[Ffmpeg-devel-irc] ffmpeg.log.20130228
burek
burek021 at gmail.com
Fri Mar 1 02:05:01 CET 2013
[01:07] <Prophet___> Hello i am looking for help streaming utilizing FFmpeg, and i have also never used this software before so i am very new to this, i am also a ubuntu newbie and am still learning.
[01:08] <Prophet___> oops
[01:08] <Prophet___> is there anyone who is willing to help me use ffmpeg?
[01:13] <gmaxwell> I assume everyone has seen the new http://xiph.org/video/vid2.shtml now?
[02:26] <pr0ph3t> i get audio from my stream but not video what do i need to do
[02:26] <pr0ph3t> i am very new to this and need to be walked through
[02:26] <pr0ph3t> sry
[02:30] <defaultro> when encoding a video, should I always set it to yuv420?
[02:31] <sacarasc> pr0ph3t: When you do your ffmpeg line, paste bin that all the way down to where the command output finishes.
[02:32] <pr0ph3t> when you say 'ffmpeg line' do you mean what i type in terminal to start stream
[02:33] <sacarasc> pr0ph3t: Yes. Unless you're using ffserver, in which case get the encoding info and paste that.
[02:36] <pr0ph3t> i used that site to make a paste how do i post it
[02:36] <pr0ph3t> im sorry im really new to all of this
[02:38] <sacarasc> Get the URL that it gives you (may be in your address bar) and paste that here.
[02:38] <pr0ph3t> http://pastie.org/6351775
[02:47] <pr0ph3t> did i do it right?
[02:51] <PaulWay> gmaxwell: great video - I now need to find the original :-)
[02:51] <PaulWay> found...
[02:52] <gmaxwell> PaulWay: you mean lossless copies or the first episode?
[02:52] <PaulWay> The first video - and I've got it. And I see the download links too - nice work, xiph.org :-)
[02:58] <prophet__> sacarasc: anything from the pastie
[03:12] <relaxed> pr0ph3t: I doubt justin.tv supports yuv444p. Add "-filter:v format=yuv420p" to your command.
[03:14] <relaxed> llogan: To be honest it's been a while since I've updated it.
[05:42] <Mista_D> Need advice on padding please: "-i 1080p.ts -s 320x240 -vf pad=320:176" get error "Input area 0:0:1920:1080 not within the padded area "
[05:42] <Mista_D> http://pastebin.ca/2317723
[05:48] <Mista_D> I need the output to be the 320x240 but actual video encoded to 320x176, and padded to x240.
[05:53] <PaulWay> Is the pad argument pad=320:176 or pad=320x176?
[05:55] <Mista_D> PaulWay: shoudl be 320:176
[05:56] <PaulWay> I'd guess from the error that your input isn't being scaled before the vf is applied.
[05:56] <PaulWay> I'm happy to defer to someone with actual ffmpeg experience
[05:58] <relaxed> Mista_D: what exactly are you trying to accomplish?
[06:00] <relaxed> -filter:v scale=320:-1,pad=320:240 ?
[06:02] <Mista_D> relaxed: i want to scale 1080p to 320x176, then pad it to 320x240 res.
[06:03] <relaxed> that won't maintain the correct aspect ratio.
[06:04] <relaxed> my command will
[06:05] <Mista_D> relaxed: your command off center it did (:
[06:05] <relaxed> hmm, is your ffmpeg old?
[06:06] <Mista_D> relaxed: the video is moved to the top and bottom is havity padded.
[06:06] <Mista_D> *heavily
[06:07] <relaxed> -filter:v scale=320:-1,pad=320:240::(ow-iw)/2:(oh-ih)/2
[06:11] <Mista_D> relaxed: 113 Fire Flower. Compiled with yesterday's version of x264.
[06:12] <Mista_D> relaxed: new command's padding is still all on the bottom.
