[Ffmpeg-devel-irc] ffmpeg.log.20121212

[burek]

[00:16] <Yulth> I everyone!
[00:18] <Yulth> I've searched on google during last 2 hours without success. I only need to delete some temporary file when connection with client is lost, for example, during streaming video. How Can I do it?
[00:22] <Yulth> sorry, this is not the channel for that question. I mistaken :)
[00:23] <Yulth> I everyone!
[00:23] <Yulth> I've searched on google during last 2 hours without success. I only need to delete some temporary file when connection with client is lost, for example, during streaming video. How Can I do it? (I'm running php through fast-cgi)
[02:27] <impy_> did i tell you i love you guys? cause i do!
[06:26] <finics> Hello, is there a way to set change PMT,PAT, PCR interval in TS stream?
[07:09] <Sashmo1> hey everyone, quick question, I've been encoding live TS streams for a while now, and I think its time to smooth out my output, I can see a little freezing in the output, its not freezing, its looks almost like a frame rate issue, but its most noticeable when there are scrolls at the bottom of the screen, anyone have some ideas? http://pastebin.com/rTJZPYg5
[10:10] <mpfundstein> ddd\
[10:32] <Scheels> Is there a way to have FFMPEG reconnect after it has disconnected from a live rtmp? Like have it keep attempting to start after it terminates?
[12:00] <ofer> hi there, what is the command to rotate the output video? (by 90 degrees for example)
[12:00] <ofer> ffmpeg -i input [?how to rotate by 90 degrees?] output
[12:03] <ofer> I see the explanation here: ffmpeg.org/ffmpeg.html#transpose but I'm not sure how does the syntax go
[12:17] <saste> ofer, -vf transpose=...
[12:19] <ofer> saste: so for 90 degrees I'll write '-vf transpose=90' ? (without the 'ticks')
[12:19] <saste> ofer: no please read the docs
[12:21] <saste> -vf transpose=dir=...
[12:28] <ofer> yeah.. the syntax isn't very clear to me the way it's written there.. I don't know how would I've guessed using the '=' sign for example, sometimes commands are using spaces instead.. anyway, thanks!
[12:33] <saste> ofer, right, doc patch incoming
[12:56] <bencc> what is the command to convert avi to mp4 that can be played with html5 video tag?
[13:41] <edgy> Hi, I am new to ffmpeg, I have a network stream coming from my receiver and wants to stream it using webm, so I launched ffmserver and ffmpeg like this
[13:41] <edgy> ffmpeg -i 'http://192.168.100.2:8001/1:0:1:7DB:24E:2BE:1040000:0:0:0:' -r 25 -s 320x256 -qmax 63 -b:v 64k -b:a 56k -ar 22050 http://localhost:8090/feed1.ffm
[13:42] <Mavrik> edgy, you need to setup encoding parameters in ffserver, not ffmpeg command line
[13:42] <edgy> when I go to my browser and use localhost:8090/live.webm because I put <Stream live.webm> nothing comes out
[13:42] <Mavrik> edgy, I wrote an article on it: https://www.virag.si/2012/11/streaming-live-webm-video-with-ffmpeg/
[13:42] <edgy> Mavrik: wow! this is exactly the article I am following now
[13:43] <edgy> Mavrik: let me pastebin my files
[13:43] <Mavrik> please do :)
[13:43] <edgy> Mavrik: https://gist.github.com/4267451
[13:45] <Mavrik> edgy, you're probably missing the global_header flags
[13:45] <Mavrik> remember, the encoding options are controled by ffserver, not ffmpeg which streams
[13:45] <Mavrik> so passing parameters to ffmpeg is pointless
[13:45] <Mavrik> add these:   AVOptionAudio flags +global_header
[13:45] <Mavrik> and turn on preroll for buffering
[13:48] <Mavrik> bbl
[13:48] <edgy> Mavrik: ok let me use your exact file because I tried to copy from it what I thought relavent first ;)
[13:51] <edgy> Mavrik: well, I copied your file and now when i launched the server I got
[13:51] <edgy> streaming/ffserver.conf:13: ACL refers to invalid host or ip address '#'
[13:52] <edgy> your line #13 seems ok, doesn't it accept comments?
