[Ffmpeg-devel-irc] ffmpeg.log.20120709

[burek]

[00:26] <linkmauve_> Ok, it was just an issue with the tmpfile function in the configure shell script. Im going to sleep, but Ill give a patch tomorrow to fix that.
[01:21] <YellowOnion> I encoded a video, but messed up the audio, how can I remux/recode audio and make it so it's not out of sync without re encoding the video as that takes too long
[01:25] <burek> make your audio proper in audio.wav and then
[01:26] <burek> ffmpeg -i audio.wav -i video.mp4 -map 0:1 -map 1:1 -vcodec copy -acodec ... out.mp4
[01:26] <burek> you should of course modify your -map according to real situation in your files
[01:27] <burek> or if you don't want to mess with -map, first extract your video only to a file: ffmpeg -i video.mp4 -an -vcodec copy video2.mp4
[01:27] <burek> and then just ffmpeg -i video2.mp4 -i audio.wav -vcodec copy -acodec ... out.mp4
[01:28] <YellowOnion> that looks like what I tried and the audio went out of sync
[01:28] <burek> use -itsoffset to move video left/right and sync it
[01:29] <YellowOnion> burek, is there no way to automatically copy the offset over?
[01:29] <burek> what do you mean
[01:30] <YellowOnion> the original file must have an offset
[01:30] <burek> that offset is not written anywhere in the file
[01:30] <burek> it's just a subjective notification there is one
[01:31] <burek> ffmpeg -i audio.wav -itsoffset 5 -i video.mp4 -vcodec copy -acodec ... out.mp4
[01:34] <YellowOnion> I'm not sure what you mean the original file plays fine, and the non-proper has in-sync audio, but the propered file has out of sync audio as the sync information was lost during remuxing
[01:35] <burek> what is "propered file"
[01:35] <YellowOnion> I forgot to down the audio
[01:35] <YellowOnion> downmix*
[01:37] <YellowOnion> propered file is the one with proper audio
[01:37] <burek> well, extract the proper video and proper audio
[01:37] <burek> and use the last command I gave you
[01:37] <burek> if you have a proper audio already encoded, then avoid -acodec ... but use -acodec copy instead
[01:38] <burek> the example I gave you was just a showcase of the proper ffmpeg usage for your problem
[01:41] <YellowOnion> burek, the audio goes out of sync if I encode the file sperately
[01:52] <YellowOnion> burek, here's an example: http://www.youtube.com/watch?v=2fh8syZBLgM&t=5m18s notice how it laggs and then the audio is out, the original file doesn't have this problem
[01:55] <burek> that is the original?
[01:58] <YellowOnion> mkv with x264 and flac audio
[02:00] <YellowOnion> ffmpeg seems to have added resync codes or something to the file, but if I remove the audio and mux in my own it goes out of sync
[02:01] <burek> well it seems to me your original is damaged.. so..
[02:02] <burek> next time try using mp4 format or flv instead of mkv
[02:02] <burek> something like this for example: ffmpeg -f x11grab -r 25 -s 1024x768 -i :0.0+100,200 -f alsa -ac 2 -i pulse -vcodec libx264 -crf 0 -preset ultrafast -acodec pcm_s16le output.flv
[02:03] <burek> to grab your screen losslessly
[02:03] <burek> and save cpu
[02:03] <burek> after you grab it, then play with it and encode it as you wish
[02:03] <YellowOnion> burek, the mkv holds the information fine, it's the audio when removed and remuxed sperately that causes it to forget the information
[02:04] <burek> well, I just saw it on youtube and it's not fine
[02:05] <YellowOnion> burek I said it lost the information after remuxing, the original mkv is fine the youtube one isn't the original fine
[02:06] <burek> well then try ffmpeg -i original.mkv -vcodec copy -acodec ... output.mp4
[02:06] <YellowOnion> obviously I'm not going to upload a file with lossless video and audio to youtube that would take days on my connection
[02:06] <burek> oh
[02:06] <burek> well then try ffmpeg -i original.mkv -vcodec libx264 -acodec aac -strict experimental -ab 128k -ar 44100 -ac 2 output.mp4
[02:07] <YellowOnion> but what if I messed up the audio? and want to proper it? without reencoding the video?