[06:17] <relaxed> I stuck an extra colon in there, it should work though
[06:18] <relaxed> -filter:v 'scale=320:-1,pad=320:240:(ow-iw)/2:(oh-ih)/2'
[06:25] <Mista_D> relaxed: Thanks!! it works.
[06:28] <relaxed> you're welcome
[06:29] <relaxed> add -sws_flags lanczos for better scaling
[10:23] <kms_> can i share my webcam live stream in webm to network via http? with ffserver. it is stable?
[11:19] <kms_> or can i share my stream from ffmpeg via another http server
[11:19] <kms_> ?
[11:43] <jeje> Hi to all...
[11:46] <jeje> Just a question. If I want to decode H264 video stream with ffmpeg, do I need to set the pts and dts of the AVPAcket struture before decoding with avcodec_decode_video2. Because I have my own function to get RTP packets from the IP camera and and recontruct them to have a whole frame before sending to FFMPEG.
[11:55] <jeje> for the moment, I just set the AVPacket with av_init_packet. My decoding parts works but it's to know if it is necessary.
[11:57] <jeje> Because when I make the capture of my RTP packet from the IP camera, the fps I have by the camera isn't a constant (like always 25 fps). It can varying from 24fps to 26 fps. I want to know if it can have a consequence on the h264 decoding part
[12:32] <wakou2> Hi folks, anyone here know about get_iplayer? I am trying to record video and then transcode so I can save to a thumbdrive and then play on my TV....
[12:33] <wakou2> My TV is not very High-tech... http://wstaw.org/m/2013/02/28/plasma-desktopfk3286.png
[12:34] <wakou2> If I use winFF to transcode from the mpeg4 from get_iplayer, it takes forever... hours, more than the actual show time..
[12:36] <wakou2> I have tried ffmpg if > of .avi and .mpg but the output is pretty horrible..
[12:37] <wakou2> I am wondering ifI can download directly to a format that will play on my TV without re encoding..
[12:37] <wakou2> My TV can recognise the file from get_iplayer, but can't play it...
[12:37] <relaxed> wakou2: what format does your tv play?
[12:38] <relaxed> mpeg4/mp3 in avi?
[12:38] <wakou2> In the paste link above is a png of the instructions from the TV..
[12:38] <wakou2> I think only up to mpeg2
[12:39] <relaxed> no xvid is mpeg4
[12:39] <relaxed> no, xvid is mpeg4
[12:39] <wakou2> Oh ty I did not know that..
[12:41] <relaxed> try, ffmpeg -i input.mkv -c:v mpeg4 -vtag xvid -q:v 3 -c:a libmp3lame -ac 2 -b:a 128k output.avi
[12:42] <relaxed> add -t 20 for a twenty second sample
[12:42] <wakou2> Ty TY ...
[12:42] <relaxed> (after the input)
[12:43] <wakou2> OK I will try that .... :)
[12:59] <relaxed> wakou2: did that work?
[13:37] <wakou2> relaxed: Sorry, I was called away, Trying it now... Is there a flag or option that will record how long ffmpeg takes to complete a certain operation?
[13:50] <wakou2> relaxed: Oh that seems much better, certainly much faster than using winFF with the settings I had guessed at before, and the picture quality is plenty good enough... Thank you..
[14:41] <Aaronds> Hi, I'm having trouble converting .wmv files to .mp4 for use in html5 video... I've converted one .wmv which works fine, however the other cant seem to be decoded when used on the web. I'm using the same ffmpeg commands for both. Commands & output for both are here: http://pastebin.com/KBdVJdPb. Can anyone help?
[14:45] <wakou2> relaxed: I put -benchmark, to give me the info..
[14:46] <JEEB> Aaronds, first of all, when specifying a bit rate, do specify :v or :a to define the audio or video bit rate. just -b is ambiguous in most cases, second of all I see some timestamp funny-bunny stuff in the second one
[14:46] <JEEB> might be that
[14:46] <Aaronds> Are you saying the -b could be the cause of the timestamp could be something to do with it or both?