[13:57] <edgy> Mavrik: if I ignore the error and launch ffmpeg exactly as you did I got the streaming of the camera
[13:57] <edgy> Mavrik: but there is a big delay
[13:58] <edgy> and I have errors like: TCP connection to localhost:8090 failed: Connection refused
[14:03] <edgy> Mavrik: are my questions annoying? ;)
[14:04] <edgy> aha! bbl means be back later, ok thanks Mavrik I will try to catch you later
[14:09] <anddam> hi, can I enhance volume level of an AAC encoded file without reencoding?
[14:10] <Mavrik> anddam, not really
[14:10] <Mavrik> there's not "volume" flag
[14:11] <Mavrik> *no
[14:13] <anddam> there's an audio filters called volumen tho'
[14:15] <anddam> but I'm not sure what filters are in ffmpeg's lingo
[14:15] <Mavrik> yeah
[14:15] <Mavrik> but you need to reencode audio to use the filters
[14:15] <Mavrik> you can't change the actual content of audio stream in compressed form
[14:15] <anddam> this page suggest the use of a -vol flag http://ubuntuforums.org/archive/index.php/t-1153179.html
[14:15] <anddam> ok, let's reencode then
[14:16] <Mavrik> anddam, that's obsolete, in new versions you should use "volume" audio filter: http://ffmpeg.org/ffmpeg.html#volume
[14:17] <anddam> that section I read in man page, I'm not sure how to use filters
[14:18] <Mavrik> sec
[14:18] <Mavrik> anddam, "-af" switch specifies audio filter chain
[14:18] <Mavrik> so i'd be something like
[14:18] <Mavrik> ffmpeg -i input.m4a -codec:a libfdk_aac -b:a 128k -af volume=2 output.m4a
[14:18] <Mavrik> to reencode
[14:19] <Mavrik> just make sure your ffmpeg is reasonably new
[14:22] <anddam> it's 2012-12-05
[14:24] <anddam> unkown encoder libfdk_aac
[14:25] <JEEB> ffmpeg -encoders |grep "aac"
[14:25] <anddam> AAC and libfaac
[14:26] <JEEB> use libfaac then, although you should build ffmpeg with fdk-aac the next time
[14:26] <anddam> but I missed the headers so I don't know if one is just for decoding
[14:26] <anddam> I will
[14:26] <JEEB> -encoders will only show encoders
[14:27] <anddam> no, there's the legenda on top
[14:27] <anddam> with the grep I was missing it
[14:27] <JEEB> there is no such video format as aac as far as I know in ffmpeg :)
[14:27] <JEEB> also it would have said V if it was video or S if it was a subtitle encoder
[14:28] <anddam> I thought one of the field could be for encoding and another for decoding since libfaac shows "(codec aac)" while AAC doesn't
[14:28] <anddam> thanks a lot all
[14:29] <JEEB> the command you used was called '-encoders' ^^;
[14:29] <JEEB> I think it'd be a problem if it would show decoders
[14:33] <anddam> much better audio level now
[14:33] <anddam> bye
[14:34] <anddam> last thing: what's the :a syntax?
[14:34] <JEEB> audio
[14:34] <JEEB> -b is bitrate
[14:34] <anddam> I guess -b stands for bitrate
[14:34] <anddam> ok and af is self explicative as well
[14:34] <JEEB> basically "set audio bit rate to this"
[14:34] <anddam> so it'd be -codec:v for video
[14:34] <JEEB> yup
[14:35] <anddam> nice
[14:35] <anddam> bye
[14:35] <JEEB> and codec:s for subtitles
[14:35] <anddam> I'll put a note for fdk-aac in macports, since I'm using ffmpeg-devel from it
[14:41] <lentferj> I have a problem with audio streams which "appear" in the middle of the video file
[14:42] <lentferj> e.g. I have a mythtv recording that is scheduled to start 5 minutes earlier
[14:42] <lentferj> at that point in time there is only 2 mp2 streams
[14:43] <lentferj> when the real movie/recording starts, a 3rd channel e.g. ac3 5.1 is added
[14:43] <lentferj> when I transcode this file with ffmpeg I lose the ac3 5.1 stream
[14:44] <lentferj> anyway to prevent this?
