[Ffmpeg-devel-irc] ffmpeg.log.20130624

Tue Jun 25 02:05:01 CEST 2013

[00:00] <bencc> I don't need to record audio with html5
[00:00] <bencc> all I need is to transcode a live rtmp stream to mp3 and broadcast it
[00:00] <klaxa> it might make things easier though
[00:00] <bencc> how come?
[00:00] <klaxa> no flash, no rtmp
[00:01] <klaxa> hmm... although, it might not be as convenient, since i'm not sure how to move the audio from the recording browser to a server
[00:01] <klaxa> send it over a socket i guess
[00:01] <bencc> there is no echo cancelation
[00:02] <klaxa> mmh
[00:02] <bencc> webrtc might work but it's not widly supported yet
[00:02] <bencc> and it's hard to work with some clients using webrtc and other flash
[00:02] <bencc> transcoding to mp3 shouldn't be that hard. that's what ffmpeg is for...
[00:05] <klaxa> well right now the only hard problem seems to be to get the audio from the client to ffmpeg
[00:06] <bencc> maybe
[00:06] <bencc> but on my server it seems to work
[00:07] <bencc> I just can't get it to stdout
[00:09] <klaxa> you can't even record it!
[00:09] <klaxa> ffmpeg obviously does not start encoding, which in turn means it's not even receiving anything properly
[00:11] <bencc> for you
[00:11] <bencc> I managed to get an mp3 file but not to stream it
[00:13] <klaxa> so you got an mp3 and you could listen to it?
[00:13] <klaxa> try to reproduce
[00:14] <bencc> yes
[00:14] <klaxa> can you reproduce and pastebin all the logs
[00:14] <klaxa> ?
[00:15] <bencc> I'll try now
[00:20] <bencc> klaxa: http://dpaste.com/1265946/
[00:20] <bencc> there is test.mp3 file but I'm not able to play it
[00:20] <klaxa> and you ca--
[00:20] <klaxa> yes just what i though
[00:20] <klaxa> it's not actually recording anything
[00:21] <klaxa> *thought
[00:25] <bencc> klaxa: I used a wrong url
[00:26] <bencc> now I can save test.mp3 and I see the log of ffmpeg
[00:26] <bencc> but I can't play the mp3 file
[00:26] <bencc> klaxa: http://dpaste.com/1265950/
[00:28] <bencc> klaxa: now I'm getting packets in the shell
[00:28] <bencc> weird
[00:32] <bencc> klaxa: something is weird with the pipes
[00:33] <bencc> it buffers the output and when I'm closing the media server suddenlty I see many asci messages?
[00:33] <bencc> without the ?
[00:33] <klaxa> huh? that's weird indeed
[00:35] <klaxa> try running it with "stdbuf -o0 -e0 -i0" i.e. stdbuf -o0 -e0 -i0 ffmpeg -i <all the other stuff>
[00:35] <bencc> ?
[00:35] <bencc> what is it?
[00:35] <klaxa> sets buffer values for pipes to 0
[00:36] <bencc> the same
[00:37] <klaxa> ah hmm... i just saw, you might try your luck with increasing analyze duration and stuff
[00:38] <bencc> ?
[00:38] <bencc> how?
[00:38] <klaxa> add "-analyzeduration 10000000" or something
[00:40] <bencc> same
[00:41] <bencc> something is wrong with ffmpeg
[00:42] <bencc> I'm sure
[00:42] <klaxa> file a bugreport then
[00:43] <bencc> ok
[01:10] <bencc> klaxa: the same happens with avconv. when I'm stopping the media server I see output in the terminal
[03:25] <lonestarr> A question regarding RTSP client (connecting to an IP camera reading H.264 directly via RTSP) ... is there a way to limit the bandwidth to say 64 or 128 kbps in the client? Spent quite some time googling but did not find any information on this.
[03:28] <lonestarr> ffmpeg -maxrate 64k -b:v 64k -sn -an -i "rtsp://admin:admin@x.x.x.x:80/videoSub"  does not work unfortunately - looking for limiting network bandwith somehow, any ideas?
[03:35] <sacarasc> lonestarr: Only if the server supports that.
[04:06] <ten1> I have a quick and perhaps stupid question: what happens if you try to convert a video to the same format it is already in?
[04:07] <llogan> ten1: depends on the format.
[04:07] <sacarasc> ten1: You'll lose quality, in most cases.
[04:07] <sacarasc> Unless you're going from lossless to lossless.
[04:07] <ten1> sacarasc: kay. I'll try and avoid that then
[04:08] <ten1> llogan, sacarasc, thanks
[04:09] <llogan> what are you trying to do?
