[Ffmpeg-devel-irc] ffmpeg.log.20130408

burek burek021 at gmail.com
Tue Apr 9 02:05:01 CEST 2013


[01:42] <gamax92> Hello, I was wondering if ffmpeg does compression on its gif files. They seem rather large when outputted to.
[01:47] <gamax92> http://pastie.org/7358089
[01:48] <gamax92> http://pastie.org/7358119 fixed.
[01:54] <ubitux> we definitely need to improve the gif encoder
[01:54] <gamax92> I'm pretty sure lzw is nolonger patented
[01:59] <gamax92> Is there some option that im missing or does it just not compress it]
[02:01] <ubitux> the lzw is done
[02:01] <ubitux> but frames are intra only
[02:02] <ubitux> afaik
[02:39] <schtinky> last time I was here, I was told to "ffmpeg -f v4l2 -i /dev/dvbdevice" to grab digital video from my tuner rather than "gnutv | ffmpeg -i -"
[02:40] <schtinky> Unfortunately, I get "operation not supported" when specifying my device as /dev/dvb/adapter0/frontend0 which is what I think my tuner card is on the machine
[02:40] <schtinky> when I google for answers, everyone references analog devices (e.g. /dev/video0)...
[02:40] <schtinky> Is capturing this way really possible with digital devices and if so, what am I doing wrong? Thanks in advance!
[02:44] <gamax92> schtinky: have you tried catting the device and then redirecting it to ffmpeg?
[02:44] <gamax92> cat /dev/dvb/adapter0/frontend0 | ffmpeg -i -
[03:24] <sinusss> hi!
[03:25] <sinusss> how do I use qt-faststart in this command? ffmpeg -i $input_file$ -threads 0 -c:a libfaac -c:v libx264 -preset medium $video_quality$ $audio_bitrate$ $audio_sample_rate$ $keyframes$ $fps$ -y video_tmp_noqt.mp4\nqt-faststart video_tmp_noqt.mp4 $output_file$
[03:28] <sacarasc> IIRC, you can make ffmpeg do it without qt-faststart now, but I do not remember how exactly.
[03:33] <sinusss> hmmm thanks sacarasc. will be searching about that...
[03:34] <sinusss> movflags +faststart
[04:38] <praveenmarkandu> Hi. I've been making mpeg-ts files for a while now. It seems to be compatible with HLS
[04:39] <praveenmarkandu> however does FFMPEG actually do proper HLS
[04:39] <praveenmarkandu> or a more proper question is, what HLS version does FFMPEG support
[04:49] <gamax92> If im encoding to mpeg4, what range is the -q:v from
[04:49] <gamax92> Someone told me it was from 1 to 31
[04:52] <praveenmarkandu> i think it starts from 0
[04:53] <praveenmarkandu> http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide
[04:53] <praveenmarkandu> actually it is until 51
[04:53] <praveenmarkandu> oh wait that is for x264
[04:53] <praveenmarkandu> what codec are you using
[04:54] <gamax92> mpeg4?
[04:55] <praveenmarkandu> errrm im pretty sure mpeg4 isnt a codec
[04:56] <gamax92> Then I don't know what you mean
[04:58] <gamax92> praveenmarkandu: http://pastie.org/7359856
[04:58] <gamax92> does that help
[05:01] <praveenmarkandu> gamax92: whats your commandd
[05:01] <gamax92> ffmpeg -r 30 -i screenshot_%06d.png -q:v 1 Videos\tptvideo0001.avi
[05:04] <praveenmarkandu> ahh i see what you are doing now
[05:06] <praveenmarkandu> i dont see that -q parameter in the man ffmpeg
[06:00] <schtinky> gamax92, (1) catting doesn't work (to my knowledge) on digital devices. Only analog ones. Hence, the use of gnutv. (2) The pipe is exactly what I'm trying to get rid of. I've been using the pipe semi-successfully for some time, as I mentioned.
[08:45] <langoliers> hi, anybody can suggest a program for converting tiff image to 10 color HP printer format?
[08:56] <pratz> Hello guys
[08:56] <pratz> I am following this guide to install ffmpeg https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[08:56] <pratz> But when compiling ffmpeg I get this error - ERROR: libfdk_aac not found
[09:01] <zap0> installed it?