[14:47] <JEEB> no, the -b is a separate thing, but you should still learn to switch to it
[14:47] <JEEB> switch to -b:a and -b:v when specifying bit rates
[14:47] <JEEB> anyways, the timestamp stuff sounds funky
[14:48] <JEEB> and problems related to that most probably are the cause of the non-workingness
[14:49] <Aaronds> ok I will change the bitrate stuff, any idea what could be causing the actual problem then?
[14:49] <JEEB> input file is wonky, or ffmpeg is doing something wrong
[14:50] <JEEB> I see you use a build from january
[14:50] <JEEB> tried current git HEAD?
[14:50] <Aaronds> no, I'm currently just using ffmpeg from the ubuntu repo
[14:51] <JEEB> not the standard ubuntu repo then
[14:51] <JEEB> anyways, try current ffmpeg HEAD is all I can say
[14:51] <Aaronds> will do
[16:00] <alexavenger> I'm looking fore someone that can help my with, dubbing cards. I have to add a dubbing card at the end of a film. Dubing card is jpeg. How can I do it?
[16:02] <alexavenger> At the end.. I need to add a photo at the end of a film.
[16:05] <durandal_1707> can you explain what final output should look like?
[16:06] <izemize> hy all
[16:06] <izemize> ffserver is accept multiple input?
[16:07] <izemize> ffmpeg -i rtmp://xxx/xx/1 http://ffserver:xxxx/1.mp4
[16:07] <izemize> ffmpeg -i rtmp://xxx/xx/2 http://ffserver:xxxx/2.mp4
[16:08] <izemize> i need convert mp4, ogg and webm type from many many rtmp stream
[16:08] <izemize> oh shit, no.
[16:09] <izemize> so... i need many many rtmp stream convert to live mp4, ogg, webm :)
[16:09] <izemize> but i think ffserver support stream only one rtmp input to multiple format output
[16:21] <diroots> I am converting a DVCPRO HD to a H264 video. on the audio part, with the libfaac codec, even if i set -ab 320k, I get a variable audio bitrate. do one of you know why ?
[16:24] <diroots> http://pastebin.com/E6EbkqrM durandal_1707
[16:25] <diroots> is it because libfaac does not support such a high bitrate?
[16:26] <diroots> (I have the same problem if I use -ab 128k ... )
[16:30] <durandal_1707> it may be mediainfo nonsense
[16:30] <durandal_1707> you could try vbr with -qscale 0 and see what it reports than :)
[16:44] <kms_> when i run ffserver i recive error
[16:44] <kms_> ffserver: relocation error: ffserver: symbol ffm_read_write_index, version LIBAVFORMAT_53 not defined in file libavformat.so.53 with link time reference
[16:44] <kms_> why?
[16:48] <kms_> hmm sudo mv /usr/lib/i386-linux-gnu/i686/cmov/libavformat.so.53 /usr/lib/i386-linux-gnu/i686/cmov/libavformat.so.53.disabled fix problem
[16:50] <diroots> durandal_1707: but I want CBR, not VBR
[16:51] <diroots> on the video stream, it is constant, but not on the audio stream
[16:51] <durandal_1707> diroots: are you sure mediainfo is right and not buggy/broken/useless?
[16:51] <durandal_1707> kms_: that is also old
[16:52] <diroots> durandal_1707: how could I ensure it is right?
[16:54] <diroots> durandal_1707: mediainfo is the latest version
[16:58] <JEEB> I'd rather trust something else than mediainfo to be honest
[16:58] <JEEB> if you set a certain audio bit rate, that is set
[17:17] <Aaronds> I've just compiled ffmpeg on ubuntu as per the guide on the ffmpeg site - it works fine except my cron script cant seem to find the ffmpeg command, yet it works fine for me. Any idea what the issue is?
[17:17] <diroots> Aaronds: use full path in your cron script
[17:18] <Aaronds> thanks diroots
[17:18] <diroots> and find full path with 'whereis ffmpeg'
[17:28] <izemize> re
[17:28] <izemize> can anyone help me (ffserver)?
