[Ffmpeg-devel-irc] ffmpeg.log.20120604

[burek]

[00:01] <robertzaccour> oh ok
[00:06] <qxt> how do I record internal sound with my cmd. ffmpeg -v 1 -r 30 -f x11grab 1920x1080 -i :0.0 -vocdec huffyuv -threads 0 out.mkv
[00:08] <qxt> how would I record internal sounds (music) with the above cmd?
[00:09] <burek> what is "internal sound" ?
[00:09] <burek> the microphone input on your audio card?
[00:09] <qxt> speaker output
[00:10] <burek> your audio card has to support the capture of audio output
[00:10] <burek> so, just try this
[00:10] <burek> ffmpeg -v 1 -r 30 -f x11grab 1920x1080 -i :0.0 -f alsa -i pulse -vocdec huffyuv -threads 0 out.mkv
[00:12] <qxt> Do I need to have pulse installed? Normally I use ALAS
[00:12] <qxt> on Debian
[00:12] <qxt> alsa*
[00:13] <sacarasc> No, you don't.
[00:13] <qxt> thx guys. Ill give it a try.
[00:25] <CuZnDragon> humm the headre has it.  core/core_c.h:CVAPI(IplImage*)  cvCreateImageHeader( CvSize size, int depth, int channels );
[00:29] <CuZnDragon> it is in /usr/include/opencv2
[00:30] <qxt> Sound coming out of my speakers it not being recoded with the above cmd. I did install pulse though.
[00:30] <qxt> still no dice
[00:34] <qxt> just a guess but dont I need something like -ac 2 -ab 128 for the sound?
[00:35] <burek> CuZnDragon, you might have all the files you need, but the version of those files is either too new or too old, i.e. it doesn't have expected structure, so you get an error
[00:35] <burek> try reading docs to see which version of opencv do you actually need
[00:35] <burek> qxt, I'm not sure, but I think you can do it using jack audio
[00:36] <burek> to create a virtual recording device
[00:36] <burek> and map it to your output
[00:36] <burek> something like that
[00:36] <burek> or, try to use: arecord -l
[00:36] <burek> (small L)
[00:36] <burek> and try various combinations of: -f alsa -i hw:0
[00:36] <burek> or hw:1, hw:2, etc..
[00:38] <qxt> was on hw:0 for once on this machine. Spent so much time dicking around with pulse that was a relief
[00:38] <Neil1> Hey folks - i'm having trouble adding a preset to my installation.  I'm running ubuntu but cant seem to find the correct syntax to use a preset in ~/ffmpeg/presets - can anyone advise?
[00:39] <Neil1> do i need to add new presets to a config table or compile them or someting?
[00:42] <burek> qxt, if you have multiple audio devices, you can use arecord -L
[00:42] <burek> and specify an exact device you want, instead of ordinal number
[00:43] <burek> for example -i front:CARD=ICH5,DEV=0
[00:44] <burek> Neil1, what exactly is your goal, because h264 presets have been now mapped to x264
[00:44] <burek> so no more files needed
[00:44] <burek> all you do is type: x264 --help
[00:44] <burek> and use those native -preset -profile and -tune with ffmpeg
[00:44] <burek> ffmpeg -i ... -preset ... -profile ... -tune ...
[00:45] <Neil1> ah!  ok - i'm reading info on websites that's obviously old news then - more reading, just make sure it's current then!
[00:46] <qxt> burek good old arecord never fails :) Got it working with min hassle thx
[00:47] <burek> :beer: :)
[01:27] <zap0> when i decode a 5.1 compressed stream into PCM how is the data layed out?
[07:40] <Zeranoe> I'm trying to get familar with FFmpeg's libavfilter and am using the simple example given in the doc, yet it fails with: http://pastebin.com/ivYNDJCE
[07:52] <ubitux> unknown paste id
[08:05] <Zeranoe> http://pastebin.com/puV5DEzh
[08:17] <ubitux> [overlay @ 0000000003027dc0] Overlay area (0,120)<->(854,600) not within the mai
[08:17] <ubitux> n area (0,0)<->(854,240) or zero-sized
[08:17] <ubitux> the example looks quite complicated for what's it's doing
[08:28] <Zeranoe> that was th first example given on in the doc. Do you know of a better one?