[10:01] <RSDRSDRSD> This question must be asked a lot of times, but I am struggling with it for a long time
[10:02] <RSDRSDRSD> I want to encode to videos with a max bitrate of 1000k
[10:03] <RSDRSDRSD> The input bitrate differs all the time, but when the input bitrate is lower than the target bitrate it shouldn´t use the target bitrate but the input bitrate
[10:03] <RSDRSDRSD> what is the command for this
[10:03] <RSDRSDRSD> i want to encode to mp4 and web,
[10:03] <RSDRSDRSD> webm
[10:16] <RSDRSDRSD> anyone?
[10:42] <bogdanp> I'm receiving individual sample buffers from a camera and I have managed to turn the video samples into AVFrames. I'm wondering if there are any examples that I can have a look at where AVFrames are used to create video files (if that's possible at all). Basically, I want to be able to stream camera input into a file.
[11:42] <FlowRiser> bogdanp, yes, look it up in the api; you can 'add' frames to a file
[11:42] <FlowRiser> i don't remember the params exactly, but it can be done
[12:20] <Rahul_21> hello i am new here,i wanted to convert files from .wav to .ogg so that i can upload them to wikimedia commons,can i get help here?
[12:21] <viric> vorbistools do that too
[12:21] <viric> it's a less generic tool, so easier to learn. oggenc.
[12:22] <Rahul_21> ok i will have a look at that ,thank you
[12:24] <Rahul_21> viric, and is their source code easy to understand
[12:24] <Rahul_21> ?
[12:25] <viric> ah, do you need to write a program that converts from .wav to .ogg?
[12:25] <viric> can't you call oggenc?
[12:27] <Rahul_21> ill explain
[12:28] <Rahul_21> i am recording pronunciation and saving them as .wav files then i need to convert them into .ogg so as to upload them to wikimedia commons
[12:28] <Rahul_21> so according to my purpose what will be appropriate viric ?
[12:28] <viric> for every wav file, call the oggenc program to get an ogg file.
[12:29] <viric> you don't have to read any source code. Simply read: oggenc --help
[12:37] <japjap> hello
[12:37] <japjap> does anyone know about wowza IRC channel ?
[12:41] <relaxed> Rahul_21: ffmpeg -i input.wav -c:a libvorbis -b:a 192k output.ogg
[12:44] <Rahul_21> relaxed, ok thats the shell command,but i actually want to automate this as part of my extension so what files/folders should i be including?
[13:01] <relaxed> you want to find and encode all audio to ogg, or...?
[13:04] <Rahul_21> relaxed, as soon as i download the .wav file a script should run which should convert it .wav to .ogg
[13:06] <relaxed> you should be able to script that in bash
[13:08] <Rahul_21> and in that script ill have to include oggenc/ffmpeg . am i right?
[13:09] <relaxed> right, and I would drop oggenc and just use ffmpeg
[13:10] <Rahul_21> hehe
[13:17] <Rahul_21> relaxed, i am gettin 2 errors
[13:18] <Rahul_21> ffmpeg -i input.wav -c:a libvorbis -b:a 192k output.ogg
[13:19] <Rahul_21> relaxed, error--->"Unrecognized option 'c:a'
[13:19] <Rahul_21> Failed to set value 'libvorbis' for option 'c:a'
[13:19] <Rahul_21> "
[13:29] <relaxed> you ffmpeg is older, use -acodec libvorbis & -ab 192k
[13:29] <relaxed> your*
[13:53] <bogdanp> http://pastebin.com/raw.php?i=64NTn3KZ swsCtx ends up being null, what am I missing?
[13:59] <bogdanp> oh wow, nvm; I got my params mixed up
[14:18] <Rahul_21> relaxed, one more question
[14:19] <Rahul_21> relaxed, a pure js solution to convert .wav to .ogg?
[14:41] <viric> Rahul_21: I don't think anyone wrote js code about that
[14:42] <megaTherion> Rahul_21: you'd have to be able to talk to an converting backend
[16:28] <DrSlony> Hi, the x264 guide says "You can see a list of current presets with -preset help", how do I run that?