[17:29] <izemize> my problem: i have many rtmp live stream. i need convert live rtmp stream to http streaming, ogg, and webm
[17:29] <izemize> it possible with ffserver?
[17:29] <izemize> multiple feed is allowed?
[17:29] <durandal_1707> should be
[17:30] <izemize> ?
[17:30] <izemize> should be convert live rtmp stream to http streaming, ogg, and webm :)
[17:32] <taqattack> anyone here?
[17:32] <kms_> <durandal_1707>
[17:33] <taqattack> is there anyway to get real-time output from ffmpeg
[17:33] <taqattack> to rtmp url
[17:33] <durandal_1707> yes
[17:33] <kms_> what format specified in the <Stream for broadcast in webm?
[17:34] <taqattack> im using "ffmpeg -re -i file.flv -vcodec copy -acodec copy -f flv rtmp://url"
[17:34] <taqattack> its all fine until theres network congestion
[17:34] <taqattack> after that the stream gets more delayed than the video
[17:35] <taqattack> is there anyway to keep it real-time by skipping/dropping frames
[17:42] <kms_> i try webm streaming and recive error Codec for stream 0 does not use global headers but container format
[17:43] <kms_> VideoCodec libvpx Format webm
[17:49] <durandal_1707> kms_: ffserver needs global header?
[17:51] <durandal_1707> kms_: what version is this?
[17:52] <kms_> ffserver version 0.10.6-6:0.10.6-0ubuntu0jon1~precise1 Copyright (c) 2000-2012 the FFmpeg developers
[17:53] <kms_> from ppa
[17:54] <durandal_1707> that is very old
[17:54] <kms_> where i can find last ppa for ubuntu?
[17:55] <jeje> hi to all
[17:55] <kms_> i install from this https://launchpad.net/~jon-severinsson/+archive/ffmpeg
[17:57] <kms_> may be better to use another http server can be gstreamer? I don't understand
[17:58] <taqattack> so any one have any idea?
[17:59] <jeje> I use FFMPEG to decode H264 video Streaming from an IP camera. I always set SPS and PPS on each keyframe when I capture RTP packet. Does I need to set something else to AVCodecContext and or AVCodec (like time_scale, sample_fmt, ticks_per_frame). Because, in documentation, I see for ticks_per_frame: Most notably, H.264 and MPEG-2 specify time_base as half of frame duration if no telecine is used ...
[17:59] <jeje> Set to time_base ticks per frame. Default 1, e.g., H.264/MPEG-2 set it to 2. So Do, I really need to set it to 2? And are there other parameters to set ?
[18:00] <jeje> for information, I use the FFMPEG libraries 1.1.2 that I compile, and use avcodec_decode_video2 to decode
[18:01] <jeje> I just use avcodec_alloc_context3 to initialize my context and avcodec_open2 to initialize my Codec
[18:01] <jeje> thnaks for your answers
[18:07] <izemize> it possible stream live ogg or webm?
[18:10] <gmaxwell> izemize: sure. Icecast supports replication for both, you need a shoutcast source I don't think ffmpeg has one integrated(?) though. Gstreamer does. you can do something like ffmpeg <opts> | oggfwd e.g. for ogg.
[18:10] <izemize> i have rtmp streams.. but it's only work with flash
[18:11] <izemize> i don't know good solution for stream to non flash browsers
[18:11] <izemize> ffserver is support webm stream but only one stream / feed
[18:12] <izemize> i have 1000+ rtmp live stream
[18:12] <izemize> :)
[18:13] <gmaxwell> As I said, icecast supports it.
[18:15] <izemize> ok. thanks
[18:18] <kms_> i think icecast only audio support
[18:22] <izemize> webm and ogg and mp4 is simple http pseudo stream right?
[18:25] <izemize> so.. if i convert rtmp live stream with ffmpeg -re -i rtmp://localhost/app/name -f flv some.mp4
[18:25] <izemize> it work?