[04:10] <ten1> llogan: video upload/transcode for web players
[04:10] <sacarasc> YouTube transcodes all videos uploaded to it, and people seem to not care about that.
[04:11] <ten1> using gearman, gearman-php, ffmpeg, ffmpeg-php
[04:11] <sacarasc> ffmpeg-php is years out of date and doesn't work with any recent ffmpeg.
[04:11] <ten1> then it should work great with what's in the debian repos
[04:12] <ten1> but I suppose I can just use shell commands instead
[04:13] <llogan> debian doesn't use ffmpeg
[04:14] <ten1> llogan: come again?
[04:18] <ten1> llogan: the package aptitude lists has Maintainer: Debian Multimedia Maintainers <pkg-multimedia-maintainers at lists.alioth.debian.org>
[04:20] <ten1> but I'll have to look into this more carefully
[04:21] <llogan> i guess ffmpeg package in squeeze may still be FFmpeg upstream
[04:21] <llogan> not squeeze backports though
[04:24] <ten1> llogan: this sounds like a minefield. But, so the php wrapper is discouraged? I should be invoking it directly?
[04:25] <sacarasc> Yes.
[04:26] <ten1> all right. I very much appreciate the information and advice, and I am very pleased that I stopped by here
[04:27] <llogan> good luck with the project
[04:27] <ten1> thanks :)
[04:38] <knIOO> I'm getting [Parsed_concat_0 @ 0245e740] Failed to configure output pad on Parsed_concat_0 - full output is here http://pastebin.com/Tzqpinjf
[04:39] <knIOO> I'm trying to join two videos of the same framerate and the same resolution
[04:39] <knIOO> Can someone let me know what's going on?
[04:39] <knIOO> I'm getting [Parsed_concat_0 @ 0245e740] Failed to configure output pad on Parsed_concat_0 - full output is here http://pastebin.com/Tzqpinjf
[04:39] <knIOO> apologies, still getting used to this new irc client heh
[04:39] <klaxa> unknown paste id
[04:39] <knIOO> argh, apologies, one moment
[04:41] <knIOO> klaxa: sorry about that, here is my current output http://pastebin.com/M1B7YSix
[04:43] <klaxa> hmm you're using quite some parameters i'm not too familiar with :X
[04:44] <knIOO> no problem heh - my goal is to join two videos of unknown codec together basically
[04:44] <knIOO> but I thiiink there's a mismatched aspect ratio somehow even though they're both 1280x720
[04:44] <knIOO> I just don't know how to set an aspect ratio for the streams, something to use with setsar/setdar but I have no idea how to use it
[04:45] <knIOO> can't find much documentation on the net about it either
[04:45] <klaxa> well i guess you've already come across this: http://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide but this might be the best guide you'll find
[04:46] <llogan> knIOO: you have a filter in your input link label: [0:1 -setsar=1:1]
[04:46] <llogan> change it to [0:1]
[04:46] <knIOO> llogan: same error - just noticed that is an old version of the output, get the exact same error without it tho
[04:47] <knIOO> that was just me experimenting on my own
[04:47] <llogan> ah, i see. sar 0:1. i've seen another user who experienced this but i forgot how it was resolved.
[04:48] <knIOO> llogan: it might have been me heh, I was here asking the same question a few weeks ago
[04:49] <knIOO> I just have no idea how to set the sar value for either stream
[04:49] <llogan> https://ffmpeg.org/trac/ffmpeg/ticket/2456
[04:49] <llogan> youre on the right track, but you're implementing incorrectly
[04:50] <knIOO> yep - also heh I actually added some info to that bug back in may
[04:51] <knIOO> I just can't find anything on how to set filters to the individual streams (if I'm understanding the problem correctly)
[04:51] <llogan> ffmpeg -i tojoin.mkv -i perftest1.avi -filter_complex "[1:0]setsar=1/1[sarfix];[0:0] [0:1] [sarfix] [1:1] concat=n=2:v=1:a=1 [v] [a]" -map "[v] -" -map "[a]" -vcodec libx264 output.mkv
[04:52] <llogan> theres a typo in that
[04:52] <llogan>  -map "[v]" -map "[a]"
[04:53] <knIOO> that..seems to have worked!
[04:54] <lonestarr> very sorry everyone, asked the question regarding limiting network bandwidth for ffmpeg as RTSP client about 90 minutes ago but had a power failure and can't see any history. Was there a solution for this? Is there a way to throttle the traffic somehow?