[16:30] <DrSlony> eix x264
[16:31] <DrSlony> ignore the eix line
[16:33] <klaxa> dunno, x264 wiki lists everything you have to know though i think
[16:33] <klaxa> http://mewiki.project357.com/wiki/X264_Settings
[16:36] <DrSlony> i use x264 through ffmpeg and i cant figure out how to list profiles, presets and tunings
[16:37] <klaxa> hmm neither can i, but here's the guide for ffmpeg with libx264 http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide
[16:48] <DrSlony> klaxa yeah thats the one i read
[16:52] <relaxed> DrSlony: x264 --fullhelp | less
[16:58] <DrSlony> relaxed I don't have x264, I use it through ffmpeg
[16:58] <DrSlony> I'm a little confused why a ffmpeg website would describe the use of a non-ffmpeg executable
[17:12] <viric> x264 has its own system of presets, apart of that of ffmpeg
[17:31] <DrSlony> Is it possible to remove the last 2 seconds of video without knowing the length of the video?
[17:31] <DrSlony> like -ss but for the end
[17:40] <tmatth> i'm having trouble doing git bisect between tags n0.9 and 0.8.14
[17:41] <tmatth> namely, after the first bad case, i get "Some good revs are not ancestor of the bad rev"
[17:54] <DrSlony> Is there a ffmpeg-endorsed or recommended encoder GUI for Windows that does x264? I'm updating a guide I wrote years ago where I recommended installing ffdshow and avidemu. Is there an easier way now?
[17:55] <DrSlony> i see winff/avanti/ffmpeggui
[17:57] <DrSlony> no, ffmpeggui is long dead
[18:20] <schtinky> Hi everybody. Easy question: Is it possible to "ffmpeg -f v4l2 -i /digitaltuner ..." just like you can with an analog tuner? I've been unable to get it to work and forced to "gnutv stdout | ffmpeg -i - " instead. Any thoughts or advice would be appreciated.
[18:20] <orak3l> hello, im trying to encode a .wav file with amr_wb my command is: ffmpeg -y -i stunt.wav -acodec amr_wb -ar 23.85k -ac 1 stunt.amr
[18:20] <orak3l> as a result i get Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[18:21] <orak3l> what to do now?
[18:21] <klaxa> are you sure 23.85k as rate is supported?
[18:22] <orak3l> i have read here
[18:22] <orak3l> http://ffmpeg.org/doxygen/0.6/libopencore-amr_8c-source.html
[18:22] <orak3l> at lin 243
[18:49] <mark4o> orak3l: -ar is sample rate, -ab is bitrate.  amr_wb supports 16k sample rate only, and various bitrates up to 23.85k
[19:16] <orak3l> mark4o: thank you, ive got it working now
[19:54] <bourbon> so like... is there any way to just excplicitly set the DAR and SAR that ffmpeg should assume
[19:54] <bourbon> rather than reading it from a video file?
[19:55] <mark4o> bourbon: -vf setsar=
[20:01] <bourbon> ah, cool
[20:02] <bourbon> thanks!
[20:04] <bourbon> well, poop - that didn't save me
[20:04] <bourbon> http://hastebin.com/vofecuguvo.vhdl < what happened
[20:05] <bourbon> I've got a guy who's sending malformed video data apparently as created by some iPhone app out of a screen capture
[20:05] <bourbon> he's flipping the video using the accepted CoreData solution for flipping video
[20:05] <bourbon> but the result has a negative SAR and DAR (as is seen in the paste)
[20:20] <llogan> bourbon: multiple filters are used like -vf filter1a,filter2a,filter3a;filter1b,filter2b
[20:20] <llogan> see filterchain and filtergraph in the docs http://ffmpeg.org/ffmpeg-filters.html#Filtergraph-description
[20:20] <bourbon> a fair point... tweaking retrying
[20:21] <bourbon> unfortunately, it changes nothing
[20:21] <bourbon> is there any way to set the SAR/DAR of the input video?
[20:21] <bourbon> I'm assuming that this is how you set an arbitrary SAR/DAR on the output video
[20:22] <llogan> The setdar and setsar filters accept a string in the form num:den expressing an aspect ratio
[20:22] <llogan> http://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar
[20:24] <llogan> and i don't think you can apply -vf to the input, but i may be wrong
[20:25] <bourbon> right... I was just hoping there was some way to bypass the stuff the video itself claims to have
[20:25] <bourbon> because I'm dealing with a negative aspect ratio, and ffmpeg hates that
[20:25] <llogan> i've never seen one of those files in "the wild".
