[Ffmpeg-devel-irc] ffmpeg.log.20120220

burek burek021 at gmail.com
Tue Feb 21 02:05:01 CET 2012


[04:36] <T3mp3sT> I use to use this cmd for xgrab. Now I am getting lossless_untrafast not found
[04:36] <T3mp3sT> ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 "$outputdir"/"$1"
[04:37] <T3mp3sT> Did something change?
[04:37] <T3mp3sT> Was about a year ago I was using lossless_ultrafast
[04:41] <T3mp3sT> Looking under /usr/local/share/ffmpeg at the presets. Seems to be some missing.
[04:41] <T3mp3sT> I compiled all from git with x264
[04:41] <T3mp3sT> and the config was right afaik
[04:53] <T3mp3sT> Using Debian wheezy and have xgrab before compiling. If that old cmd does not work how do I use xgrab?
[04:53] <T3mp3sT> pulled all gits from today
[05:21] <burek> "I'm used to"
[05:21] <burek> :)
[05:22] <burek> also "lossless_untrafast" does not exist
[05:22] <T3mp3sT> This looks like crap ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 output.mkv
[05:22] <burek> more on that
[05:22] <burek> vpre is deprecated
[05:22] <burek> you should type x264 --help to see what presets are valid
[05:22] <burek> but iirc those are: ultrafast, faster, fast, ...
[05:23] <burek> also, if you want lossless, why don't you use RLE compression
[05:24] <burek> because -crf 0 and "ultrafast" don't go together not even in real life :)
[05:26] <T3mp3sT> DarkShakra or what ever his was was a few years ago gave me some tips on this. Turned out there was a codec (MicroSoft?) that had a really nice color space and was darn nice.
[05:27] <T3mp3sT> Yeah we used ffmpeg the files were darn big....kike 50GB
[05:28] <T3mp3sT> burek so what would you recommend if you did not care at all about size and have a computer faster then God
[05:29] <burek> just a sec
[05:33] <burek> well
[05:33] <burek> it depends on your goal
[05:34] <burek> if your goal is not size nor speed
[05:34] <burek> then use vcodec with uncompressed codec
[05:34] <burek> ffmpeg -codecs
[05:34] <burek> to see all
[05:34] <burek> so you will get pretty much something like bitmap images :)
[05:38] <T3mp3sT> To make a crystal clear copy of what I see on my screen. A 30" HD 1020 screen. My disks can keep up with huge files ( raid 0 OCZ Vertex 3 Max IOPS) so thats all good even with bitmap
[05:38] <burek> so, what is actually your goal
[05:38] <burek> i see
[05:38] <burek> well, then get the software that was written for that
[05:38] <burek> like camtasia
[05:38] <burek> or camstudio
[05:39] <T3mp3sT> Never heard of that. Only use the main repo of debian testing. I can take a look though.
[05:39] <burek> its a windows software
[05:39] <T3mp3sT> lol....
[05:39] <T3mp3sT> no thx
[05:39] <burek> so google for something that works on linux
[05:39] <burek> or spend your time having hard time with ffmpeg
[05:39] <T3mp3sT> ArchLinux or Debian is OK though
[05:39] <burek> saying "no thx"
[05:39] <burek> :)
[05:39] <burek> its your life, your choice :)
[05:40] <T3mp3sT> So you saying this is not possible with this project?
[05:40] <burek> I didn't say that
[05:40] <burek> you can wait for somebody, who had more experience with your problem, to come online and help you, though
[05:41] <T3mp3sT> prolly should go to bed. Its 5:41am here in Sweden. Been coding since...emmm. I think yesterday...or maybe the day before =P
[05:42] <burek> get some sleep or read on
[05:42] <burek> wikipedia about polyphasic sleep
[05:42] <burek> it might help :)
[05:42] <T3mp3sT> or some big fat roofies so I fall asleep on my keyboard.
[05:43] <burek> :)))
[05:43] <burek> or that :)
[05:43] <T3mp3sT> Later. I work this out 2mRO
[05:43] <T3mp3sT> Cya
[05:43] <burek> :beer: :)
[05:43] <burek> o/
[06:19] <JCDenton> I'm getting an error when trying to add a watermark to my video using ffmpeg. I'm using the same command I used on another box, and as far as I remember, it always worked without a problem. The error I get is:
[06:19] <JCDenton> [NULL @ 0x14a0540] Unable to find a suitable output format for '[watermark]'
[06:19] <JCDenton> [watermark]: Invalid argument
[06:19] <JCDenton> ffmpeg -threads 8 -i input.mp4 -vf movie=watermark-small.png [watermark]; [in][watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out] output.mp4
[06:19] <JCDenton> Is what I'm running.