[10:53] <rainmaker1> Heh, how do I calculate -bufsize size? I can't find any reference/formula about this
[13:52] <Spider-Pork> Hi. I would to stream rtp sound with ffmpeg-ffplay. I run "ffplay rtp://127.0.0.1:10000" from one terminal, then to another one: "ffmpeg -i <myfile.wav> -acodec pcm_alaw -f rtp rtp://127.0.0.1:10000".  Iget an error from ffplay: "Unable to receive RTP Payload type 72 without an SDP file describing it". What am I missing? Thank you
[13:54] <burek> you are missing sdp file
[13:54] <burek> try using udp instead
[13:55] <burek> ffmpeg run like this: ffmpeg -i input -f mpegts udp://127.0.0.1:10000
[13:55] <burek> and then ffplay: ffplay udp://localhost:10000
[13:55] <Spider-Pork> ok thank you burek. How can I configure an SDP file?
[13:55] <Spider-Pork> thank you
[13:55] <Spider-Pork> thank you fflogger
[14:00] <Spider-Pork> ffplay says: unable to seek back to the start
[14:00] <Spider-Pork> no sound then
[14:00] <burek> can you please use pastebin.com, to show your command line and its output?
[14:01] <Spider-Pork> yup
[14:09] <Spider-Pork> ow, i cant paste output from my terminal to ideone
[14:09] <Spider-Pork> ctrl+shift doesn't works
[14:09] <Spider-Pork> ops
[14:09] <Spider-Pork> ctrl+ins
[14:14] <burek> try ffmpeg ... 2>out.txt
[14:14] <burek> or ffplay
[14:22] <Spider-Pork> burek: http://ideone.com/HIa14
[14:24] <burek> could you try changing ffmpeg cmd line with this: ffmpeg -i input -acodec aac -strict experimental -ab 96k -ac 2 -ar 44100 -f mpegts udp://127.0.0.1:10000
[14:25] <Spider-Pork> no results
[14:26] <burek> can you please use pastebin.com, to show your command line and its output?
[14:26] <Spider-Pork> yup
[14:26] <Spider-Pork> if i do your command with > output.txt it works
[14:26] <burek> do you hear music playing with ffplay
[14:27] <burek> and it's not ">output.txt" it's "2>output.tx"
[14:27] <burek> we are redirecting stderr not stdout
[14:27] <burek> txt*
[14:29] <Spider-Pork> i would to redirect all stdout because my ctrl+ins don't work with chrome
[14:29] <Spider-Pork> I'm not using WM like kde or gnome
[14:29] <burek> ffmpeg doesn't use stdout for err log
[14:29] <burek> it uses stderr
[14:29] <burek> so 2>out.txt
[14:29] <burek> not 1>out.txt or >out.txt
[14:30] <burek> ffmpeg -i ... 2>out.txt
[14:30] <Spider-Pork> this is the output http://ideone.com/qnRrc
[14:31] <burek> that looks good
[14:31] <burek> what does ffplay say?
[14:31] <Spider-Pork> unable to seek back to the start
[14:31] <burek> did you restart ffplay
[14:31] <Spider-Pork> yep
[14:32] <Spider-Pork> this is the sample http://www.wav-sounds.com/cartoon/bugsbunny1.wav
[14:32] <Spider-Pork> downloaded it with wget
[14:32] <Spider-Pork> (if you want to try in your machine)
[14:33] <burek> well the sample is too short
[14:33] <burek> so ffmpeg probably just encodes it even before you are able to start ffplay
[14:33] <burek> :D
[14:33] <burek> try this
[14:33] <Spider-Pork> i start ffplay before ffmpeg
[14:33] <burek> ffmpeg -re -loop 1 -i input.wav -acodec aac -strict experimental -ab 96k -ac 2 -ar 44100 -f mpegts udp://127.0.0.1:10000
[14:34] <Spider-Pork> Option loop not found ^_-
[14:35] <burek> oh, it's for video only..
[14:35] <burek> it sucks..