[20:25] <bourbon> http://alan.appredeem.com/wtf.mov
[20:25] <bourbon> wait
[20:25] <bourbon> http://alan.appredeemdev.com/wtf.mov
[20:26] <bourbon> and now you know my full name, where I work, etc
[20:26] Action: bourbon shrugs
[20:26] <llogan> i gave you my ip address in return
[20:28] <bourbon> so it goes
[20:28] <llogan> i guess you were right: num:den syntax is deprecated, please use num/den or named options instead
[20:28] <bourbon> I've got half a mind to tell this guy to stop fucking around
[20:28] <bourbon> yeah, I was
[20:28] <bourbon> http://hastebin.com/hisusameba.vhdl
[20:28] Action: llogan makes note to try to remember to fix that
[20:30] <bourbon> so the fun part is... it orients in the desired direction in one piece of software: apple quicktime
[20:31] <bourbon> when I ffprobe -show_streams
[20:32] <bourbon> I get the stream data trying to report SAR of 0:1 and DAR of 0:1
[20:32] <bourbon> while the stuff that goes to stderr still reports the negative SAR/DAR
[20:32] <bourbon> (-1:1, -40:71)
[20:32] <llogan> bourbon: consider submitting a bug report. ffmpeg should at least give a more user friendly error message.
[20:34] <llogan> make sure to include: ffmpeg -i wtf.mov out.mpg and the complete console output in addition to your sample
[20:35] <llogan> and mention that the sample plays as expected in quicktime
[20:36] <llogan> ...and the ffprobe difference
[20:36] <bourbon> ffprobe the difference between?
[20:37] <bourbon> I can't get ffmpeg to output *anything* from this video due to negative aspect ratio
[20:37] <bourbon> I mean, aside from data about the streams
[20:37] <llogan> no, i meant just mention that ffprobe shows a different value (show the ffprobe command and the console output)
[20:37] <llogan> interestingly ffplay can play it
[20:37] <bourbon> oh, you mean the difference in output between ffprobe's stderr output and the -show_streams output?
[20:38] <llogan> although upsidedown
[20:38] <bourbon> yeah... everything seems to play it, but only quicktime re-orients it
[20:38] <bourbon> the rotation tag is conspiciously missing
[20:38] <llogan> i just mean the -show_streams output. it differs than what ffmpeg shows, right?
[21:45] <Aelingil> I am outputing a video, in the file name is "$(date +%F).avi" However i am only getting "2013-4-8.avi" is there a way to change it so it shows "Monday April 8th 2013.avi" ?
[21:47] <klaxa> refer to the date manpage
[21:47] <CampinSam> so I'm trying to delay my audio (to match up with my video) when I stream, with this command "ffmpeg -f x11grab -s $INRES -r "$FPS" -i :0.0 -f alsa -ar 44100 -ac 2 -itsoffset 4 -i hw:0,0 -map 0:0 -map 1:0 -acodec libmp3lame -vcodec libx264 -s $OUTRES -preset $QUAL  -threads 2 -qscale 3 -b 512k -bufsize 512k out.flv" It seems I have -itsoffset in the right place, but no matter what I set, the audio still seems to stay the same... am I
[21:47] <CampinSam> going about this wrong?
[21:50] <klaxa> Aelingil: something similar would be: $(date "+%A %B %d %Y")
[21:51] <klaxa> that's rather close to what you want
[21:53] <Aelingil> klaxa, I was tying %c at the moment, i was going to give up the time and just look off the "created" time.  Thank you for your help again :)
[22:18] <bourbon> llogan: this is correct. -show_streams is different from what ffmpeg/ffprobe shows without any flags
[22:18] <llogan> weird
[22:26] <bourbon> so weird.
[22:52] <Sc0rc3d> Good Evening guys! :) I hope i can ask a question. i wannt to stream a movie from my nas (with ssh and ffmpeg) to my root server (red5 media server) that i can watch a movie @work/hotel. :D but it dont will work any ideas this is my string: ffmpeg -re -i test.avi -acodec copy -vcodec copy -f flv rtmp://x.x.x.x:1935/test   i got an error: Input/output error
[22:54] <Mavrik> Sc0rc3d, you got several lines of output
[22:54] <Mavrik> not only last error
[22:54] <Mavrik> so read those
[22:54] <Mavrik> I'm also quite sure whatever you have in avi can't be put into flv :)
[22:55] <Sc0rc3d> hmm yeah ffmpeg is complete a new page for me ;)
[22:55] <Sc0rc3d> can i just stream it or i need to convert?