[06:38] <JCDenton> Any idea why?
[07:31] <grepper> JCDenton: "movie=...[out]" quoted ?
[07:32] <grepper> should be if it isn't
[07:43] <JCDenton> grepper: Oh wow, I'm an idiot, thanks
[07:43] <qop> is it possible to get ffmpeg to use an iso or dvd as input?
[07:46] <grepper> np
[07:47] <JCDenton> DVD maybe, ISO I'm not so sure.
[07:47] <qop> how? cant find out
[07:48] <qop> and if dvd is possible I can just mount the iso
[07:48] <qop> ot trick through script
[09:32] <grepper> qop: you could take a look at http://godromo.com/gmt/vgtmpeg
[09:40] <AdamWorld> Hi! When I update FFmpeg git-6cb2085 32-bit Dev (Latest) but "'UINT64_C': identifier not found" error occured. what's the problem.
[09:59] <qop> grepper: thanks!! ^^
[11:34] <elkng> how to procees only first 2 minutes of file ? there is and option "-ss" to seek to position, is there similar option ?
[11:40] <kriegerod> should i rescale pts/dts of incoming AVPacket to decoder timebase before decoding, or i can just rescale the decoded frame pts to encoder timebase after decoding?
[11:40] <kriegerod> elkng, -t
[12:32] <qop> ffmpeg still encoding after 3 days :P
[12:32] <qop> ~23000 files (its 50%)
[12:33] <qop> I have a flight Wednesday! wonder if it will end
[12:33] <qop> \in time
[13:10] <al4711> Hi all
[13:12] <al4711> I try to insert some frames into a movie which was calculated by previous frames => slow motion effect.
[13:12] <al4711> I have try to search with google but I'am not sure if I search for the right syntax "ffmpeg slow motion"
[13:13] <al4711> thanks for help
[14:24] <elkng> there is an example in "man ffmpeg": "ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi", why "snatch" ? why isnt there "lord_of_the_rings.avi" or "star_wars.avi" ?
[14:25] <sacarasc> Maybe the devs liked Snatch.
[14:29] <ubitux> originally committed in 2002
[14:30] <ubitux> (3e12d48a89e3385a7da76cc086333359911c2131)
[14:30] <ubitux> not much about it, maybe in the ml
[15:14] <MindSpark> hello, anyone into some serious debugging?
[15:15] <MindSpark> bind(port 8090): Address family not supported by protocol
[15:15] <MindSpark> that's the output I get when I start ffserver
[15:15] <MindSpark> does anyone know what causes this?
[15:19] <kriegerod> first of all tell us what actually you're doing
[15:19] <ubitux> we don't have a maintainer for ffserver at the moment
[15:19] <ubitux> so i don't think anyway has a gdb open on ffserver now :)
[15:20] <MindSpark> I am trying to convert an rtmp stream to rtsp
[15:21] <MindSpark> I got ffserver to run on different machines. For some odd reason, I just tried on a different one (the computer I am on right now) and I get this error
[15:21] <MindSpark> however even on the other machines, that I got it to run on, I failed to correctly transcode rtmp to rtsp
[16:01] <stryjan_> is there a way to use ffmpeg only to copy metadata between two files without any conversion?
[16:02] <djook> hye guys
[16:03] <djook> I try to use loop input and drawtext at the same time but drawtext doesnt work
[16:04] <djook> I loop a single jpeg picture for video source
[16:04] <djook> and would like to insert text aswell
[16:05] <djook> http://pastebin.com/yAbMELEm
[16:05] <djook> here is my command line
[16:06] <djook> video comes out without the text
[16:06] <djook> is itnormal behaviour ?
[16:07] <djook> any body awaken?
[16:07] <djook> 16h00 in France ...
[16:08] <cbreak-work> hmm
[16:10] <djook> I feel it is not taking the filter into account while looping the image
[16:10] <djook> but doesnt complain
[16:10] <cbreak-work> have you moved the filter after the input?