[14:54] <lentferj> any way for ffmpeg to detect mid-stream audio stream additions?
[15:36] <an3k> hi everybody. I'm currently converting my dvds to iTunes compatible files and im stuck at tagging the files. I can set any tag i want except moov.udta.meta.ilst.----.name:[iTunMOVI]. I found AtomicParsley but the stock version cannot write that atoms it just can delete it. i also found a modified version but "AtomicParsley error: unsupported MPEG-4 file brand found 'M4VP'"
[15:37] <an3k> Is there any tool on windows which can write that tag? or better: is there any tool that let me tell it what atom i want to write? like bla.exe -moov.udta.meta.ilst.----.name:[iTunMOVI] "this is my text"
[15:37] <an3k> thanks
[16:02] <lake> i have an unfortunate issue where my video and audio get out of sync.
[16:02] <lake> ffmpeg -vsync 1 -async 1 -i /dev/video0 -vcodec copy -acodec copy -f matroska $FILE
[16:02] <lake> i am using a hauppauge 1212 capture device
[16:03] <lake> at first it is perfect but over time becomes out of sync.
[16:03] <lake> are there any option i am missing?
[16:05] <Mavrik> lake, I think the vsync/async options only come into effect if you're actually transcoding
[16:06] <lake> Mavrik: ok, i toggled it on to see if it would do anything, which it did not, so i believe you are correct
[17:08] <burek> <lake> ffmpeg -vsync 1 -async 1 -i /dev/video0 -vcodec copy -acodec copy -f matroska $FILE
[17:08] <burek> what is your audio input?
[17:58] <elad_g> anyone can help on a simple wav to mp3 convert?
[17:58] <elad_g> i'm getting "Invalid audio stream. Exactly one MP3 audio stream is required."
[17:59] <elad_g> since ffmpeg 1.0
[18:18] <lake> burek: my audio input is magically added, i don't know how. I am using v4l2 for my hauppauge 1212 device.
[18:19] <lake> burek: i don't have to specify my audio input because audio is added automatically from the device to the resulting .mkv
[18:24] <lake> burek: http://pastie.org/5516777
[18:26] <burek> try this ffmpeg -i /dev/video0 -c copy test.ts
[18:28] <llogan> why -vsync 1 and -async 1?
[18:28] <lake> burek: why do you recommend trying that? I am trying but will need to run it for about 30 minutes probably.
[18:28] <lake> llogan: it's not needed. i was just effing around.
[18:29] <lake> llogan: i thought it would actually do something but it's obvious to me now that is only for transcoding
[18:29] <lake> burek: would you like to see the output of the "test.ts" command?
[18:30] <burek> lake, test it first to see if it works as expected
[18:33] <lake> burek: can you explain how audio is coming through?
[18:34] <burek> because your /dev/video0 is not (just) a video device
[18:34] <burek> it's emulated mpegts node
[18:34] <burek> so you are receiving mpegts stream (container) which contains audio/video
[18:36] <lake> burek: thank you kindly!
[18:36] <lake> very interesting stuff
[18:36] <burek> Input #0, mpegts, from '/dev/video0':
[18:37] <lake> burek: so do you think specifying the audio codec was screwing it up because it was converting unecessarily?
[18:37] <burek> i think remuxing to mkv did that
[18:37] <burek> it just ignored your vsync async
[18:37] <burek> (i think)
[18:38] <burek> since you were just copying (literaly) audio/video streams
[18:38] <burek> into a new container
[18:40] <burek> you were justing changing the container for your audio/video stream (remuxing), you didn't touch the streams (no re-encoding)
[18:45] <lake> burek: unfortunately i'm still experience audio lag (mouth moves shortly before audio) near the 12 minute mark.