[14:36] <burek> try without -loop 1
[14:37] <burek> ffmpeg -re -i input.wav -acodec aac -strict experimental -ab 96k -ac 2 -ar 44100 -flags +global_header -f mpegts udp://127.0.0.1:10000
[14:37] <burek> that should work
[14:39] <burek> it would be a funny idea to put on someone's machine, in the crontab, a simple: ffplay udp://localhost:10000
[14:39] <burek> and then remotely send him various audio sounds :D
[14:39] <burek> he would think his machine is cursed :D
[14:40] <Spider-Pork> again no audio on first launch. On the second one (without close ffplay) the audio start to play with one loop
[14:40] <Spider-Pork> (only one repetition)
[14:40] <burek> yes
[14:40] <burek> ffplay buffered something
[14:40] <burek> from before's attempts
[14:40] <burek> when it threw that error
[14:40] <Spider-Pork> ok thank you burek for your help
[14:40] <burek> so next time when you set global header, it played both
[14:40] <burek> :)
[14:40] <burek> :beer: :)
[14:41] <Spider-Pork> ok, I'll send you by email
[14:41] <burek> :)
[14:41] Action: Spider-Pork try to fit beer inside his notebook)
[14:41] <Spider-Pork> what i should do to stream from audio input with pcm?
[14:42] <vladan> hi, I'm trying to build ffmpeg on arch with --enable-librtmp but I get this error --   ERROR: librtmp not found
[14:42] <vladan> I have rtmpdump-2.3
[14:42] <burek> you need rtmp devel
[14:42] <burek> most probably rtmpdump-2.3-dev
[14:43] <burek> Spider-Pork, why pcm
[14:44] <vladan> burek thanks
[14:48] <Spider-Pork> I'll move to Wm machine, is impossible to work without copy&paste to chrominum
[14:48] <Spider-Pork> bbs
[14:58] <Spider-Pork> with "ffmpeg -f alsa -i hw:1,0 -acodec pcm_alaw -f mpegts udp://127.0.0.1:10000" it start stream but i can't hear no sound
[14:58] <burek> try adding -flags +global_header
[14:58] <Spider-Pork> with arecord -D hw:1,0 output it works (so the hw is correct)
[14:58] <Spider-Pork> ok
[14:59] <Spider-Pork> nope, always mute
[15:03] <Spider-Pork> ffmpeg -f alsa -i hw:1,0 -acodec pcm_s16le output.wav works (so hw is correct)
[15:10] <Spider-Pork> I have no idea :/
[15:10] <burek> are you sure pcm_alaw can be put into mpegts?
[15:13] <Spider-Pork> nope
[15:13] <Spider-Pork> I used mpegts only because you suggest to me
[15:16] <Spider-Pork> but damn pcm is the oldest codec in the world
[15:16] <burek> so?
[15:16] <burek> it doesn't mean it's streamable
[15:16] <burek> just like wav
[15:17] <burek> you should look into aac or mp3
[15:17] <burek> I suggest aac
[15:17] <burek> ffmpeg -f alsa -i hw:1,0 -acodec aac -strict experimental -flags +global_header -f mpegts udp://127.0.0.1:10000
[15:18] <Spider-Pork> i need to stream in lossless way
[15:18] <burek> why
[15:18] <Spider-Pork> because i need entire audio, no audio with numeric elaboration
[15:19] <Spider-Pork> in the process of lossy compress some data will be lost
[15:19] <Spider-Pork> i need that data
[15:19] <burek> use ftp then :D
[15:19] <Spider-Pork> ftp for realtime audio streaming?
[15:20] <Spider-Pork> aac doesn't work too
[15:22] <burek> use flac then
[15:22] <burek> ffmpeg -f alsa -i hw:1,0 -acodec flac -flags +global_header -f mpegts udp://127.0.0.1:10000
[15:23] <Spider-Pork> nope again...
[15:24] <burek> well, then wait for somebody else that might help you, I'm out of ideas..
[15:24] <Spider-Pork> burek: no problem thank you for the help :)
[15:26] <burek> this works for me
[15:26] <burek> ffmpeg -f alsa -i hw:0 -acodec libaacplus -ab 48k -ac 2 -ar 44100 -flags +global_header -f mpegts udp://127.0.0.1:10000
[15:26] <Spider-Pork> with mic?
[15:26] <burek> aac+ is really a superb codec
[15:27] <burek> read on wikipedia about it
[15:27] <Spider-Pork> burek: I know is a superb codec but i need to have entire sound, without compression
[15:27] <burek> and if you really really need lossless, then I don't know, try using google to find something
[15:28] <burek> btw, what do you need it for?