[16:10] <djook> seems logicalanyway
[16:10] <cbreak-work> I think the order of arguments in ffmpeg is very important
[16:10] <djook> yes I tried to change
[16:12] <djook> wherever I placed it ,itdidnt appear on the final output
[16:13] <djook> I'll try with a non looping input
[16:15] <cbreak-work> maybe your whole drawtext is incomplete
[16:15] <cbreak-work> the one that works for me specifies a font
[16:18] <djook> it complains when taken into account too
[16:18] <djook> and asks for fontfile
[16:19] <djook> when using a non loop input
[16:20] <djook> so I do it in two times
[16:21] <djook> first time,from loop input without drawtext, then again with drawtext from the video file I just obtained
[16:26] <djook> okguys see you
[16:29] <fj_> hi
[16:29] <fj_> I am using the old version of ffmpeg but can' t use pstrcpy
[16:29] <fj_> whats up with that
[16:45] <Freakshow> anyone have examples of how I would multiple bitrates out of a single input?
[16:45] <Freakshow> looking at this as an example
[16:45] <Freakshow> http://smorgasbork.com/component/content/article/35-linux/92-ffmpeg-the-mother-of-all-command-lines
[16:45] <Freakshow> but was curious how applicable it was.
[16:46] <Freakshow> given recent release changes...
[16:47] <Freakshow> holy crap... just saw the date on that...
[16:48] <Freakshow> nevermind, I'll cross reference and see what I can come up with today
[16:48] <Freakshow> and then ask again
[16:48] <kriegerod> is there any usage example of av_asrc_buffer_add_samples() on earth?
[16:51] <kriegerod> Freakshow, that example is over-complicated. You should try "ffmpeg -i in.file -b:v 100k 100k.file -b:v 200k 200k.file" - you get the idea
[16:51] <kriegerod> -b is for bitrate
[16:51] <kriegerod> input file options go before -i option, output file options go before output file name
[17:00] <Freakshow> kriegerod: what if I want to specify multiple rtsp outputs? does the same principle apply?
[17:01] <kriegerod> yes
[17:02] <Freakshow> hmm... is the simple notion of the additional -b:v 200k indicating that it's another output?
[17:03] <Freakshow> so like -b:v 100k -c:v libx264 -profile:v baseline -f rtsp rtsp://output -b:v 200k -c:v libx264 -profile:v baseline -f rtsp rtsp://output
[17:03] <kriegerod> simply options are split by places where input and output file names/URLs are located
[17:04] <kriegerod> your line is right
[17:04] <Freakshow> ok ok
[17:04] <Freakshow> cool
[17:04] <Freakshow> I'll give that a shot, thanks!
[17:04] <MindSpark> I think I just missed the beginning of this conversation
[17:04] <MindSpark> can someone pastebin/send it to me?
[17:05] <Freakshow> sure, hang on
[17:05] <MindSpark> this is probably very similar to what I am trying to achieve
[17:13] <Freakshow> done
[17:43] <AdamWorld> I run ffserver as in document and in client's command I run ffplay http://192.168.157.129:8090/test.asf but it fails to play.
[17:43] <AdamWorld> what's the problem?
[17:43] <AdamWorld> In other player like VLC works well.
[17:52] <MindSpark> AdamWorld, what OS?
[17:52] <MindSpark> what is the output?
[17:55] <MindSpark> can someone tell me how to convert this rtmp stream to rtsp?
[17:55] <MindSpark> rtmp://aljazeeraflashlivefs.fplive.net/aljazeeraflashlive-live/aljazeera_ara_med
[17:58] <Velmont> Hmm. FFmpeg doesn't support old Ogg Vorbis files. I tested with ogg123, and that works. -- Is it intended?
[17:59] <sacarasc> Output to a pastebin, etc.
[18:03] <ubitux> Velmont: a sample would be welcome too
[18:03] <Velmont> http://dpaste.org/3rNO0/
[18:04] <ubitux> [mp3 @ 0xf0a1e0]
[18:04] <ubitux> what :D
[18:04] <Velmont> looked kinda strange
[18:04] <ubitux> can you share the file?
[18:05] <Velmont> http://odin.s0.no/tmp/Marte%20Wulff%20-%2001%20-%20Run.ogg
[18:05] <ubitux> btw, you are using a pretty old version
[18:05] <ubitux> plays fine here
[18:05] <ubitux> try upgrading your ffmpeg
[18:06] <Velmont> Hmm. Strange, - it worked in older versions, - maybe the one Ubuntu 11.04 has is just broken.
