEEB> burek, see -formats for opus
[13:28] <burek> thanks
[13:29] <Mavrik> DeLoreann, your quality drops because you're re-encoding video to 200kb/s
[13:29] <Mavrik> DeLoreann, use "-codec copy" as parameter
[13:29] <Mavrik> to force only copy of stream, not reencode
[13:30] <DeLoreann> "Unrecognized option 'codec'"
[13:30] <Mavrik> DeLoreann, ah, you have an older version, it's "-vcodec copy"
[13:31] <DeLoreann> [flv @ 0x808040020] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 100 >= 0
[13:31] <DeLoreann> av_interleaved_write_frame(): Invalid data found when processing input
[13:32] <DeLoreann> output file: http://bocik.pl/ss/player.php?v=abfwcZ.flv
[13:32] <DeLoreann> 51kB, (one frame?)
[13:33] <Mavrik> hmm
[13:33] <JEEB> DeLoreann, update your ffmpeg
[13:33] <Mavrik> that shouldn't happen really... but you do have a rather old fmfpeg
[13:33] <Mavrik> *ffmpeg
[13:33] <JEEB> yes, it's /very/ old
[13:34] <DeLoreann> im doing it on shell account, this will take time :)
[13:34] <burek> JEEB, sorry for bugging you, but what exactly should i be looking in the -formats ?
[13:34] <DeLoreann> anyway, thanks Mavrik!
[13:34] <JEEB> it should have a raw format for opus, if there isn't... then there isn't :D
[13:34] <burek> there isn't :)
[13:34] <Mavrik> DeLoreann, grab static builds :)
[13:35] <JEEB> Mavrik, he's on freebsd
[13:35] <JEEB> lol
[13:35] <Mavrik> oh.
[13:35] <Mavrik> ah.
[13:35] <Mavrik> that could be problematic then :D
[13:35] <JEEB> burek, well then try -f ogg or something, can't really come up with anything better :D
[13:36] <burek> ok, i'll try various test and see which one works then, thank you :)
[13:42] <someone-noone> Hello! I'm trying to pass "magic cookie"(extradata) data to mac os Core Audio framework for decoding AAC format. File is in m4a format. For this I'm using extradata from my AVCodecContext. But extradata_size is 7, however if I read "magic cookie" using OS X API, it's size is 44 for same file. Looks like magic cookie != extradata in ffmpeg (but should be). Any help how can I extract this data from ffmpeg?
[16:24] <elkng> anyone around
[16:24] <elkng> ?
[16:26] <elkng> that is the file http://sprunge.us/QQeM I'm trying to convert, that is the command http://sprunge.us/MAgd I use to convert it, that is the output http://sprunge.us/FHRa, it's length is 23:19 but it converts only first 4:57 and then stoped convert
[16:26] <elkng>  with errors, what is wrong ?
[16:27] <elkng> and similar problems happened for last couple days about 4 times, it converts first few minutes and then stopped with errors
[16:31] <durandal_1707> elkng: does number of channels really change to that big number?
[16:31] <elkng> what number ?
[16:31] <durandal_1707> in log, ich
[16:32] <elkng> ffprobe says: "Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 192 kb/s"
[16:32] <durandal_1707> [SWR @ 0x8a20640] Value 24.000000 for parameter 'ich' out of range
[16:32] <durandal_1707> Unsupported number of input channels
[16:32] <elkng> something wrong with video ?
[16:33] <durandal_1707> maybe something get corrupted, does it play ok with other software?
[16:35] <elkng> durandal_1707: no, seems like I need to redownload video, because mplayer crashes playing original mp4 file after 4 minutes
[16:36] <durandal_1707> mplayer uses old lavf/lavc too
[16:37] <durandal_1707> you need qt time or wmp that does not use lavf/lavc
[21:12] <llogan> DeLoreann: did you figure it out. there is at least one other *BSD user here that may be able to provide assistance.
[21:14] <DeLoreann> i managed to solve the problem by "-acodec copy -vcodec copy", here is sample: http://bocik.pl/ss/player.php?v=AiBeEC.flv
[00:00] --- Sun Feb 10 2013