[04:55] <knIOO> llogan: you are a hero - that's fixed the problem completely.
[04:55] <knIOO> thank you so much
[04:56] <llogan> so the filterchain translates as "take the first stream from the second input, and apply setsar and name output link label [sarfix]"
[04:57] <llogan> read the following 5 times and then you'll get the hang of it: http://ffmpeg.org/ffmpeg-filters.html#Filtergraph-syntax-1
[04:57] <knIOO> this looks interesting
[04:58] <llogan> also http://ffmpeg.org/ffmpeg-filters.html#concat
[04:59] <llogan> lonestarr: "<sacarasc> lonestarr: Only if the server supports that"
[05:00] <lonestarr> okay, thanks for letting me know. So if the server does support it, how do I request a stream at say 64 kbps?
[05:02] <lonestarr> -maxrate 64k or -b:v 64k or something else? I tried this, but it was still downloading the stream at 500 kbps regardless.
[05:05] <sacarasc> It's usually a slightly different URL, like rtmp://blah/feed/64 or something.
[05:08] <knIOO> llogan: so looking at -filter_complex, am I correct if I am understanding that [0:0] for example represents 1st input, video stream, and [1:1] represents second input, audio stream - if the containers just contain one video and one audio stream
[05:09] <llogan> yes. usually. unless the first stream is audio. [0:0] just represents first input, first stream of that input.
[05:10] <knIOO> that makes sense, for most cases is it video first stream, audio second or is it just completely random?
[05:12] <llogan> most cases, but i've seen some inputs were audio is listed first. or you could probably just use [0:v:0] to be sure: first input:video:first stream that is video
[05:12] <llogan> http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1
[05:13] <llogan> http://ffmpeg.org/ffmpeg.html#Advanced-options (see -map for examples)
[05:13] <knIOO> you've taught me more in 10 minutes than I've learnt in the few months I've been playing with ffmpeg
[05:14] <llogan> it's a good distraction from a boring task that i am postposing.
[05:16] <knIOO> Hahah dare I ask
[05:17] <llogan> fixing someone elses ancient PHP code.
[05:19] <lonestarr> okay, thanks for your help / time guys
[05:19] <llogan> knIOO: apparently that user that experienced sar 0:1 was you, and the sarfix example worked for you last time.
[05:21] <knIOO> llogan: I worry about my short term memory sometimes
[05:21] <knIOO> looking at my own bug report i even listed the fix..facepalm
[05:27] <knIOO> and oh God I feel your pain llogan, I find php a language that is particularly irritating to edit/add to after the fact
[05:27] <knIOO> curly braces can go to hell
[08:14] <ilove11ven> lonestarr: I think you have  throttled the traffic on the wrong side.
[08:41] <enelar> guys, i am trying to emerge ffmpeg in ARM gentoo. but i get compilation error
[08:41] <enelar> i am already tried to install older version, the same
[09:47] <ink3> Hello
[09:48] <ink3> Can you help with 'burn' subtitles? http://pastebin.com/U7JxeJDC
[11:22] <LordDoskias> when i'm muxing stuff with libav but i have already the encoded data (audio/video) do i still need to set up codecs in the output context etc?
[11:24] <JEEB> libavformat would be the one you'd be needing in that case
[11:24] <JEEB> and I think that nowadays takes in AVPackets or so? not sure
[11:32] <LordDoskias> i've been reading the muxing example
[11:32] <LordDoskias> but there they setup codecs and whatnot
[12:03] <LordDoskias> i have 2 questions: 1. how exactly does av_rescale_q rescale numbers e.g. what is the end result? 2. where can i set the framerate of the resulting container after muxing?
[12:43] <LordDoskias> is avformat_alloc_output_context2 a new functions, since in my distribution of avlib i do not have it ?
[12:50] <LordDoskias> which is the current stable version of libav 0.9 or 0.8 branch?
[12:54] <sacarasc> #libav
[12:54] <sacarasc> #ffmpeg deals with ffmpeg, not libav.
[12:54] <LordDoskias> okay, thanks
[12:55] <LordDoskias> i thought they were practically identical since ffmpeg uses avlib underneath, no?
[13:00] <sacarasc> Read that.
[13:12] <LordDoskias> but why is that that when i read: https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/muxing.c the includes point to libavformat/libavcodec etC?
[13:19] <LordDoskias> on the other hand, how can i use ffmpeg's distribution of libav where for example the function avformat_alloc_output_context2 is implemented?