[18:06] <sacarasc> Also, using libav, not ffmpeg.
[18:06] <Velmont> sacarasc: ffplay?
[18:07] <Velmont> Oh, yes, it uses it as well.
[18:07] <sacarasc> ffmpeg was forked a few months back, Ubuntu ships the fork.
[18:07] <ubitux> Velmont: i'm testing with ffplay from upstream, and it works
[18:07] <ubitux> and indeed this is libav, not ffmpeg
[18:08] <ubitux> (they still use the ff* tools name? :/)
[18:08] <ubitux> seriously& :(
[18:08] <ubitux> Velmont: nice music btw ;)
[18:08] <sacarasc> I think that's for backwards compatibility.
[18:08] <ubitux> or backward stabbing!
[18:09] <Velmont> ubitux: Yes, I like her. Suddenly wanted to listen to some old stuff, and in Ubuntu's fork it didn't work.
[18:09] <ubitux> https://launchpad.net/~jon-severinsson/+archive/ffmpeg
[18:09] <ubitux> maybe check this out
[18:10] <ubitux> if it still does not work, i can only suggest to update to upstream
[18:11] <Velmont> ubitux: Well, I've been thinking about switching back to arch linux. Was always easier to get upstream stuff etc then.
[18:11] <Velmont> But sysadmin had installed Ubuntu on my work computer when I came here already, so I just went with the flow.
[18:12] <ubitux> :)
[18:14] <ubitux> i'm not sure changing its OS because of a packaging issue is the correct answer to the problem; maybe you should complain on lp (or even better do sth more productive) about that :p
[18:14] <ubitux> though, archlinux doesn't have (yet?) the "libav issue" :)
[18:37] <MindSpark> I can confirm
[18:37] <MindSpark> ubuntu's ffmpeg in 11.04 is just broken
[18:37] <MindSpark> and for me neither ffserver nor ffmpeg can connect to anything
[18:56] <Freakshow> MindSpark: is that stream even working atm?
[18:57] <Freakshow> rtmp://aljazeeraflashlivefs.fplive.net/
[18:58] <Freakshow> nm... mplayer got it
[19:15] <twinther> hi, I have a problem with a rtsp stream where the video freeze after a while (~80 secs)
[19:16] <twinther> this is logged just before the stream stops: [rtsp @ 0x20658e0] CSeq 9 expected, 1 received.
[19:17] <twinther> full log here http://pastebin.com/zTrhxs7N
[19:26] <MindSpark> Freakshow, sorry for the delay, was afaik
[19:26] <MindSpark> Freakshow, yes, it should be working. I am trying to restream it over rtsp, but ffserver is not playing it as far as vlc is loggin
[19:26] <MindSpark> logging
[19:28] <Freakshow> no worries, I was trying to determine if the source feed itself was actually up
[19:29] <Freakshow> flowplayer wasn't picking it up
[19:29] <Freakshow> sorry, my test flowplayer
[19:29] <MindSpark> :)
[19:29] <Freakshow> but mplayer latched onto it quickly
[19:29] <Freakshow> can you paste the actual ffmpeg command line you're using?
[19:30] <MindSpark> yes, playing it directly hasn't been a problem for me. I just don't know what to do to get ffserver to restream it
[19:31] <MindSpark> ffmpeg -i rtmp://aljazeeraflashlivefs.fplive.net/aljazeeraflashlive-live/aljazeera_ara_med -vcodec libx264 -b 300000 -vpre default -vpre baseline -sameq -s 480x270 -acodec copy -re -f rtsp rtsp://127.0.0.1/jazeera.sdp
[19:31] <MindSpark> this is one way I read about that should work
[19:32] <MindSpark> but I get Could not write header for output file #0 (incorrect codec parameters ?)
[19:32] <Freakshow> which version of ffmpeg?
[19:32] <Freakshow> looks like e 0.7.1
[19:32] <Freakshow> but < 0.9.1
[19:33] <MindSpark> 0.6.4-4:0.6.4
[19:34] <Freakshow> ugh... can you update to something newer? head perhaps?
[19:34] <MindSpark> that's the binary I got from ubuntu's package manager, do you think that will solve the problem?