[22:56] <Sc0rc3d> Seems stream 0 codec frame rate differs from container frame rate: 23.98 (65535/2733) -> 23.98 (24000/1001) Input #0, avi, from 'test.avi':   Metadata:     encoder         : VirtualDubMod 1.5.10.2 (build 2540/release)   Duration: 02:45:22.07, start: 0.000000, bitrate: 2004 kb/s     Stream #0.0: Video: mpeg4, yuv420p, 720x304 [PAR 1:1 DAR 45:19], 23.98 tbr, 23.98 tbn, 23.98 tbc     Stream #0.1: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s 
[22:56] <Sc0rc3d> is the full error
[22:56] <Mavrik> can you put your whole output to a pastebin? gonna be alot easier that way
[22:57] <Sc0rc3d> sure
[22:59] <Sc0rc3d> therer you go http://pastebin.com/iL3Chydr
[23:00] <Mavrik> hmm, you won't be able to stuff those two streams into flv
[23:00] <Mavrik> also as you can see, the time bases don't match for flv
[23:00] <Mavrik> (FLV is a pretty annoying container)
[23:01] <Mavrik> so you'll have to reencode
[23:01] <Mavrik> not just remux
[23:02] <Sc0rc3d> hmm two streams? no one local file to the server it will be the same with mkv?
[23:02] <Sc0rc3d> mkv is the normal container i watch
[23:02] <Mavrik> of course two streams - one video (mpeg4) and one audio(ac3)
[23:02] <Mavrik> afaik FLV can't contain ac3
[23:02] <Sc0rc3d> aahh okay
[23:02] <LithosLaptop> what about TS?
[23:03] <Mavrik> yeah, what LithosLaptop proposed
[23:03] <Mavrik> Sc0rc3d, check if red5 supports MPEG2-TS over UDP as input
[23:03] <Mavrik> that should work
[23:03] <Mavrik> if not, just reencode to H.264 / AAC
[23:04] <Sc0rc3d> Streaming Video (FLV, F4V, MP4, 3GP) Streaming Audio (MP3, F4A, M4A, AAC)  and just Protocols: RTMP, RTMPT, RTMPS, and RTMPE
[23:04] <Mavrik> Sc0rc3d, input or output?
[23:05] <Sc0rc3d> i think input
[23:05] <Mavrik> Sc0rc3d, basically use something like
[23:05] <Sc0rc3d> so just for my head i need to reencode the mkv to flv than i can try to stream it
[23:06] <Mavrik> ffmpeg -re -i test.avi -acodec libfaac -vcodec libx264 -crf 25 -preset medium -ab 128k -ac 2 -f flv rtmp://...
[23:06] <Mavrik> this should give "ok" quality
[23:06] <Mavrik> higher crf = less quality (And less bitrate)
[23:06] <Sc0rc3d> hmm without to reencode?
[23:07] <Mavrik> with reencode of coure.
[23:07] <Mavrik> *of course
[23:07] <LithosLaptop> does it need to be FLV?
[23:07] <Sc0rc3d> so this process will make my tiny cute nas sweating? :D
[23:07] <Sc0rc3d> no mp4 too
[23:08] <Sc0rc3d> Streaming Video (FLV, F4V, MP4, 3GP)
[23:08] <LithosLaptop> MP4 can contain C3 audio
[23:08] <Mavrik> ah
[23:08] <LithosLaptop> AC3
[23:08] <Mavrik> Sc0rc3d, then just switch to mp4
[23:08] <Mavrik> couldn't find if red5 supports mp4
[23:08] <Sc0rc3d> so like  ffmpeg -re -i test.avi -acodec libfaac -vcodec libx264 -crf 25 -preset medium -ab 128k -ac 2 -f mp4 rtmp://...
[23:08] <Sc0rc3d> ?