[18:25] <Fjorgynn> or rtmpdump?
[18:25] <izemize> i think apache mod_h264_stream
[18:26] <kms_> <izemize> from ffmpeg stream webm to network with gstreamer is possable?
[18:26] <izemize> source is rtmp server. not ffmpeg.
[18:27] <izemize> rtmpdump.. i check out now
[18:27] <izemize> thx for tipp
[18:29] <kms_> anybody now, can i stream to network with gstreamer using ffmpeg as source?
[18:34] <izemize> i don't know gstreamer
[18:34] <izemize> :(
[18:36] <kms_> i don't know both
[18:37] <kms_> but i wery need live stream my webcam with webm
[18:40] <kms_> now i compile last ffmpeg, may be i install it without troubles
[18:40] <kms_> libav dont support avserver
[18:42] <kms_> hmm, i install it and i see "ffserver version N-50376-gc4735ee"
[18:42] <kms_> what is it?
[18:43] <izemize> you are install from git + checkinstall?
[18:45] <kms_> without checkinstall
[18:50] <izemize> rtmpdump same as ffmpeg -re -i rtmp://..... -f flv tofile.mp4 yes?
[18:50] <izemize> who is different?
[19:17] <brontosaurusrex> is there a magical command line to extract all audio to stereo or mono individual tracks? (from a multitrack input)
[19:17] <brontosaurusrex> i have 4 stereo streams for example
[19:19] <Fjorgynn> -ac 4 ?
[19:19] <Fjorgynn> I think
[19:20] <brontosaurusrex> nope
[19:21] <brontosaurusrex> i have 8 discreet channels in 4 tracks
[19:21] <brontosaurusrex> in single avi or mxf file
[19:21] <brontosaurusrex> and i would like 4 stereo wav files generated (or 8 mono files)
[19:29] <brontosaurusrex> ok, this seems to work: ffmpeg -i file.avi -map 0:1 "out.1.wav" -map 0:2 "out.2.wav" .....
[19:30] <brontosaurusrex> but lacks any abstraction of course
[19:30] <brontosaurusrex> anything smarter?
[19:37] <durandal_1707> channelsplit filter
[19:38] <durandal_1707> and join filter to join multiple streams into one multichannel stream
[19:38] <brontosaurusrex> <brontosaurusrex> and i would like 4 stereo wav files generated (or 8 mono files) <<<<
[19:39] <durandal_1707> soo?
[19:39] <durandal_1707> what you do not understand?
[19:40] <brontosaurusrex> not sure how to refraze
[19:53] <taqattack> can someone please help
[20:20] <Sashmo_> can anyone tell me if its possible to send to multiple detinations when using -y rtmp://.......
[20:32] <JodaZ> Sashmo_, did you try, documentation claims you just need to repeat the output parameters and specify another output
[20:32] <Sashmo_> JodaZ: trying that now, I just dont like the example they gave
[21:34] <hughmanwho> Anyone ever encounter a 8kb memory leak when repeatedly calling 'av_read_frame'
[21:59] <hughmanwho> Anyone have any memory leak problems with 'av_read_frame'?
[22:16] <teratorn> hughmanwho: I can assume you're using valgrind?
[22:32] <letoram> Trying to troubleshoot a problem with sound encoding going silent on me (encoding / muxing playback works when input / output sample formats match so it boils down to swr_convert use) http://pastebin.com/US96XwGj -- (inbuf is properly aligned, returned buffer will be fed into avcodec_fill_audio_frame -> avcodec_encode_audio2, testcase is going from AV_SAMPLE_FMT_S16 -> AV_SAMPLE_FMT_FLTP. returned #of frames match but contents is silent. Any ideas?
[22:39] <mpfundstein_> i heard **RUMORS** that Frauenhofer is trying to sue companies that use ffmpeg..
[22:39] <mpfundstein_> can someone pls confirm this is big bs?