[15:28] <Spider-Pork> AAC, like MP3 remove sound that your ear can't hear. But the mic can hear it, and with some post process you can restore that portion of sound and finally hear it
[15:29] <Spider-Pork> compressing with AAC/AAC+/MP3 will cut that info and there is no way to get after post-processing
[15:29] <burek> I'm aware of that
[15:30] <burek> so, what do you need it for?
[15:31] <Spider-Pork> to post-process the sound and have that hidden info
[15:32] <Spider-Pork> the hidden info can be a discussion near a louder device
[15:32] <Spider-Pork> (for example)
[15:32] <Spider-Pork> or in a noisy ambient
[15:32] <burek> and you need that in real-time
[15:32] <Spider-Pork> yep
[15:32] <Spider-Pork> wait
[15:32] <Spider-Pork> the hidden info can be fetched in post-processing work
[15:33] <Spider-Pork> with pcm (or other lossless codec) i can restore it
[15:33] <Spider-Pork> with lossy no
[15:34] <burek> man, no need to repeat yourself
[15:34] <burek> I've understood you the first time
[15:35] <carnau> Hello, Is there any way to list which options I could pass to ffmpeg compilation, based on the codecs I've installed previously?
[15:35] <burek> I'm just asking why do you need real-time + lossless? what's the use case for that kind of thing?
[15:35] <burek> carnau, how did you install your codecs previously
[15:35] <Spider-Pork> burek: restore particular sound from noisy ambient
[15:35] <Spider-Pork> and other security stuff
[15:35] <burek> you can try sonicls
[15:36] <burek> instead of aac
[15:36] <carnau> burek, ./configure, make && make install
[15:36] <burek> it's lossless
[15:36] <burek> see if it works for you
[15:36] <Spider-Pork> thank you burek
[15:36] <burek> carnau, check the config.log
[15:36] <burek> it shows options which you used
[15:36] <carnau> burek, ok, thank you!
[15:37] <mystica555> out of curiosity, when doing -ss before the input file, to decode all frames, why does the ffmpeg process seem to not discard the decoded data and instead bloat to multiple gigabytes of memory?
[15:37] <Spider-Pork> I think there is something wrong in my PC, i can't hear sound
[15:37] <Spider-Pork> it can register perfectly to file but no output from ffplay
[15:38] <burek> well Spider-Pork there is some missunderstanding of how things work from your side
[15:38] <burek> it's not the same if you are listening to the sound on the same machine you are creating it
[15:38] <burek> (using files)
[15:38] <burek> with the streaming the sound to another machine (using udp and ffplay)
[15:39] <Spider-Pork> I'm streaming to localhost, so it should works if ffplay is listening to localhost
[15:40] <carnau> burek, config.log it's the output of the codec compilation. What I mean is, if I install h264, when I want to configure ffmpeg I use "./configure --enable-libx264". But dunno where to get every argument for each codec.
[15:45] <burek> carnau, in ffmpeg directory, type ./configure --help
[15:45] <burek> and read :)
[15:46] <burek> Spider-Pork, you still don't get it
[15:46] <burek> it doesn't matter it's localhost
[15:46] <burek> what matter is that you don't have the access to the sound data as if it was file
[15:46] <burek> you can't seek to the beginning, to the end, in the middle..
[15:46] <burek> like you can when you use file access
[15:47] <carnau> burek, mmmmm interesting... thank you so much :-)
[15:47] <burek> with streaming, you only have the access to the data you just received
[15:47] <burek> and if, for example, you use some audio codec/format that keeps important data at the end of the file (when the file has finished encoding, encoder wrote some index stuff there)
[15:47] <burek> then you have a problem streaming audio in that codec/format
[15:48] <burek> shortly, you need to choose codec/format that can packetize your audio data
[15:48] <burek> so that it can be sent over network
[15:48] <burek> and each packet needs to have all the info needed to play that chunk of audio data
[15:48] <burek> I hope it is more clear now what your issue actually is
[15:48] <burek> carnau, :beer: :)
[15:54] <Spider-Pork> burek: thank you for the info :)
[15:55] <Spider-Pork> I'm still trying to fit beer inside my PC
[16:09] <carnau> burek, last one, I promise :-). If I've configured my codecs with --enable-shared option, I have to specify the option when configuring ffmpeg, or I can use --enable-shared to get them all?
