[Ffmpeg-devel-irc] ffmpeg.log.20121123
burek
burek021 at gmail.com
Sat Nov 24 02:05:01 CET 2012
[03:02] <Verte> Just doing research at the moment. I have a situation where I'll have three (possibly more later) video files/streams coming in, where I want to take all three streams and create a single output stream that periodically switches between the inputs with a fade or something. If I can have the other two streams appear with a picture-in-picture type arrangement that would be icing on the cake, but not necessary. My initial research s
[03:02] <Verte> ays ffmpeg should do the trick, properly set up. Just checking that I'm not barking up the wrong tree before I start.
[04:22] <clopez> hi
[04:23] <clopez> i have been trying for several hours trying to encode video with ffmpeg and libvpx in webm format using VBR without having to specify a target bitrate
[04:24] <clopez> and it always defaults to a bad quality... it skips the "-q:v" value
[04:48] <clopez> ok... seems that raising qmin up to 15 solves the issue
[04:50] <dgbaley27> Is there a way to scan the whole file for streams? I just joined a bunch of vobs using -i contact:a.vob|b.vob|... -c copy all.mpg. I want to convert them to x264 now, but the streams available in each file are different
[04:51] <burek> omg
[04:51] <burek> dgbaley27, try reading our wiki first
[04:51] <burek> search for concat or join or merge
[04:52] <dgbaley27> burek: I read those FAQ items. It doesn't address what happens with discontinuities in audio/cc/subtitle streams.
[04:53] <burek> what discontinuities
[04:54] <dgbaley27> VOB_1 and VOB_3 have a particular stream, but VOB_2 doesn't.
[04:56] <burek> what particular stream? audio/sub?
[04:57] <burek> or video
[04:57] <burek> whichever it is, you can add a dummy one
[05:02] <dgbaley27> burek: http://pastie.org/5421465. I was hoping to make a giant vob/mpg that showed all of those streams at any point during playback, with any timestamps correct.
[05:03] <burek> oh man...
[05:03] <dgbaley27> Then convert that into x264 or whatever gets a better file size
[05:03] <burek> it's easier to answer your first question than to dive into this more :)
[05:03] <burek> Is there a way to scan the whole file for streams? <-- see -analyzeduration
[05:04] <dgbaley27> burek: I tried analyzeduration but it always stops with "max_analyze_duration 5000000 reached"
[05:04] <burek> then you didn't read carefuly about it
[05:05] <burek> ffmpeg -analyzeduration 100000000 -i ...
[05:06] <dgbaley27> Same thing
[05:08] <dgbaley27> http://pastie.org/5421480
[05:09] <burek> what's the problem with that output?
[05:10] <burek> oh that's the joined one?
[05:10] <burek> well ffmpeg reads the header of the first file i guess :)
[05:10] <burek> that's why you'll never get all thestreams
[05:11] <burek> in that case, put some insane value for -analyzeduration
[05:13] <dgbaley27> burek: I made it with http://pastie.org/5421497
[05:13] <dgbaley27> burek: yeah but no matter what I choose for analyzeduration, it times out with "max_analyze_duration 5000000 reached"
[05:13] <burek> ok, give me an example
[05:13] <burek> because in your last output example
[05:14] <burek> i didnt see max_analyze_duration 5000000 reached
[05:14] <burek> that one http://pastie.org/5421480
[05:16] <dgbaley27> ah yeah. but it exited in a second, there's no way it searched the whole 11 GiB file. I guess it's possible that all of these streams didn't make it from the concat.
[05:16] <dgbaley27> The joined file doesn't even show the all streams from the first VOB
[05:18] <burek> i guess ffmpeg reads the header (container), gets the info about the streams
[05:18] <burek> and starts analyzing them to see what they are
[05:18] <burek> as soon as it figures all them out, it stops
[05:18] <burek> no need to analyze anything anymore, right?
[05:19] <dgbaley27> Well, at the moment I have the need. But I could be completely wrong about trying to rip this double-sided movie.