[18:46] <lake> the lag appears to increase slightly as time goes on
[18:49] <burek> then try
[18:49] <burek> ffplay -f mpegts /dev/video0
[18:49] <burek> (no -i option for ffplay)
[18:49] <lake> burek: i'm getting alot of this when I  playback in mplayer: [h264 @ 0xfbc940]mmco: unref short failure
[18:52] <lake> burek: that ffplay command won't play more than 8 seconds before it freezes
[18:52] <llogan> ffplay has an optional -i option.
[18:53] <dericed> curious, what does the .comp array refer to in http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavutil/pixdesc.c
[18:53] <burek> lake, that means your input is damaged or something
[18:53] <burek> so, you are remuxing a damaged input..
[18:53] <burek> try cat /dev/video0 out.ts
[18:55] <burek> dericed, maybe try in #ffmpeg-devel but please be patient
[18:55] <lake> burek: that's working, in that it continuously writes to the file and I can play it w/ mplayer.
[18:56] <lake> burek: i'm not getting any "unref short failure", either. interesting
[18:57] <burek> any asyncs?
[18:58] <lake> burek: not sure yet but i do get "unref short failure" when I scrub the video
[18:59] <burek> in other words
[19:00] <burek> test if your /dev/video0 node is producing valid multimedia streams
[19:00] <lake> burek: i'm waiting for about 12 to 15 minutes to check it out since it tends to take that long
[19:01] <burek> oh...
[19:01] <burek> :S
[19:02] <lake> burek: thanks for your input on debugging this stuff
[19:03] <burek> np :beer: :)
[19:16] <lake> burek: damn audio sync issues still
[19:16] <lake> :(
[19:18] <burek> well, i don't know how did you get mpegts on /dev/video0
[19:18] <burek> but whatever is doing that, it's creating problems for you
[19:26] <lake> burek: oh man what a bummer!
[19:43] <edgy> Hi, as Mavrik told me the encoding option of ffserver is on the ffserver.conf, then why those options passed for ffmpeg for the feed?
[19:45] <Mavrik> edgy, em, you're not supposed to pass them.
[19:46] <edgy> Mavrik:I remember you did passed some options, can you please repaste your link because I am now away from my system where I have it
[19:46] <Mavrik> https://www.virag.si/2012/11/streaming-live-webm-video-with-ffmpeg/
[19:47] <Mavrik> you just do "ffmpeg -i <input> http://localhost:<port>/feed1.ffm"
[19:47] <Mavrik> no encoding parameters - those are taken from the server
[19:48] <edgy> Mavrik: and what about the option you used like -r 25 -s 1280x800, ...?
[19:49] <Mavrik> edgy, those are BEFORE the input specification and are used to configure the input
[19:49] <Mavrik> edgy, in that example I'm grabbing from a webcam where you need to specify what resolution and framerate should the camera create
[19:49] <Mavrik> edgy, if you're taking your stream from network you're not supposed to do that :)
[19:51] <edgy> Mavrik: let me explain my scenario, please. I want to stream a channel from my receiver which gives the http network stream, so my understanding is I need to modify the capture (by lowering it's size and such) and then send it to a remote ffserver where people can access it
[19:52] <edgy> Mavrik: I cannot allow people to access my local server for bandwidth issues
[19:52] <Mavrik> yeah
[19:52] <Mavrik> ok
[19:52] <edgy> Mavrik: the stream is mpeg ts
[19:52] <lake> burek: i can change the output from the capture device from ac3 to aac. maybe that will produce a different result.
[19:52] <Mavrik> edgy, you need to set the transcoding parameters in the ffserver configuration file
[19:53] <edgy> Mavrik: so can I just modify it and send it to the remote ffserver?
[19:53] <Mavrik> edgy, and when you run the "ffmpeg" targeting your ffserver it will READ those parameters
[19:53] <Mavrik> and start to transcode with your size, codec, etc.
[19:53] <edgy> Mavrik: but the stream is high quality and it will not reach the remote ffserver
[19:53] <Mavrik> (it's not logical, but that's how it works)
[19:53] <Mavrik> edgy, it will be transcoded locally
[19:54] <Mavrik> edgy, just parameters will be READ from ffserver not from command line
[19:54] <Mavrik> it will ignore what you set in command line and take those you've set on ffserver
[19:54] <edgy> Mavrik: shall I have two ffservers? one local and one remote?