[02:08] <burek> I don't understand your question
[02:08] <burek> you have "lossless video and audio"
[02:08] <burek> and you don't want to reencode the video?
[02:10] <YellowOnion> I re encoded the video but the audio is messed up for example I forgot to change the codec bitrate etc, now I have the correct video bitrate and codec, but not the correct audio codec, how would I fix the audio without waiting for the video to re encode again
[02:11] <burek> man, I answered that question an hour ago
[02:11] <burek> just use -vcodec copy
[02:11] <YellowOnion> yes but it caused the problem in my youtube video
[02:12] <YellowOnion> some information is discarded when I dump the audio track
[02:13] <burek> then your video is damaged
[02:13] <burek> if -vcodec copy doesn't give out a proper video stream
[02:13] <burek> simply your original is damaged
[02:13] <burek> and you need to do it all over
[02:15] <burek> most probably you had a very dynamic scene in the game so the cpu got starved for a second, and that's the lag you see
[02:15] <burek> start ffmpeg with higher priority (use "nice" or "renice")
[02:15] <burek> in order to keep-up with the high cpu usage in the game
[02:15] <YellowOnion> I like how you assume I'm using Linux...
[02:15] <burek> change the priority in task manager?
[02:18] <YellowOnion> Yes I know how to do that, the problem is somethings only happen once I can't reshoot everything, and anyway the original file plays fine (baring the frame drops), but extracting the audio causes it to lose the sync information and only the youtube version goes out of sync the original doesn't so some information is being lost
[02:19] <burek> try another encoder then
[02:20] <burek> encoding tool*
[02:20] <YellowOnion> I could do a complete re encode but at 7fps it'll take forever specially if I was to record longer videos
[02:20] <burek> well, if your video was properly encoded, it doesn't matter what you do with audio
[02:21] <burek> btw, what happens at that point in time, when you watch the original video
[02:21] <burek> does it have any quircks or something?
[02:21] <burek> quirks*
[02:22] <YellowOnion> visually looks the same but the audio seems to wait and corrects it's self
[02:22] <burek> so, it's damaged
[02:22] <wishie> so I'm trying to compile ffmpeg with libfaac support.. for that i need to have faad installed previously, correct?
[02:22] <burek> wishie, usually you would have to get libfaaC not libfaaD
[02:23] <wishie> i thought that was strange, but the main source forge page seems to have a faad download button.. hmm, checking again
[02:23] <YellowOnion> burek, yes, but there's enough information to make it barible.
[02:24] <wishie> http://sourceforge.net/projects/faac/
[02:24] <wishie> ok, so i have to click "files"
[02:24] <burek> YellowOnion, well, ffmpeg is great, but sometimes even ffmpeg can't correct things
[02:24] <burek> I'm sorry but your input is just damaged
[02:24] <wishie> then the faac_src dir.. why do they have faaD on the main page?
[02:24] <burek> wishie, I really wouldn't know, I don't use it at all :)
[02:25] <burek> either I use internal ffmpeg's aac encoder or libaacplus :)
[02:26] <wishie> i need gpac with lavf support, so ffmpeg, which i thought needed libfaac
[02:26] <wishie> how can i have it use the internal encoder?
[02:27] <burek> just use -acodec aac -stric experimental
[02:27] <burek> strict*
[02:28] <YellowOnion> burek, I don't want it fixed, I said the original keeps the audio in sync, but if I extract/seperate the audio it then loses sync when I remux them.
[02:28] <burek> YellowOnion, ok
[02:30] <YellowOnion> wishie, do you need aac? I haven't heard many good things about faac's quality, lame is usually better at high bitrates
[02:31] <burek> aac+ is way better than mp3 otoh
[02:31] <wishie> well, i basically need GPAC with lava support (from ffmpeg) so i can compile x264 with MP4 output support
[02:31] <wishie> and ultimately, the output files need to contain AAC audio for iDevices
[02:32] <wishie> and i have to use x264 binary, so i can specify --frame-packing 3 (for Side-by-Side 3D)
[02:32] <wishie> err
[02:32] <wishie> lava = lavf
[02:36] <YellowOnion> burek, I said lame and faac for a reason, aac+ and mp3 aren't what I'm talking about, if you need an aac encoded, use nero or itunes, forget every other encoder
[02:38] <YellowOnion> wishie, hmm, I would be curious why Apple wouldn't allow mp3 to play in a .mp4 container
[02:38] <wishie> not sure if they do or not
[02:39] <burek> I use neroaacenc too, beside libaacplus when needed
[02:39] <burek> btw, what is lame if not mp3?