[23:09] <Mavrik> Sc0rc3d, no, as you had it
[23:09] <Sc0rc3d> red5 is pretty awsome the live streming over obs works so great :)
[23:09] <Mavrik> with "copy" (which means don't reencode that stream)
[23:09] <Mavrik> just replace -f flv with -f mp4
[23:09] <Mavrik> even though I'm not 100% sure ffmpeg supports that :\
[23:11] <Sc0rc3d>  ffmpeg -re -i test.avi -acodec copy -vcodec copy -f mp4 rtmp://x.x.x.x:1935/test  same error
[23:14] <LithosLaptop> HMM
[23:17] <LithosLaptop> ffmpeg -re -i test.avi -acodec libmp3lame -ab 128k -ac 2 -vcodec copy -f flv rtmp://x.x.x.x:1935/test
[23:17] <Sc0rc3d> i try
[23:19] <Sc0rc3d> [mpeg4 @ 0x806a1c0] Invalid and inefficient vfw-avi packed B frames detected ... Unknown encoder 'libmp3lame'
[23:19] <Sc0rc3d> nolame
[23:20] <LithosLaptop> oh
[23:20] <Santi_> is there recent changes in Byte range request handing by Youtube? i am trying to optimize a youtube video(flash) using ffmpeg on the fly and get errors
[23:20] <Santi_> while using Browsers
[23:20] <Santi_> but works fine on Devices?
[23:20] <Santi_> like i phone and i pad
[23:21] <LithosLaptop> Sc0rc3d: try libfaac, but use a higher bitrate like 192
[23:21] <LithosLaptop> ibfaac sucks at 128kbit/s :)
[23:21] <LithosLaptop> *libfaac
[23:22] <Sc0rc3d> damn
[23:22] <Sc0rc3d> no libfaac xD
[23:22] <LithosLaptop> hmmmm
[23:22] <Sc0rc3d> just      libavutil    51.  9. 1 / 51.  9. 1   libavcodec   53.  8. 0 / 53.  8. 0   libavformat  53.  5. 0 / 53.  5. 0   libavdevice  53.  1. 1 / 53.  1. 1   libavfilter   2. 23. 0 /  2. 23. 0   libswscale    2.  0. 0 /  2.  0. 0   libpostproc  51.  2. 0 / 51.  2. 0
[23:22] <LithosLaptop> you on windows?
[23:23] <Sc0rc3d> no NAS system
[23:23] <LithosLaptop> ah ok
[23:23] <Sc0rc3d>  cat /proc/version Linux version 3.4.6 (root at NasX86-10) (gcc version 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2)) #1 SMP Fri Mar 1 06:34:04 CST 2013
[23:23] <LithosLaptop> you could.....use aac
[23:24] <LithosLaptop> -acodec aac -strict -2
[23:24] <LithosLaptop> but then use a bitrate of 300 or more
[23:25] <Sc0rc3d> [mpeg4 @ 0x806a1c0] Invalid and inefficient vfw-avi packed B frames detected ...  Input/output error
[23:25] <Sc0rc3d> i think i quit ^^
[23:25] <LithosLaptop> haha
[23:26] <LithosLaptop> that the only error
[23:26] <LithosLaptop> ?
[23:26] <Sc0rc3d> and input output yes i think so
[23:27] <LithosLaptop> ok try -vcodec libx264
[23:27] <LithosLaptop> instead of copy
[23:27] <Sc0rc3d> lol Unknown encoder 'libx264'
[23:28] <LithosLaptop> oh lol
[23:28] <Sc0rc3d> i think i should update somehow ffmpeg
[23:29] <LithosLaptop> you will have to build ffmpeg with the required libraries you want
[23:29] <LithosLaptop> I don't know how
[23:29] <LithosLaptop> haha
[23:29] <LithosLaptop> https://ffmpeg.org/trac/ffmpeg/wiki/WikiStart
[23:30] <LithosLaptop> Compiling guides on the wiki
[23:30] <Sc0rc3d> i think is use vnc to connect to my desktop and stream it with obs then it works but my pc needs to be on ^^
[23:31] <Sc0rc3d> thanks for the link
[23:31] <Sc0rc3d> and the help :)
[23:33] <LithosLaptop> no prob
[23:36] <Santi_> is there changes in query parameter in Youtube requests
[00:00] --- Tue Apr  9 2013


More information about the Ffmpeg-devel-irc mailing list