[22:41] <gmaxwell> mpfundstein_: I haven't heard that lately, but I doubt ffmpeg has anything to do with it.
[22:41] <JEEB> if you don't use formats that fraunhofer controls, or software under their license while breaking the license, I wouldn't be surprised if they'd go against you if you get big enough. That said, it'd be completely unspecific to whether it's ffmpeg or not
[22:42] <gmaxwell> exactly what jeeb said.
[22:42] <gmaxwell> If you use formats that fraunhofer believes they hold patents on, and you're not paying to license those patents& they may ask you to pay up. But that you're using ffmpeg would be irrelevant.
[22:45] <JEEB> yup
[22:45] <sybariten> hi
[22:45] <sybariten> what does "Unable to find a suitable output format for 'c:a'" mean?
[22:46] <JEEB> put both your command line and full terminal output onto a pastebin site of your choice
[22:46] <sybariten> i'm using this: ffmpeg -ss 00:43 -t 30 -i $videofile -vn -sn c:a copy ./i_blend.mp4
[22:46] <JEEB> and then link it here
[22:47] <taqattack> JEEB, can u help me please
[22:47] <JEEB> I said exactly what is needed
[22:47] <mpfundstein_> ok guys, thx that was what i was thinking, there is just this manager guy and he spreads this rumors :D
[22:47] <JEEB> mpfundstein_, the only people who are bullies are basically companies that wouldn't let you get a license to begin with, like Dolby or DTS
[22:48] <JEEB> those specifically hate open source
[22:48] <mpfundstein_> JEEB: can it be because of libfdk_aac ?
[22:49] <JEEB> yes, it's theirs and has a license of theirs. Read it.
[22:50] <sybariten> JEEB: http://pastebin.ca/2326815
[22:50] <mpfundstein_> JEEB: already on it ..
[22:51] <JEEB> sybariten, you're using libav, so use the 'avconv' binary instead
[22:51] <JEEB> the ffmpeg (app) in libav is out-of-date
[22:51] <sybariten> hmmm ok
[22:52] <sybariten> does this mean i should get a newer/different ffmpeg install, or that libav has a separate program called avconv that i probably have?
[22:52] <JEEB> libav is a fork of ffmpeg, and the up-to-date binary there is 'avconv'
[22:52] <JEEB> ffmpeg (project) has its own up-to-date binary called 'ffmpeg'
[22:53] <sybariten> oh... so, in other words, avconv is quite similar to ffmpeg in useage etc ?
[22:53] <JEEB> yes, since elenril rewrote parts of ffmpeg when the name change was done
[22:53] <JEEB> and then those changes were mostly if not fully merged to ffmpeg's ffmpeg
[22:54] <sybariten> interesting, didnt know about this
[22:54] <sybariten> ah damn, all that info was in bright yellow text in the standard text being printed when ffmpeg is run
[22:55] <sybariten> and... the only difference i can see, when using avconv, is that i get the same error message but now its in red text :)
[22:55] <JEEB> try putting -t after -i
[22:55] <sybariten> aiight
[22:56] <sybariten> hmmm... nope... same error message...
[22:57] <sybariten> $ avconv -ss 00:43 -i $videofile -t 30 -vn -sn c:a copy ./i_blend.mp4
[22:57] <JEEB> pastebin it all again :P
[22:57] <sybariten> ok...
[22:57] <JEEB> also you don't need to add the ./
[22:57] <sybariten> JEEB: its in my instinctive behaviour....
[22:57] <sybariten> makes me certain i'm putting or using stuff where i am
[22:59] <sybariten> http://pastebin.ca/2326818
[22:59] <JEEB> sybariten, tried removing the ./ ?
[23:00] <sybariten> hehe, damn, if thats the problem than i'm gonna have to "eat" what i just said, as we say in swedish
[23:01] <sybariten> nope, exactly the same message...
[23:01] <JEEB> show it to elenril then, you know where :)
[23:02] <sybariten> aiight
[00:00] --- Fri Mar 1 2013
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