[16:16] <burek> carnau, what exactly is your goal?
[16:17] <burek> if you specify --enable-shared while configuring ffmpeg, then all the libav* libraries will be built as shared objects too
[16:17] <burek> it doesn't hint ffmpeg that you compiled your codecs using --enable-shared
[16:18] <burek> the best practice (IMO) is to build all the dependent libraries (codecs) using --enable-shared --enable-static
[16:19] <burek> just to be sure later when you need different types of builds that you don't have to rebuild all of them again
[16:19] <burek> but, you do whatever it suits your needs
[16:20] <carnau> well I'm quite new with ffmpeg. Just have to configure a server that can process my videos(adds them an animation and a text), an convert them to mp4.
[16:23] <carnau> I haven't previously experience with video processing, so I'm a bit lost. So, I'ts time to read and read more... Thank you for your hints, burek.
[16:23] <burek> :beer: :)
[16:27] <dougle> does anyone have an idea why:
[16:27] <dougle> CMD="ffmpeg -xerror -y -f image2 -i ... </dev/null"
[16:27] <dougle> bash -c $CMD
[16:27] <dougle> would hang a bash script?
[16:27] <dougle> i am running a job queue of ffmpeg commands (various params)
[16:29] <dougle> i have heard that ffmpeg will sometimes eat all stdin and pause if </dev/null isn't directed in, but am having same problem with or without </dev/null
[16:30] <dougle> running the command normally from the prompt works like a charm
[16:31] <burek> dougle, what is your ffmpeg version
[16:32] <dougle> http://pastebin.com/V97XrKFw    latest from git (built on Jun  2 2012 18:34:22)
[16:33] <burek> oh, you didn't redirect output/err
[16:33] <burek> try like this
[16:33] <dougle> as in 2> ?
[16:33] <burek> ffmpeg ... </dev/null >/dev/null 2>&1
[16:33] <dougle> right will give a try
[16:36] <dougle> it's hanging the same way
[16:36] <dougle> the mp4 is output but the script does not continue
[16:37] <burek> what do you mean does not continue?
[16:37] <burek> it's blocked until the encoding is finished?
[16:37] <dougle> after the ffmpeg command i have some cleanup code for input files but they remain
[16:37] <dougle> am running the script with bash -x and can see the last command being ffmpeg
[16:37] <burek> maybe it's still encoding?
[16:38] <dougle> yes i think ffmpeg is not returning or some kind of exit code
[16:39] <dougle> it's 150 images at 15 fps
[16:39] <burek> well, add -t 5
[16:39] <burek> just to test
[16:39] <dougle> only 500kb mp4s, definitely finished encoding
[16:40] <burek> and what's the purpose of -xerror?
[16:40] <dougle> the result of my scrabbling around in the ffmpeg docs
[16:40] <dougle> i read it was to exit on error
[16:41] <dougle> but prob not applicable to the latest version
[16:41] <dougle> "-xerror             exit on error"
[16:42] <burek> so, your script runs and stops at ffmpeg command and never finishes?
[16:42] <dougle> yes correct
[16:42] <dougle> what is -t?
[16:42] <dougle> threads ?
[16:42] <burek> no, duration of the output
[16:42] <burek> that'll make ffmpeg end in 5 seconds approx
[16:43] <burek> i.e. output file will be 5 seconds of duration
[16:43] <ubitux> 't' for "turation"
[16:43] <dougle> :D
[16:44] <burek> anyway, try debugging the exact ffmpeg cmd line
[16:44] <burek> at the command prompt
[16:44] <burek> and see what does it do
[16:45] <dougle> on command prompt all recent command param combinations have worked well
[16:46] <dougle> and dropped me back at a prompt
[16:46] <dougle> -t 5 paused in the same way, though an mp4 5 seconds long was output as expected
[16:48] <burek> ok, try creating new batch script with just that line in it
[16:48] <burek> chmod +x
[16:48] <burek> and run it
[16:48] <dougle> ah good idea, will give it a go
[16:51] <svenstaro> hi, I got a few questions about the new release 0.11
[16:51] <svenstaro> blender doesnt compile anymore and that is kinda bad since we need to rebuild for new ffmpeg
[16:52] <svenstaro> I run into a lot of errors and can't find porting notes for the new version anywhere
[16:52] <svenstaro> for instance
[16:52] <svenstaro> /build/src/blender-2.63a/source/blender/blenkernel/intern/writeffmpeg.c:710:5: error: 'AVFormatContext' has no member named 'mux_rate'
[16:52] <svenstaro> /build/src/blender-2.63a/source/blender/blenkernel/intern/writeffmpeg.c:713:4: error: 'AVFormatContext' has no member named 'preload'
[16:52] <svenstaro> where did those 2 go?