[05:19] <burek> also im not sure many players would be able to playback that kind of concatenated otput
[05:19] <dgbaley27> The FAQ says this format works for concatenation like this.
[05:20] <burek> well, try adding -map 0 after -i '...'
[05:25] <dgbaley27> burek: Do you mean a -map for every stream?
[05:26] <burek> just one -map 0
[05:26] <burek> after your -i concat...
[05:26] <burek> btw
[05:26] <burek> did you read how concat works?
[05:28] <burek> "All segments must have the same number of streams of each type, and that will also be the number of streams at output."
[05:28] <dgbaley27> ah
[05:29] <dgbaley27> So I think I need -i ... -i ... -i ... and the concat filter (rather than the concat input format)
[05:29] <burek> no
[05:30] <burek> i don't know exactly what's your plan
[05:30] <burek> but you need to read a bit more on how multimedia works :)
[05:30] <burek> so that you know what your options are :)
[05:32] <dgbaley27> I've read, including a lot of options, and still don't know what all my options are. This is not a straightforward thing to do.
[05:32] <burek> why concating in the first place?
[05:35] <dgbaley27> Because it's a DVD video with 1G .VOBs and I want a single file, in an up-to-date format, with all of the a/v/s streams. The timestamps shouldn't be broken, chapters would be nice, and I think that the file would be more reliable if every stream was available throughout the entire playback.
[05:36] <burek> what do you need it for?
[05:36] <dgbaley27> The disc is inconvenient
[05:36] <burek> for what
[05:40] <burek> dgbaley27, inconvenient for what
[05:42] <dgbaley27> Oh, you're serious... I have a DVD, and it would be easier for me to watch if it was stored on a hard disk. Is it relevant?
[05:42] <burek> make an iso?
[05:42] <burek> just mount it and that's it..
[05:44] <dgbaley27> That's 8GB, I know this movie could be 2GB and of acceptable quality
[05:48] <uglyoldbob> Im trying to put together a system where ffmpeg captures webcam video (15fps) and sends to ffserver with a stream that can be opened on an ipad. How can I do this? I think I have to use HTML5 to do this, is this correct?
[06:04] <pinkette> how do i rip subtitles from video that has hardcoded subs?
[06:07] <burek> use ocr
[06:10] <pinkette> okay but how
[06:13] <dgbaley27> pinkette: You know that has to be terribly involved. Are you sure you need to?
[06:13] <pinkette> huh
[06:14] <dgbaley27> pinkette: performing ocr on a video to rip subtitles. It's gotta be difficult
[06:15] <pinkette> i can do it fine if the video is DVD-video/VOB
[06:22] <okhattab> hi. im trying to compile omxplayer for the raspberry pi (fedora remix). when running make ffmpeg, it fails with "arm-unknown-linux-gnueabi-gcc is unable to create an executable file. c compiler test failed." does anyone know what might be going wrong?
[06:34] <uglyoldbob> ok so the documentation is ok, but how do I go about finding a format that an ipad can open and ffserver can also produce?
[07:21] <burek> dgbaley27 its not that difficult actually
[07:21] <burek> you just define a region for scanning
[07:22] <burek> the rest is just an image processing, which you leave over night and that's it
[07:24] <uglyoldbob> so in the example config file, there is a list of formats for a stream. Is this an exhaustive list? (mpeg, mpegvideo, mp2, ogg, etc...)
[10:28] <csholmq> Is it possible to remux to MPEG4 as it is to remux to MPEG2TS?
[10:32] <Tjoppen> just try it. something like ffmpeg -i foo -acodec copy -vcodec copy out.ts
[10:33] <csholmq> I posed my question wrong. What I meant was "Does MPEG4 also have a transport stream layer?"
[10:34] <Tjoppen> you can mux avc and aac in a ts stream if that's what you're asking
[10:35] <csholmq> But isn't that MPEG2-TS?