[13:33] <saste> LordDoskias, you're confusing libav* libraries and Libav project
[13:33] <saste> libavcodec, libavformat, etc. are included in FFmpeg (the project) as well
[13:33] <LordDoskias> and you apparently do some other stuff with it
[13:33] <LordDoskias> e.g. add functinos
[13:34] <saste> and the API is mostly equivalent, since FFmpeg merges all changes from Libav
[13:34] <LordDoskias> but you also add your own funcs
[13:34] <saste> on the other hand Libav doesn't do the same with FFmpeg, so in general it has less features, including less functions
[13:35] <LordDoskias> so what do i have to do if i want to use ffmpeg in my project programatically ?
[13:35] <LordDoskias> e.g. use its apis not the command line tool
[13:36] <bencoh> use the api described in ffmpeg doxygen
[13:39] <LordDoskias> i do not have a function described there
[15:29] <Onkeltem> Hi. I forgot the name of GUI utility for editing contents of MP4 containers. Any thoughts?
[15:33] <Mavrik> mp4box?
[16:00] <setkeh> hey guys im starting ffmpeg with "ffmpeg -f pulse -i default -f x11grab -acodec flac -r 60 -s 1920x1080 -i :0.0+1360x768 -vcodec libx264 -crf 0 -preset ultrafast out.mkv" but i cannot figure out why the sound is not in sync with the video any idea's ??
[16:27] <smoke_> hi im running ubuntu 13.04 and im having trouble with dvdstyler and ac3, i read on the ubuntu forums ot get the newer version of ffmpeg is there a specific ppa i should grab or will the jons one do?
[16:27] <Mavrik> get a static build
[16:27] <Mavrik> ppas are also mostly obsolete
[16:28] <smoke_> Mavrik, sorry whats a static build?
[16:29] <smoke_> Mavrik, in the official repo for ubuntu?
[16:29] <Mavrik> no.
[16:29] <Mavrik> forget about official repos.
[16:29] <Mavrik> http://dl.dropboxusercontent.com/u/24633983/ffmpeg/index.html
[16:30] <smoke_> k thx
[16:31] <setkeh> hey guys im starting ffmpeg with "ffmpeg -f pulse -i default -f x11grab -acodec flac -r 60 -s 1920x1080 -i :0.0+1360x768 -vcodec libx264 -crf 0 -preset ultrafast out.mkv" but i cannot figure out why the sound is not in sync with the video any idea's ??
[16:39] <setkeh> any one ??
[16:39] <klaxa> it's a common issue, blame pulse maybe?
[16:39] <klaxa> just shift the audio after recording
[16:39] <klaxa> shouldn't be that hard
[16:39] <setkeh> its worked for me before :/
[16:40] <setkeh> i would rather a fix for the problem then some cheap hack
[16:41] <klaxa> i'm amazed you can write 60 fps at fullhd even
[16:43] <setkeh> the fps isent causing the issue i have tried 10 fps 20 40 and 60 same problem
[16:43] <setkeh> and i get around 100-130MB/s iop out of my drives
[16:46] <klaxa> have you tried alsa instead of pulse?
[16:47] <setkeh> yeah same issue
[16:48] <klaxa> well just post-process the audio delay, it's not a big deal
[16:48] <klaxa> i doubt you will publish your recording right away anyways
[16:49] <setkeh> well thats the idea of fixing the issue and not some cheap hack
[17:20] <jure> hullo
[17:21] <jure> I've managed to transcode some MLP audio to FLAC, but it ended with a message "lossless check failed"
[17:21] <jure> is this significant?
[17:25] <jure> ok never mind
[17:25] <jure> probably because I didn't use concat
[18:10] <Mavrik> BigWhale, asm ja, bi alo :)
[18:30] <sacarasc> sdl240: You probably wanted:
[18:31] <sdl240> sacarasc, thats exactly what i want!
[18:31] <sdl240> thanks
[18:35] <bencc> klaxa: installing red5 to see if the problem is on ffmpeg side or the media server side
[18:35] <klaxa> alright
[18:35] <bencc> klaxa: if red5 won't work I'll try FMS free starter version
[18:35] <bencc> it must work with FMS :)
[19:13] <bencc> klaxa: now I'm having truble with red5. this is redicolus...
[19:13] <bencc> I mean I can't get red5 to work. not related to ffmpeg
[19:34] <bencc> klaxa: I have red running. trying to transcode to a file
[19:34] <klaxa> ok
[19:48] <bencc> klaxa: I'm getting Operation not permitted error when trying to connect to red5 rtmp stream
[19:49] <bencc> klaxa: others see this error too but I havn't found a solution yet
[19:50] <bencc> http://ffmpeg.org/trac/ffmpeg/ticket/138
[20:08] <sisco12> Hi all
[20:09] <sisco12> im facing a little problem here with ffmpeg any expert here can help so i can post my commands ?