[16:52] <Mavrik> I don't think those notes exists actually :\
[16:52] <svenstaro> well that sucks for me :<
[16:53] <Mavrik> there are some notes in the docs directory though
[16:53] <Mavrik> grepping through those could probably give you a lot of answers
[16:53] <Mavrik> what's the reason that you need new ffmpeg?
[16:54] <svenstaro> well this is arch linux
[16:54] <svenstaro> and doc/ doesn't even match mux_rate
[16:55] <svenstaro> I'm the blender maintainer and just noticed that nobody had yet tried to build against new ffmpeg
[16:56] <svenstaro> it wouldnt be so bad if I could find info
[16:57] <Mavrik> svenstaro: you should probably map out all the API calls that are problematic and check ffmpeg.c for "proper" calls
[16:57] <Mavrik> svenstaro: and for those you can't match / know how to use send mail to [libav-users]
[18:29] <svenstaro> where did mux_rate go now?
[18:37] <TristamWrk> I'm trying to rotate a video by an increment less than 90 degrees, anyone know if/how this might be done preferably with ffmpeg?
[20:24] <dougle> hey burek thanks for your help earlier, i was calling my script through socat that way timing out after (the default) 0.5 seconds
[20:24] <dougle> so all logging paused
[20:25] <dougle> you win a :beer: though for setting your mind to it :D
[20:26] <burek> :beer: :)
[20:28] <dougle> you should set up a flattr account or something for contribs
[20:29] <dougle> i'd send you a few ¬s (or $s or £s)
[20:29] <cbsrobot> last years ffmpeg demo (burek is on the right): http://25.media.tumblr.com/tumblr_lvrwj30zuP1qbwvhpo1_500.jpg
[20:30] <cbsrobot> dougle: if you want to send burek some beer
[20:30] <cbsrobot> it needs to fullfill certain conditions
[20:31] <cbsrobot> f.ex: tahts ok: http://cdnimg.visualizeus.com/thumbs/3a/40/fit,thinspo-3a4099ed6bb1fffb58039a80ae845219_h.jpg
[20:31] <dougle> lol
[20:31] <cbsrobot> that is not: http://nomnommonster.com/wp-content/uploads/2010/02/021110-beer-girl-e1265899845405.jpg
[20:32] <dougle> http://farm4.static.flickr.com/3120/2650710435_cc4831f6d4.jpg
[20:32] <dougle> i get the idea
[20:38] <iam8up> i was here a couple of days ago and i'm making a video out of still images (jpegs) and this command normally works: cat *.jpg|ffmpeg -f image2pipe -r 10 -vcodec mjpeg -i - -vcodec libx264 output.mp4
[20:38] <iam8up> but the problem is that the number of files is making cat complain the argument list too long and hence ffmpeg gets no input
[20:39] <pasteeater> never seen that before
[20:39] <pasteeater> pastebin the output
[20:39] <iam8up> http://pastebin.com/hpdH3xBU
[20:40] <iam8up> even -> cat *|head gives me the same error (argument list too long)
[20:40] <iam8up> i couldn't find any good answer as to what kind of image2pipe expects, but it seems ls -1 doesn't work (which gives the same output)
[20:41] <cbsrobot> iam8up: that the limit of your shell
[20:42] <pasteeater> are your images in sequential, uniform order starting with "1"?
[20:42] <cbsrobot> normally I use "find" if the argument list is too long - but that probably wont work when piping to ffmpeg
[20:42] <iam8up> right, makes sense, so what solution?
[20:43] <iam8up> ffmpeg didn't seem to like find's output
[20:43] <iam8up> pasteeater, they are not sequential
[20:43] <iam8up> wait
[20:43] <iam8up> they are sequential but not like 1/2/3/4/5/etc...can't think of the word
[20:44] <pasteeater> you could rename them all to work with ffmpeg and then: ffmpeg -i img%d.jpg
[20:44] <pasteeater> inelegant, but will probably work
[20:44] <iam8up> maybe, wouldn't be complete hell to test the theory
[20:45] <pasteeater> how many images are there?