[19:35] <Freakshow> I mean, I understand if not... but yikes, I had some issues with rtmp/rtsp output on the older versions
[19:35] <MindSpark> it says built in jan 2012
[19:35] <Freakshow> not sure about ubuntu's main line
[19:35] <MindSpark> Freakshow, ok, makes sense I guess, maybe I need to recompile
[19:35] <Freakshow> you would leave it be and install ffmpeg-devel
[19:35] <Freakshow> just to test
[19:35] <Freakshow> if you have an alt machine to use, that might be worthwhile
[19:36] <Freakshow> devel on bsd is 0.9.1
[19:36] <Freakshow> not sure what ubuntu devel port is at though
[19:36] <Freakshow> pasteeater may know
[19:38] <MindSpark> alright, I'll give this a shot
[19:40] <MindSpark> I'll just try to compile on my local machine and see from here
[19:40] <MindSpark> but does the above line look valid?
[19:42] <Freakshow> yes, that seems valid...
[19:43] <MindSpark> so I don't need to run ffserver to restream?
[19:43] <MindSpark> using rtsp://link as an ouput would be enough to open up rtp/rtsp ports?
[19:43] <Freakshow> well... I presume you have ffserver on the local machine
[19:43] <Freakshow> you may need to declare a port
[19:44] <Freakshow> but I'm not sure... I stream to wowza, not ffserver
[19:44] <Freakshow> but that does seem valid from what I recall
[19:44] <Freakshow> with the newer version you'll probably want to force rtsp over tcp
[19:44] <Freakshow> it defaults to udp, and it's not pretty
[19:46] <MindSpark> Freakshow, yes, do you know how this is done?
[19:46] <Freakshow> sure... -f rtsp -rtsp_transport tcp rtsp://rtsp.server.com:1935/my/stream/path.sdp
[19:47] <Freakshow> would be what I would use
[19:47] <Freakshow> and which is also 'what' I use
[19:47] <Freakshow> ;)
[19:49] <Freakshow> but again, this is to wowza...
[19:49] <MindSpark> :) alright, sounds good. Is wowza hosted somewhere or do I download and install?
[19:52] <Freakshow> oh boy... that's an entirely different ball of wax
[19:52] <Freakshow> try with ffserver first
[19:52] <Freakshow> and yes, you DO need to specify the protocol to localhost
[19:52] <MindSpark> yea, I am tight on time, I am supposed to find a solution to this by the end of the week
[19:53] <MindSpark> and I've already spent a week trying to get things done with ffserver
[19:53] <Freakshow> well, spend the hour or so getting the newer version installed and see where that gets you
[19:55] <MindSpark> alright, thanks. I'll be back later I guess
[19:55] <MindSpark> Freakshow, what's your timezone?
[19:55] <Freakshow> PST
[19:55] <Freakshow> you?
[19:56] <MindSpark> EET
[19:56] <MindSpark> I guess that's like 9 hours apart
[19:56] <Freakshow> looks like
[19:59] <MindSpark> I am going to install a new version on some amazon cloud instance
[20:00] <MindSpark> actually, 0.8-4:0.8
[20:00] <MindSpark> is that an "ok" version?
[20:02] <Freakshow> well, I'm not necessarily sure that the previous version was not ok, but it's always best to use closer to head as possible
[20:02] <Freakshow> security fixes, etc.
[20:37] <pasteeater> Velmont: there is a ubuntu compile guide link in the channel topic
[20:46] <pasteeater> i finally found "open command window here" in win7.
[21:00] <ago> Hi folks
[21:02] <ago> I have just a pair of doubt about cpu/arch optimizations
[21:03] <ago> When starts configure phase, I see: ARCH x86 (core2) but this is an x86_64 environment, it is ok or should be x86_64/amd64/other?
[21:04] <ago> I I enable cpudetenction, in configure phase I see always --cpu=host, is expected?
[21:04] <ago> *If I enable
[21:07] <burek> ago
[21:07] <burek> can you please use pastebin.com, to show your command line and its output?
[21:10] <ago> burek: as you can see I didn't make it manually, is a gentoo environment http://paste.pocoo.org/show/554139/
[21:10] <ago> and I see --cpu=host when I use ffmpeg
[21:11] <burek> does it work?
[21:12] <ago> burek: yes
[21:13] <ago> burek: there is a list of argument that --cpu accepts?
[21:13] <ago> e.g. as -march core2 corei7 prescott etc.
[21:13] <burek> I don't see --cpu in your paste?
[21:13] <burek> just type ./configure --help
[21:13] <burek> you'll get all the info you need I guess
[21:14] <ago> burek: http://paste.pocoo.org/show/554143/
[21:14] <cbreak> trying to cross compile?