[10:36] <Tjoppen> "maybe"
[10:36] <Tjoppen> I've certainly seen ts streams with h.264 video in them
[10:37] <Tjoppen> I say go look into the specs a bit
[10:40] <csholmq> Tjoppen: Ok, thanks
[10:44] <bennypr0fane> hello, I'm trying to rip the sound track of an .mp4 video file to mp3 using ffmpeg. I get this ouput: "Encoder (codec id 86017) not found for output stream #0.0" Does that mean I don't have mp3 codec on my system? cause sorta sounds silly to me ...with MPlayer coming ootb in Lubuntu it's gotta be present, right? I used basically this command: ffmpeg -i input.mp4 -ab 128 -ar 44100 output.mp3
[10:45] <klaxa> try adding "-c:a libmp3lame" before -ab 128
[10:57] <bennypr0fane> klaxa I get: Unrecognized option 'c:a' Failed to set value 'libmp3lame' for option 'c:a'
[10:57] <klaxa> they your version of ffmpeg might be outdated, try -acodec instead of -c:a
[11:00] <bennypr0fane> "unknown encoder libmp3lame"
[11:00] <bennypr0fane> I installe ffmpeg from the Lubuntu repo after updating it
[11:02] <bennypr0fane> mp3 should be there though: http://pastebin.com/pivKJm0R
[11:04] <bennypr0fane> here's the complete output of that command : http://pastebin.com/reg83VJA
[11:16] <nit3ch> i am trying to install ffmpeg on suse 11.0, but there is no package of packman available for 11.0
[11:18] <pinkette> what is next video codec that will replace h264/vc1 ?
[11:19] <klaxa> nit3ch: if you can't find a package, build from source i guess? that's more recent too
[11:19] <klaxa> pinkette: probably h265 HEVC
[11:19] <pinkette> what is hevc ?
[11:19] <nit3ch> ffmpeg need yasm for compliation , even that is not avaiable
[11:19] <klaxa> http://en.wikipedia.org/wiki/High_Efficiency_Video_Coding
[11:20] <pinkette> and this is totally different standard than h264?
[11:20] <klaxa> nit3ch: https://www.ffmpeg.org/download.html see ffmpeg linux builds
[11:22] <nit3ch> ya i downloaded the linux build but when i run the command "./configure" i got these error "asm not found, use --disable-yasm for a crippled build"
[11:22] <nit3ch> when i try installing yasm , the i am getting the same error which i got for ffmpeg
[11:24] <pinkette> does ffmpeg support h265 ?
[11:25] <klaxa> not yet
[11:25] <pinkette> what is taking so long
[11:25] <klaxa> lol
[11:25] <klaxa> >High Efficiency Video Coding (HEVC) is a draft video compression standard
[11:26] <klaxa> it's not even finished completely
[11:26] <klaxa> nit3ch: if you download pre-built binaries you shouldn't need to compile anything
[11:27] <nit3ch> can you please point to to the exact build for 64 bit suse klaxa
[11:27] <klaxa> nit3ch: http://dl.dropbox.com/u/24633983/ffmpeg/index.html
[11:28] <klaxa> there are 64-bit binaries in that archive
[12:00] <JEEB> <pinkette> does ffmpeg support h265 ? <- just a note, there is no "H.265"
[12:00] <JEEB> H.something is an ITU-T callsign
[12:00] <JEEB> just like HEVC is the MPEG one
[12:00] <JEEB> right now ITU-T has it under "H.HEVC"
[12:01] <JEEB> it will probably give it a number after the spec is finished
[15:56] <cosmopolitan> H
[16:39] <Francexi> Hi everyone, anyone can help em?
[16:39] <Francexi> me*
[16:39] <Francexi> i've a problem with grabbing of a video from the x11 server
[16:42] <Francexi> is someone alive?
[16:46] <durandal_1707> Francexi: what is your problem?