[19:54] <Mavrik> *sign*
[19:54] <Mavrik> no.
[19:54] <edgy> Mavrik: deeply sorry
[19:55] <Mavrik> [your receiver] <---> [box with ffmpeg which will transcode] <---> [box with ffserver which will stream to clients]
[19:55] <edgy> Mavrik: yes exactly
[19:55] <Mavrik> the transcoding parameters are set in the ffserver.conf
[19:55] <Mavrik> but the actuall transcoding will be done by ffmpeg on the box connected to your receiver
[19:55] <Mavrik> and the stream will be sent already compressed to the box with ffserver
[19:56] <Mavrik> when ffmpeg will connect to ffserver it'll check the parameters set on ffserver and use that as it's transcoding settings
[19:56] <Mavrik> so all you have to do is set everything in ffserver.conf on the third box (the one that'll stream to clients)
[19:56] <Mavrik> and then start ffmpeg with "ffmpeg -i <your http input> http://<your ffserver>/feed1.ffm" on the box connected to the receiver
[19:56] <edgy> Mavrik: ok I will try this after one hour when I am back, really really appreciate your explanation
[19:57] <an3k> isn't there any good mp4 tagger tool?
[19:58] <Mavrik> mp4box?
[19:58] <edgy> Mavrik: though it's not optimal for me because the remote box is the one with strong CPU and I would prefer to do the encoding there but I guess you are saying this is not possible
[19:59] <Mavrik> edgy, it is, but you'll need a pair of ffmpegs
[19:59] <an3k> Mavrik mp4box can't set the itunes specific atom tags
[19:59] <Mavrik> one on the box connected to the receiver (which will just pass streams to the ffserver box)
[19:59] <Mavrik> and another on the ffserver box to do the actual transcoding
[20:01] <an3k> Mavrik currently im trying to work with AtomicParsley but that tool sucks too. it can remove those tags but not add them (how stupid is that!?!). additionally AP doesn't like M4VP files which are identical to MP4, M4V, etc.
[20:02] <edgy> Mavrik: and can the one connected to the receiver pass the stream without transcoding but with lowering size and quality? or do you call this transcoding after all and still takes CPU?
[20:04] <edgy> Mavrik: I will go now and connect after half an hour from home, hope I can catch you if things didn't go smooth ;)
[20:13] <llogan> an3k: what are the tags, exactly?
[20:13] <an3k> llogan: moov.udta.meta.ilst.----.name:[iTunMOVI]
[20:15] <an3k> Mp3Tag supports it but writes it in UPPERCASE and that's wrong. I also found a version of AtomicParsley which supports writing that tag (correctly) but it doesn't support M4VP.
[20:19] <an3k> llogan: that tag contains a XML structure which contains infos about cast, producer, screenwriters, etc.
[20:20] <an3k> QuickTime does not use that tag (thus it cannot read/write it) and iTunes only reads it but no writing.
[20:21] <llogan> how utterly useful
[20:22] <dakta> howdy llogan
[20:22] <an3k> i would love to use MKV because of MKVToolNix but thats not supported on iOS and since i can play mp4 everywhere i want and have to use this
[20:22] <lake> burek: do you think lowering the audio bitrate would help?
[20:22] <llogan> dakta: hello
[20:26] <llogan> i wonder what side splicing tape goes on (Hi8).
[20:27] <llogan> an3k: edit the source code of mp3tag and send the patch upstream
[20:32] <an3k> llogan: there is no source available but for AtomicParsley ... just need to get svn (uuh) working here so i can download and change the code. would you compile it for me?
[20:33] <an3k> just need to add something like "case 0x6D703431 : //'m4vp' to get it working. dunno where to get the correct hex from..
[20:35] <llogan> ah, mp3tag is typical for that OS. what's your distro?