[20:45] <iam8up> 11725
[20:46] <pasteeater> less than i imagined
[20:46] <iam8up> they're named image`date +%s`.jpg
[20:47] <sacarasc> You could put a bunch into different directories, then turn them into lossless videos and then merge the videos... :o
[20:47] <iam8up> i suppose that would be a good idea, though it seems kind of inelegant again
[20:48] <iam8up> i have an ARG_MAX of 131072, i'm going to guess that's bytes
[20:48] <pasteeater> the ffmpeg faq has an example to rename. http://www.ffmpeg.org/faq.html#How-do-I-encode-single-pictures-into-movies_003f
[20:48] <pasteeater> never tried it myself
[20:48] <iam8up> but changing that would require a new kernel...and that's not great...
[20:49] <iam8up> that looks like the solution i'll go with if all the work is done
[20:49] <iam8up> especially if it's in a faq, seems at least one other person has been down this road =)
[20:50] <pasteeater> also this: http://pastebin.com/X4gPBkiY
[20:51] <mzcl-mn> hello
[20:52] <pasteeater> but the example in the faq is symbolic, so that might be an advantage
[20:52] <cbsrobot> pasteeater: I guess the example in the docs will have the same problem
[20:52] <cbsrobot> for i in *jpg
[20:52] <pasteeater> hm. didn't think of that.
[20:52] <cbsrobot> will also hit the limit
[20:53] <pasteeater> no beer for us
[20:53] <cbsrobot> use find !
[20:53] <mzcl-mn> i am trying to watch dvb-t which is coded in a h.264 codec and i am having these issues: http://pastebin.com/AjaAn1G2 can the problem be related to x264 codec?
[20:54] <pasteeater> x264 is an encoder only, not a decoder
[20:54] <mzcl-mn> pasteeater: thanks
[20:54] <pasteeater> short answer: no
[20:55] Action: pasteeater returns to git bisecting
[20:56] <iam8up> looks like it is processing the movie...so that's a good sign...
[20:56] <mzcl-mn> pasteeater: and what about this sample http://pastebin.com/T9X1aDha that points to libav?
[20:56] <iam8up> pasteeater, the soft links are easier on disk usage, but that's very good too =)
[20:56] <mzcl-mn> [aac_latm @ 0x7f2998052fc0] SSR not implemented. Update your Libav version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[20:56] <mzcl-mn> [aac_latm @ 0x7f2998052fc0] If you want to help, upload a sample of this file to ftp://upload.libav.org/incoming/ and contact the libav-devel mailing list.
[20:56] <mzcl-mn> net_buf_ctrl: dvbspeed 99.5% @ video 1980 ms 354 buffers
[20:57] <pasteeater> This is FFmpeg.
[20:57] <pasteeater> libav is another project.
[20:57] <mzcl-mn> ok
[20:57] <iam8up> sounds like you need to update libav =P
[20:57] <pasteeater> or use ffmpeg
[20:57] <pasteeater> but i'm biased
[20:57] <iam8up> i have no idea what libav is, sorry =(
[20:58] <pasteeater> (although you'll probably get the same message)
[20:58] <pasteeater> are you using ubuntu?
[20:58] <mzcl-mn> iam8up: and pasteeater thanks
[20:58] <mzcl-mn> pasteeater: yep
[20:58] <mzcl-mn> using it
[20:59] <pasteeater> they dropped ffmpeg for libav because the debian/ubuntu maintainer is a libav guy
[20:59] <mzcl-mn> pasteeater: why do you ask ah ok
[20:59] <iam8up> reason 4292845 to hate debian
[20:59] <mzcl-mn> and do you have grievances amongst you?
[20:59] <mzcl-mn> lol
[21:00] <mzcl-mn> sorry
[21:00] <iam8up> i've not like debian for quite some time, it's all personal taste
[21:00] <mzcl-mn> so do you think i should abandon deb
[21:00] <pasteeater> more like annoyances
[21:00] <mzcl-mn> ok
[21:00] <iam8up> uh i wouldn't start...some people love debian and hate rhel
[21:00] <mzcl-mn> so i am kind of trapped in here
[21:00] <iam8up> if you're going to use ffmpeg only, ubuntu is a bad choice
[21:00] <mzcl-mn> and u?