[16:46] <Francexi> weel
[16:46] <Francexi> well*
[16:47] <Francexi> i'm tring to do somthing like a screencast
[16:47] <Francexi> but i noticed that the video have a "wrong speed"
[16:49] <Francexi> durandal_1707: this is the command that i use ffmpeg -f alsa -i pulse -f x11grab -s 1280x720 -r 25 -i :0+0,141 -acodec libfaac -vcodec libx264 -vpre lossless_ultrafast -threads 0 /dati/video.mp4
[16:52] <Francexi> durandal_1707: the video result accelerated, probably cause it fails to capture all the frames, but i don't know how to say to ffmpeg it
[16:53] <Francexi> durandal_1707: also the sound have the correct speed, but it's cut when the video finish(so i can't only "expand" it
[16:53] <durandal_1707> Francexi: does it happens for small resolutions like 60x50 too?
[16:54] <Francexi> mmm, i never tried
[16:58] <samek_> Hi does anyone here have a clue on why the ffmpeg (latest git/ and stable) would simply hang during streaming to rtmp FMS server ? Or what should I watch for when encoding it for rtmp. My source is RTP multicast.
[16:59] <Francexi> durandal_1707 it seems not
[16:59] <durandal_1707> Francexi: so you cant do realtime encoding for that video codec
[17:00] <Francexi> durandal_1707 how to set not to do the realtime encoding?
[17:00] <durandal_1707> use faster machine or encode to some other codec and than do h264 conversion later
[17:02] <Francexi> mmm, which container support the raw video format?
[17:02] <Francexi> durandal_1707
[17:02] <durandal_1707> or lover fps if you do not care for slideshow effect
[17:03] <Francexi> i need to screencast a game, so the "slideshow" is required
[17:04] <durandal_1707> then lover it to say 1 and voila
[17:04] <Francexi> durandal_1707: i need to grab all, 'cause i'm recording a videogame
[17:06] <durandal_1707> than pick other container and use rawvideo/rawaudio and when you finish recording convert to mp4
[17:06] <durandal_1707> that will require bunch of space on faster hard disk
[17:07] <Francexi> i have it. Which container is better to use?
[17:07] <durandal_1707> depending on duration of recording
[17:08] <durandal_1707> use nut, try first with vcodec ffv1 and acodec flac
[17:09] <durandal_1707> but even that may not be enough if your machine does not have enough resource while running game
[17:09] <durandal_1707> depends on game off course
[17:12] <Francexi> durandal_1707 which codec is ffv1?
[17:13] <durandal_1707> you just replace libx264 with ffv1
[17:14] <Francexi> durandal_1707 damn coding after the recording doesn't work
[17:14] <ritslinux> Hi all, I have a Sony rdr AT-105 video hdd/dvd/recorder that can also play DivX.
[17:14] <ritslinux> I have little or no success with some files I have on my pc when I want to play them on
[17:14] <ritslinux> the recorder: it does not recognise the file/codec.
[17:14] <ritslinux> I want to covert with ffmpeg but ffmpeg has overwelming amount of settings..
[17:14] <ritslinux> I tested some conversions to make a .avi but no or little succes on the Sony.
[17:14] <ritslinux> Any help please to get a "generic" avi-file that can be played on the Sony.
[17:14] <Francexi> now i'm trying the ffv1 codec
[17:21] <Francexi> durandal_1707 the codec you suggest me seems work, thacks a lot
[17:21] <Francexi> thanks*
[17:32] <ritslinux> The only specs that Sony gives for Divx: file-extension .avi or .divx, picturesize 720x576 max.,
[17:32] <ritslinux> no composite divx-files
[17:36] <microchip_> ritslinux: if you don't mind using MEncoder -> http://divxenc.sourceforge.net/index.html -> run it as divxenc -2p -p hwp (hwp = hardware preset)
[17:50] <Francexi> durandal_1707 are you still here?
[17:52] <Francexi> durandal_1707: the video is correctly recorded using the ffv1 codec, but when i try to convert it, it gains again the problem
[17:56] <klaxa> huh?
[17:56] <klaxa> if you reencode it to h264 you drop frames or what?
[17:56] <klaxa> oh, it gets sped up?
[17:57] <klaxa> did you set the -r option?