[21:14] <ago> no
[21:14] <cbreak> did you compile the app?
[21:14] <ago> help does not show what I need
[21:14] <ago> cbreak: yes
[21:14] <cbreak> then use file
[21:14] <cbreak> file ffmpeg
[21:14] <cbreak> it should tell you which architectures are in the binary
[21:15] <abec0> Hi there. I gott this nice command line that records my x11 actions, but I'd like to stream this video instead. First on LAN, then on the Internet.
[21:15] <burek> --cpu=CPU                select the minimum required CPU (affects...
[21:15] <burek> it does show
[21:15] <cbreak> well... at least it does on OS X :)
[21:16] <abec0> So if someone has a link to a working example, I would really like to hear from it.
[21:16] <burek> abec0, instead of output file, put -f mpegts udp://dest.ip.addr:port
[21:17] <abec0> burek: That would stream to one host on the LAN ?
[21:17] <ago> cbreak: no, I mean a different thing
[21:17] <burek> abec0, yes, for example to a vlc, which can act as a http server
[21:17] <burek> to rebroadcast it
[21:17] <cbreak> ago: what does it show?
[21:18] <ago> cbreak: when you use --enable --disable you can activate or not an 'option' but when you have --foo=bar I think you should have a list of supported arguments that can be $bar
[21:18] <ago> where I can see the syntax to pass to --arch options?
[21:18] <ago> e.g
[21:18] <abec0> burek: OK. So that would be vlc that does the broadcasting. I should go check the man page to understand what mpegts means.
[21:19] <cbreak> ago: arch accepts things like ppc
[21:19] <ago> cbreak: you know it but i'm asking where is documented ;)
[21:19] <cbreak> Don't know. I read the configure script
[21:20] <ago> cbreak: e.g for ppc could be obvious, but for amd64, it can be: amd64, em64t, x86_64
[21:20] <ago> and there is no explanation about it
[21:20] <cbreak> I've only used armv7, armv5, i386 and the default
[21:20] <burek> mpeg-ts = transport stream, abec0, but you can also use ffserver
[21:20] <burek> its just the matter of choice
[21:20] <cbreak> ago: but you said you don't want to cross compile
[21:20] <cbreak> so why do you want to change it?
[21:21] <abec0> ffserver. I'll go do some reading/testing and report back. Thanks.
[21:21] <burek> :beer: :)
[21:22] <abec0> That might compromise the reading, but... ;-).
[21:26] <barhom_> I'm looking for a developer that is available for hire that has good knowledge of transcoding video, mainly from mpeg2-ts to h264-ts. Im wishing to build some "dynamic" transcoder that would serve the transcoded video+audio as a HTTP stream. What I mean with dynamic is that whilst there are users watching the stream, then transcode, when last one quits then stop transcoding and free up CPU power. One could use lib curl (f
[21:26] <barhom_> fetching HTTP mpeg2-ts), transcode with libavcodec (h264) and then serve the stream with libmicrohttpd
[21:27] <barhom_> If there is anyone that is interested please private message me so we can discuss it in more detail. If anyone can refer me to someone else that isn't on this IRC, please do that as well.
[22:10] <PeraZdera1> Hi
[22:12] <PeraZdera1> I am trying to find out why ffmpeg does not 'like' some switches, although I compiled the last git version
[22:12] <abec0> burek: about «instead of output file, put -f mpegts udp://dest.ip.addr:port», I tried -f localhost:1234, but can't find a way to read the thing.
[22:14] <PeraZdera1> here is pastebin http://pastebin.com/BMF711ix
[22:15] <abec0> Well I tried -f mpegts udp://localhost:1234, to be precise... but don't know if I should do something before with vlc, or after, or what I should do.
[22:16] <pasteeater> PeraZdera1: is there a reason you are not using the libx264 encoding presets? declaring each x264 option is not supported.
[22:18] <PeraZdera1> pasteeater: Well thats another "problem", all the tutorials I have been reading does state that there are .ffpresets files, but I can only find libx264-ipod320.ffpreset and libx264-ipod640.ffpreset in ffmpeg presets directory
[22:19] <burek> open vlc and go to media - open network stream and type:    udp://@:1234
[22:20] <pasteeater> PeraZdera1: they are now "internal" rahter than text files.