[21:01] <mzcl-mn> u have a rhel?
[21:01] <pasteeater> there is always this: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[21:01] <iam8up> centos myself
[21:01] <pasteeater> and you can test recent ffmpeg without installing it too
[21:01] <PineappleCLock> Is there a libavfilter constant for video framerate and/or duration that is usable in fading out?
[21:02] <pasteeater> and this since you mention centos: https://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide
[21:02] <mzcl-mn> I got here from there
[21:02] <PineappleCLock> I want to fade out 3 seconds from the end without needing to precompute the samplerate / video duration
[21:03] <mzcl-mn> if you ask me I got to linux myself due to ub
[21:04] <iam8up> for sure that's very much what ubuntu does
[21:04] <mzcl-mn> which is sort of a entrance gate
[21:04] <iam8up> the guy has a bunch of money, liked debian, expanded and marketed
[21:05] <mzcl-mn> ok i know that is somehow sinful
[21:05] <mzcl-mn> still I didn't pay him any and am using it too
[21:06] <pasteeater> PineappleCLock: i think the fade filter requires that you know what frames you want to fade.
[21:06] <PineappleCLock> ok thanks just wondering if there was a constant I could use; I'll precompute it
[21:06] <mzcl-mn> as I did fedora :( but somehow when using it got messed up with ffmpeg or libav.. one or the other don't remember it now
[21:07] <pasteeater> PineappleCLock: that would be nice. a feature request in the bug tracker would be a good start.
[21:07] <mzcl-mn> summing up you don't think updating ffmpeg or something of the like would help me at all?
[21:08] <mzcl-mn> or, pasteeater, should I give it a try?
[21:08] <mzcl-mn> to your ffmpeg?
[21:08] <pasteeater> always worth a try.
[21:09] <mzcl-mn> ok
[21:09] <mzcl-mn> one more thing and maybe this question should be made to libav
[21:10] <mzcl-mn> is it easy to replace it (libav) for ffmpeg?
[21:11] <pasteeater> not really for stuff in the repository that depends on libav, but if you're not using any of that then yes.
[21:12] <mzcl-mn> pasteeater: and iam8up thank you very much for all
[21:24] <iam8up> so the video came out, looks great, but it stops on saturday around th emiddle of the day; i have images from saturday sunday and some of this morning too
[21:25] <iam8up> i was thinking img%03d.jpg was 3 digits but %04d and %05d say no such file or directory?
[21:26] <LexSfX> do all of the numbers have the same number of digits?
[21:26] <LexSfX> in your file names
[21:26] <iam8up> no
[21:26] <LexSfX> make sure they do
[21:26] <iam8up> i used the faq method
[21:26] <LexSfX> if they don't, it won't work properly
[21:27] <LexSfX> for %04d, for example, it depends on starting at 0000
[21:27] <iam8up> ok that's what i want
[21:27] <iam8up> thank you
[21:34] <iam8up> is frame= 2539 fps= 17 q=25.0 size=   26079kB time=00:04:09.00 bitrate= 858.0kbits that output of how long it's been processing or the timestamp of the output video?
[21:35] <mzcl-mn> pasteeater: can i barge once more?
[21:38] <mzcl-mn> these are my libavcodec-extra-53 dependants /http://pastebin.com/ryJN0QcS i guess i am in a lot of trouble am i not?
[21:39] <mzcl-mn> i mean to be able to use them or some of them in the future if i want to move to ffmpeg
[21:40] <bearclaw> Hi. Is the h264 decoder outputing frames in the correct display order?
[21:46] <iam8up> so i did the images with %05d and ffmpeg with %05d and the new video is the exact same size =(
[22:22] <pasteeater> mzcl-mn: you can keep libavcodec-extra-53 and use a compiled ffmpeg.
[22:23] <pasteeater> you can compile ffmpeg and just not install it and still run it.
[22:30] <iam8up> it seems my problem is once it hits too many 0 byte files in a row it quits and gives up
[22:30] <iam8up> i just deleted all of the 0 byte files and it's gotten much farther than before
[22:31] <iam8up> thanks a ton for your help you guys!
[00:00] --- Tue Jun  5 2012