[17:57] <klaxa> if so, drop it to keep the original framerate
[18:13] <Francexi> klaxa: excuse me i was not to the pc
[18:13] <Francexi> no, i didn't set it
[18:14] <klaxa> then pastebin all input and output you have
[18:14] <Francexi> hold
[18:20] <Francexi> klaxa http://pastebin.com/TjfVHphA
[18:21] <klaxa> if you play the file recorded with ffv1 and flac it plays fine?
[18:22] <Francexi> yep
[18:22] <Francexi> before this i tried to record coding on the fly in h264, but the result was the same
[18:22] <Francexi> (it's a screencast of a game)
[18:23] <Francexi> klaxa: ideas?
[18:23] <klaxa> not really... never had issues like that
[18:24] <klaxa> Francexi: can you upload short samples?
[18:25] <Francexi> hold, i upload on mediafire the original video in ffv1 and the one in mp4
[18:30] <Francexi> klaxa http://www.mediafire.com/?uhmvvpibmpqv4c6 the mp4
[18:30] <Francexi> the original is big, do you need it?
[18:31] <Francexi> klaxa i have to go
[18:31] <Francexi> pls send the answers to francexi at hotmail.it
[18:32] <Francexi> bye, and thanks
[18:32] <klaxa> k, doubt i'll find anything though :X
[18:32] <klaxa> bye o/
[20:25] <Oxymoron> Anyone know a channel or website to research custom mp3 player firmware? I have this cheap prototype from china, and I would like to flash my own stuff on it. Thanks!
[21:19] <alx-> hey, how can i send multiple photos from different locations to this command: cat *.jpeg | ffmpeg -f image2pipe -r 1/4 -vcodec mjpeg -i - -vcodec libx264 out.mp4
[21:22] <sacarasc> cat blah.jpeg /foo/bar/wibble.jpg etc
[21:35] <alx-> thanks!
[22:59] <dlb76> hey guys
[22:59] <dlb76> is there a place where i can dl fresh preset files like libx264-slow_firstpass.ffpreset and so on
[22:59] <dlb76> ?
[23:00] <dlb76> because i started to get this:
[23:00] <dlb76> .ffmpeg/libx264-slow_firstpass.ffpreset: Invalid option or argument: 'directpred=3
[23:00] <dlb76> ', parsed as 'directpred' = '3'
[23:00] <dlb76> and so on
[23:00] <dlb76> trying to use static builds from http://ffmpeg.gusari.org/static/
[23:01] <dlb76> but they are without presets
[23:25] <dlb76> ...
[23:25] <dlb76> also this
[23:25] <dlb76> The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it.
[23:25] <dlb76> Alternatively use the non experimental encoder 'libvo_aacenc'.
[23:25] <dlb76> what i do ?
[23:25] <dlb76> add -strict -2 or better use 'libvo_aacenc' ?
[23:25] <dlb76> i was using libfaac before
[23:28] <maister> I'm using avcodec_fill_audio_frame(). It can allocate data. How to make sure it gets freed?
[23:32] <Mavrik> maister, avcodec_free_frame should do the trick
[23:33] <maister> Mavrik, I see. I guess I have to use avcodec_alloc_frame() as well.
[23:34] <Mavrik> maister, avcodec_free_frame won't deallocate the "data" array for frame
[23:34] <Mavrik> that you have to check for yourself
[23:34] <Mavrik> if it gets freed or not
[23:35] <maister> Mavrik, but it av_freep()'s the frame itself
[23:36] <maister> So I can't have the AVFrame just on stack then :(
[23:41] <Mavrik> maister, the point is, the data structure usually isn't owned by the AVFrame itself
[23:41] <Mavrik> it depends on what you're doing
[23:41] <Mavrik> e.g. the encoders usually take ownership of the data buffer so you don't have to free it youself
[23:41] <Mavrik> (that's why avcodec_free_frame does't free it)
[23:45] <maister> Mavrik, well, ye. But I mean, say I do AVFrame frame; then set defaults with avcodec_get_frame_defaults(), and I potentially allocate some data with avcodec_fill_audio_frame(). I can't call avcodec_free_frame() because it'll try to free the stack (and it takes a double pointer anyways).
[00:00] --- Sat Nov 24 2012
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