[20:10] <sisco12> ....???
[20:10] <josemanuel_> just ask
[20:11] <sisco12> ok here we go
[20:11] <sisco12> when i use ffplay with this command :ffplay "rtmp://stream.marociptv.com/livepkgr/livestreamAlAoulaInter?9700d88cc84a6681ce26aaffa0482c0f_51c87e93 swfUrl=http://www.marocstream.com//e mbed/player.swf pageUrl=http://www.marociptv.com/tvaccess.cfm?tv=5E4B50B9959C61EA4AC18FD3_72a5d8e982cab2c57a0ff49bbb76673a live=1 swfVfy=1"
[20:11] <sisco12> it works
[20:12] <sisco12> but when i try to use ffmpeg to get it as input is not working
[20:13] <sisco12> ffmpeg -i "rtmp://stream.marociptv.com/livepkgr/livestreamAlAoulaInter?9700d88cc84a6681ce26aaffa0482c0f_51c87e93 swfUrl=http://www.marocstream.com //embed/player.swf pageUrl=http://www.marociptv.com/tvaccess.cfm?tv=5E4B50B9959C61EA4AC18FD3_72a5d8e982cab2c57a0ff49bbb76673a live=1 swfVfy=1"-vcodec libx264 -acodec mp3 -f  flv 22.mp4
[20:13] <sisco12> i got this error : HandleCtrl: Ignoring SWFVerification request, use --swfVfy! rtmp server sent error rtmp server requested close
[20:14] <sisco12> do u have any idea where it stuck?
[20:16] <sisco12> josemanuel_ have any idea?
[20:16] <sisco12> have u*
[20:17] <josemanuel_> have you used --swfVfy?
[20:17] <josemanuel_> as the error message suggests
[20:17] <sisco12> well the command is sfwVfy=0 or 1
[20:18] <josemanuel_> have you tried both to see if any works?
[20:18] <sisco12> yes i did
[20:18] <sisco12> ffplay works and ffmpeg no thats weird
[20:18] <sisco12> i mean the same link
[20:21] <sisco12> i got ty
[20:21] <sisco12> its ok :D
[20:37] <mykul> hi folks, i'm using amix to multiplex several audio files, and I've noticed that the final mixdown has significantly reduced volume
[20:37] <mykul> is the gain getting normalized somehow?
[20:37] <mykul> is there a way to prevent this or counteract it?
[21:47] <RSDRSDRSD> I am having some huge problems of making screenshots with ffmpeg from a wowza media server
[21:47] <RSDRSDRSD> problem is listed on http://pastebin.com/3KbGyurd
[21:47] <RSDRSDRSD> can someone please help me?
[22:35] <RSDRSDRSD> What is the best methog to create thumbnails from rtmp streams
[22:35] <RSDRSDRSD> if I want a thumbnail of the current live stream
[22:36] <bencc> anyone is able to get rtmp stream from red5
[22:41] <blink> Hi,  I'm running into an issue using 20130611-git-0f88a98-win64-static, output codec is prores quality 2.  When I create a file using an image sequence and define a framerate of 23.976, it not only rounds up to 28.98 (not critical) but drops frames in so doing.  I see the debug saying 25fps; is this a known issue or have I done something incorrectly?
[22:42] <blink> pardon, input is an image sequence.  output is mov container, prores 422 essence
[22:56] <JEEBsv> blink: image sequence might need -r before -i, and you should be able to set a proper X/Y frame rate, 24000/1001
[22:56] <JEEBsv> which is what "23.976" is
[23:09] <JustinInTheFlesh> I have a problem that I believe has to do with directshow. I changed libavformat\rtp.c to have a few custom payload types due to a device I am trying to stream to. Whenever I capture from directshow to a file, the quality is perfect. From a file, I can stream to multicast and it is played without an issue. If I directly stream from dshow, I get a lot of dropped packets. However, if I stream to a multicast, use a second
[23:09] <JustinInTheFlesh> version of ffmpeg to catct the stream and rebroadcast without re-encoding, the device plays the stream without issue.
[23:09] <JustinInTheFlesh> And suggestions?
[23:12] <JustinInTheFlesh> Anyone that can assist?
[23:26] <blink> yes, I know that I can pass it a fraction; is that the preferred way to express it for ffmpeg?
[23:27] <blink> thanks JEEBsv, i'll try -r before i
[00:00] --- Tue Jun 25 2013