[20:35] <llogan> i guess i should assume you're using windows then
[20:35] <an3k> windows 7, yes
[20:36] <llogan> i've never compiled anything in/fo windows
[20:36] <llogan> *for
[20:37] <an3k> damn but thanks. compiling on linux is easy ;) already did so often
[20:38] <an3k> i could work with the current tools if i would know how to mux a file to M4V. Im using MeGUI which uses mp4box but none of the "Device Type" settings creates a M4V. i just get stuff like M4VP or ISO or such
[20:38] <an3k> guess creating a supported container is easier than fixing code and compiling it :)
[20:40] <dakta> llogan: Are you familiar with OSX's method of software dependency sandboxing?
[20:40] <llogan> no
[20:41] <an3k> MeGUI uses this: C:\Program Files (x86)\MeGUI\tools\mp4box\mp4box.exe" -add "D:\some.264#trackID=1:fps=23.976" -add "D:\some.m4a#trackID=1:lang=ger" -tmp "D:\temp" -new "D:\output.mp4
[20:46] <dakta> llogan: It makes a lot of things much simpler in terms of software maintenance. Software is packaged in a directory, which contains all binaries and dependencies. Entirely self-contained. When the software is run, it uses its own internal dependencies. This avoids all problems with incompatible dependencies as found in package-managed systems.
[20:47] <dakta> llogan: Sorry, I'm whining about package management because I can't get the package I want and installation from sources isn't working.
[20:47] <an3k> dakta: i love that too!
[20:47] <dakta> an3k: A sympathizer!
[20:48] <dakta> an3k: After doing software developmen and maintenance on OSX, package management always feels insane. "You want me to have every single piece of software always up to date on the cutting edge, because otherwise there will be dependency version conflicts?"
[20:49] <dakta> an3k: I mean, sure, sometimes packaging all the dependencies with the software makes for a huge file, but you don't ever have to worry about breaking it by updating something else on your system.
[20:50] <an3k> thats why i never touch a running system, even if its 2 years old. well, except windows but thats never running. its always just one step before crashing :D
[20:50] <dakta> hah!
[20:52] <dakta> llogan: So, maybe you can help me with my package management nightmare. All I want to do is get completely fresh ffmpeg + dependencies from the official Debian squeeze backports package system, which I've found exists for powerpc using their web interface, but every time I try to `aptitude install ffmpeg` it somehow installs a source version I tried using.
[20:52] <dakta> llogan: To be clear, I'd just like to get a clean install of this: http://packages.debian.org/squeeze-backports/ffmpeg including dependencies.
[20:52] <an3k> llogan: found the mp4box option. its called -ipod. that creates a CodecID: M4V and that is exactly what i need. now i have to demux and remux all files ... woohoo
[20:53] <an3k> dakta: why is building from source not working? or don't you want to compile yourself?
[20:55] <dakta> an3k: I suppose I could, but I really would rather not have shit from source on there, you know? Mixing source builds and packaged stuff is not my favorite thing, especially on a system I really don't want to have to maintain.
[20:56] <dakta> an3k: I did build ffmpeg from source, barely... barely enough disk space on the system partition to build and install it. damn consumer electronics with semi-embedded *nixes
[20:57] <an3k> dakta: i understand but then you really should build from source :) i did so with php and it works like a charm. i can update the whole system and it does not break php just because every depency is in the source directory.
[20:57] <dakta> an3k: Don't talk to me about PHP or, I have a CentOS server with about fifteen versions all working happily together.
[20:57] <dakta> * or python
[20:57] <dakta> :P
[20:58] <an3k> dakta: niec :) have you used a package manager?
[20:58] <dakta> So, maybe what I need to go is go through the entire filesystem and remove every single libav* file manually...
[20:59] <dakta> Or, get the .debs for everything, and `dpkg -i --force-overwrite` on all of them in the correct order.
[20:59] <an3k> i would chose the latter
[21:00] <dakta> yeah...
[21:00] <dakta> less chance of breaking other things that way
[21:00] <an3k> indeed, and easier to accomplish
[21:01] <dakta> http://pastebin.com/3JKk569r
[21:02] <an3k> hmm...skynet ... do you have a website with skynet in its URL?