[22:21] <PeraZdera1> pasteeater: Thank you for your answer, can you tell me then is this valid anymore? https://sites.google.com/site/linuxencoding/x264-ffmpeg-mapping
[22:21] <PeraZdera1> pasteeater: I mean can I create my own preset files?
[22:22] <pasteeater> that's a waste of time, generally
[22:23] <PeraZdera1> Ok, thank you very much for your answers
[22:23] <pasteeater> PeraZdera1: a good start would be...
[22:23] <pasteeater> ffmpeg -re -i udp://230.0.0.7:5000 -r 25 -c:v libx264 -preset slow -level 30 -b:v 1000k -c:a libfaac -ar 44100 -b:a 96k -f flv rtmp://192.168.1.10/ffmpeg
[22:23] <pasteeater> -preset slow contains most of the options you previously were using
[22:24] <pasteeater> -threads 0 is now automatically applied, so you don't need that anymore
[22:25] <PeraZdera1> Aha, I will try to find which options are in slow profile
[22:25] <PeraZdera1> Thank you :)
[22:26] <pasteeater> note that there is a difference between presets and profiles
[22:26] <PeraZdera1> Huh?
[22:27] <pasteeater> see x264 --fullhelp for a list of options (in addition to default options) that are applied for each preset
[22:27] <pasteeater> a preset is not a profile. there is a -preset option and a -profile option
[22:28] <pasteeater> if you want to overwrite any preset options then see "ffmpeg -h" under the "libx264 AVOptions" section
[22:29] <pasteeater> ...for a list of libx264 specific options.
[22:29] <abec0> burek: Output #0, mpegts, to 'udp://localhost:8512'
[22:31] <abec0> But vlc says:
[22:31] <abec0> [0x8fcd47c] main stream error: cannot pre fill buffer
[22:31] <abec0> When I use STOP. Using PLAY did not play a thing.
[22:35] Action: abec0 regrets screencasting is not as simple as screenshooting... 
[22:35] <abec0> I'll try again later.
[22:36] <barhom_> (out of topic from you guys) the output ramp:// to what media server are you posting the transcoded stream to?
[22:36] <barhom_> *rtmp://
[22:37] <pasteeater> screencasting to justin.tv isn't too hard
[22:40] <barhom_> btw, does anyone know if you can pipe the output from ffmpeg to vlc so that you can stream it from an http link, or should I use something else?
[22:42] <barhom_> curl -s http://10.0.0.101:2004/bysid/5 | ffmpeg -i - -acodec libfaac -b:a 96k -vcodec libx264 -b:v 900k -preset superfast -threads 4 -bt 2500 -qcomp 0.7 -deblockalpha 2 -deblockbeta 2 -deinterlace -f mpegts 1.ts < something like this, but instead of saving to 1.ts make it available at http://something/1.ts without saving to file
[22:50] <PeraZdera1> pasteater: what is the best aac encoder for ffmpeg? I can see that many people say that internal is still unstable and that is best option to use libfaac, but also it is very slow. Does libaac have some bugs when very little people use it or it is something else?
[22:53] <pasteeater> PeraZdera1: i'm not sure about any libfaac bugs.
[22:54] <pasteeater> you should try the interal encoder and if it sounds good enough for you then use it.
[22:54] <pasteeater> there is also libvo_aacenc and libaacplus, so you have 4 AAC type encoders to choose from.
[22:55] <PeraZdera1> pasteeater: and libaacplus nobody use?
[22:56] <pasteeater> it's probably good for low bitrates. i've never used it.
[22:56] <PeraZdera1> pasteeater: I see, I am just looking for the most stable encoder, as I am trying to transcode live stream and stability is pretty much everything
[22:57] <PeraZdera1> pasteeater: Is it ok if I use daemontools for ffmpeg monitoring? I mean I would like in a case of crash to start ffmpeg again
[22:58] <pasteeater> i don't know. i have no experience with that.
[22:59] <PeraZdera1> tnx, you helped me very much
[22:59] <PeraZdera1> and sorry for bothering :)
[22:59] <pasteeater> i know ffmpeg has the -d option to enable "daemon mode", but i know nothing of it and there is little documentation as far as i know
[22:59] <pasteeater> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=39aafa5ee90e10382e2f991e4d79d7852231df68
[23:03] <PeraZdera1> well, it works, thats for sure, I can see the stream
[00:00] --- Tue Feb 21 2012


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