[21:04] <dakta> an3k: Nope, this is a WD My Book Live NAS. I decided "MyBookLive" was a boring name, so I renamed it "Skynet".
[21:04] <an3k> hehe :)
[21:04] <dakta> full search results paste: http://pastebin.com/K4Erxweh
[21:04] <dakta> So, I think I need to get the .debs for almost all of those
[21:13] <an3k> llogan: works now. the good thing with AtomicParsley is: it does not change anything except the changes you want. other tools like to change the bitrate tag in the container
[21:13] <dakta> an3k: I would like to know why, even though I added it to my sources.list, I can't seem to get packages from http://packages.debian.org/squeeze/powerpc/
[21:14] <an3k> download them manually
[21:14] <dakta> an3k: Dude, that's what I'm doing.
[21:15] <an3k> what package do you need for exmaple?
[21:16] <llogan> dakta: apt-get install should not install anything you compiled from source
[21:17] <an3k> llogan: dpkg does :)
[21:17] <llogan> you need to enable the backport repository in /etc/apt/sources.list, run apt-get update, and then apt-get install ffmpeg
[21:17] <llogan> you said apt-get, not dpkg
[21:17] <dakta> llogan: I understand what's happening. It's reporting the uninstallation even though it's not actually unistalling, because I compiled from source and overwrote the package install.
[21:18] <llogan> but i didn't read all of your wall of text
[21:18] <an3k> dakta: tried --purge ?
[21:18] <dakta> llogan: Not running apt-get update was my problem. thanks
[21:18] <an3k> ssshhhhh dakta :)
[21:19] <dakta> Is there any way to set dpkg force from apt-get?
[21:20] <dakta> nevermind, I'll ask that again if I need it.
[21:20] <an3k> dakta: with "manual download" i mean "wget http://ftp.us.debian.org/debian/pool/main/f/ffmpeg/libavformat52_0.5.9-1_powerpc.deb" :)
[21:20] <llogan> dakta: i don't understand the question
[21:20] <dakta> an3k: That's what I was doing.
[21:20] <an3k> dakta: works very well for me
[21:20] <an3k> dakta: but wget has nothing to do with the sources.list or apt-get or dpkg :)
[21:21] <llogan> that's a messy way of doing things
[21:21] <dakta> an3k: I know. Wget was working, but forget that if I can use apt-get instead :)
[21:21] <an3k> hehe
[21:21] <dakta> and fuck it didn't work again...
[21:22] <dakta> apt-get, why do you tell me you're installing and then not even error when there's already a file there?
[21:22] <dakta> http://pastebin.com/bfN3Bgxt
[21:23] <dakta> that is not correct
[21:23] <dakta> ffmpeg version should not be from git...
[21:23] <edgy> Mavrik: hi
[21:24] <edgy> Mavrik: I am now testing the local installation first and it works but some errors are there: http://pastebin.ca/2291978
[21:25] <edgy> any one has idea what's errors like "mpeg_decode_postinit() failure", "[tcp @ 0xed30e0] TCP connection to localhost:8090 failed: Connection refused"
[21:25] <edgy> [mpeg2video @ 0xc9bfa0] ac-tex damaged at 9 10
[21:25] <edgy> [mpeg2video @ 0xc9bfa0] Warning MVs not available
[21:25] <edgy> [mp2 @ 0xc9d9a0] incomplete frame
[21:25] <edgy> Error while decoding stream #0:1: Invalid data found when processing input
[21:25] <edgy> and the stream stopped after a while
[21:26] <llogan> dakta: your compiled ffmpeg binary is probably in /usr/local/bin, and the repo, if installed, is in /usr/bin
[21:26] <dakta> llogan: Yep, it was
[21:27] <dakta> llogan: I just figured that out right as you were typing that response. I appreciate the help, though.
[21:27] <dakta> And lo and behold, ffmpeg works.
[21:27] <dakta> It's a miracle.
[21:28] <dakta> llogan: And the best part? This new version of ffmpeg correctly reads the iTunes M4A "gnre" atom!
[21:29] <dakta> llogan, an3k: Thanks for your help, I really appreciate it.
[21:29] <llogan> you mean the backported version or your compiled one?
[21:29] <dakta> llogan: I removed the compiled one, which was not working, and the backport package works.
[21:31] <dakta> Now let's see if forked-daapd still works. forked-daapd is an iTuned media server implementation. I just told it to rebuild its library database, which means it's getting song info using ffmpeg on around 200GB of song files.
[21:31] <dakta> This may take a little while. :)
[21:32] <an3k> many years ago i thought with upcoming mpeg4 that you only need one stream and devices could just drop everything they can't support and play/stream the file. then i noticed that this isn't possible and you have to encode specially for every device. but i think this is coming with mpeg6, isn't it? :)
[21:34] <llogan> dakta: the backported version is not using source from FFmpeg, but from a fork
[21:35] <dakta> llogan: Aaand? The important thing for me is that it appears to be working.
[21:37] <llogan> we can not support that here
[21:38] <llogan> it would be nice to be able to duplicate the issue with current ffmpeg so it can be fixed
[21:40] <dakta> llogan: The bug has already been fixed in ffmpeg, as far as I am aware. I just needed to get an updated version that contained the bugfix.
[21:52] <dakta> llogan: for the record, it's this bug: https://github.com/jasonmc/forked-daapd/issues/76 I haven't been able to find the ffmpeg patch or bug report, otherwise I'd link those.
[21:54] <an3k> Windows: I HATE YOU: "<" kann syntaktisch an dieser Stelle nicht verarbeitet werden.
[22:11] <bencc> how can I convert avi to webm that can be played in web browsers?
[22:15] <nikil> is there a way i can demux audio from a file using ffmpeg and have output to the native file format of the audio stream?
[22:16] <Mavrik> nikil, container format is always chosen via file extension or "-f" switch so no
[22:16] <nikil> ah, bummer.  i'm running ffmpeg in a script that is processing lots of different files, so this means i have to figure out what the container is for every audio stream :(
[22:23] <llogan> bencc: ffmpeg -i input -c:v libvpx -c:a libvorbis output.webm
[22:24] <llogan> would be the most basic using probably crappy default settings
[22:27] <llogan> libvorbis default is good now (VBR, quality "3")
[22:53] <bencc> llogan: thanks
[22:54] <bencc> llogan: Unrecognized option 'c:v'
[22:54] <bencc> Failed to set value 'libvpx' for option 'c:v'
[22:54] <bencc> under ubuntu
[22:55] <ubitux> the ffmpeg distributed in ubuntu is an old non-maintained version from a fork of the project
[22:55] <ubitux> you can use -vcodec, or use the fork
[22:55] <ubitux> & or upgrade to ffmpeg
[22:57] <bencc> ubitux: how can I upgrade to ffmpeg
[22:57] <bencc> ?
[22:57] <bencc> use -vcodec where?
[22:57] <ubitux> -vcodec is the old -c:v
[22:58] <ubitux> this can help as well.
[22:58] <bencc> and c:a? -acodec?
[22:58] <ubitux> yes
[22:58] <bencc> now something happens :)
[22:58] <bencc> will it put the seeking index at the beginning of the file?
[22:59] <bencc> so I can seek before loading all the file
[23:01] <ubitux> afaik mkv isn't designed like mp4 and thus has not that issue
[23:02] <bencc> ubitux: I tried with milo on windows and wasn't able to seek while seeking works for other files
[23:03] <ubitux> i don't know milo, try with an up-to-date ffmpeg
[23:03] <ubitux> if it doesn't work, it might be a bug with your player
[23:03] <ubitux> if you believe it's not, please open an issue on the bug tracker
[23:04] <bencc> ok. thanks
[23:10] <llogan> bencc: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[23:10] <llogan> if you feel like compiling
[23:18] <llogan> mkclean claims to "reorder the elements with the Cues at the front, so your Matroska files are ready to be streamed efficiently over the web"
[00:00] --- Thu Dec 13 2012