[02:41] <YellowOnion> because you said aac+ is better than mp3, but for example libvo_aacenc is utter crap compared to lame
[02:42] <burek> libvo_aacenc cannot encode acc+ afaik
[02:42] <burek> only AAC-LC
[02:42] <burek> wich is simply AAC
[02:42] <burek> I'm talking about aac+ (HE-AAC v2)
[02:42] <burek> which is superb extension of basic AAC
[02:42] <YellowOnion> anyway that is not the point, the point is that the codec is only as good as the encoder
[02:43] <burek> well, if wikipedia says that aac+ is far better than mp3
[02:43] <burek> then so be it
[02:43] <wishie> man, trying to get x264 compiled with lava support seems near impossible :P
[02:44] <wishie> all so x264 will support input formats other than raw
[02:44] <burek> lava?
[02:44] <wishie> lavf
[02:44] <wishie> damn brain
[02:44] <burek> this? "--disable-lavf           disable libavformat support"
[02:45] <wishie> i need lavf support
[02:45] <burek> well, just compile ffmpeg prior to compiling x264
[02:45] <wishie> oh wait, its this damn irc client doing spell check
[02:45] <burek> that's it
[02:45] <wishie> I've done that.
[02:45] <burek> so you have x264 with lavf support
[02:45] <wishie> nope
[02:46] <burek> then you did something wrong :D
[02:46] <wishie> x264 config script says it can't find lavf support
[02:46] <burek> did you type ldconfig after make install of ffmpeg
[02:46] <YellowOnion> burek, I don't see what you mean faac is worse than lame, sure aac+ has more potential, but if you're using faac you're throwing it out the window
[02:46] <burek> YellowOnion, I'm telling you for the 3rd time.. aac+ != aac
[02:46] <burek> aac+ = aac + sbr + ps
[02:47] <burek> "Data from this testing also indicated that some individuals confused 48 kbit/s encoded material with an uncompressed original."
[02:47] <burek> mp3 couldn't have this compliment not even with 320 kbps encoding
[02:48] <burek> btw, I re-encoded all my mp3s into aac+ without any noticeable change in quality, but the output files were 10x smaller!
[02:50] <wishie> rebuilding ffmpeg again to be sure
[02:50] <burek> no need
[02:50] <burek> just ldconfig so that x264 can see it properly
[02:50] <wishie> tried that :)
[02:51] <burek> did you --enable-shared --enable-static
[02:51] <YellowOnion> burek, I said faac, you're the one that bought up aac+, and still aac+ is a format not an encoder, it's the encoder than makes the codec work corrently, if you have a bad encoder it'll sound crap, faac(yes I know not aac+) is worse than lame and libvo_aac a terrible terrible encoder
[02:51] <burek> any of that?
[02:51] <wishie> i think i have --enable-shared
[02:51] <burek> ok
[02:51] <burek> "aac+ is a format not an encoder" wtf YellowOnion ?
[02:51] <wishie> i didn't use static though
[02:52] <burek> wishie, no need if you are not building x264 statically
[02:52] <burek> YellowOnion, what are you talking about?
[02:52] <YellowOnion> burek, aac+ is a format for storing bits of data to makes sound, it doesn't create the data to store in the file, it's up to the encoder
[02:53] <wishie> GRR!
[02:53] <wishie> lavf: no
[02:53] <burek> YellowOnion, lol
[02:54] <burek> you don't know what you are talking about man
[02:54] <burek> if that was true, there would be no need for libaacplus encoding library
[02:54] <burek> wishie, check the config.log of x264
[02:54] <wishie> hmm ok, so the x264 config script is..
[02:54] <burek> find where does it check for lavf
[02:54] <wishie> yeah, I'm checking that as we speak.. lol
[02:55] <burek> :)
[02:55] <YellowOnion> burek, lol man you're funny, you transcoded mp3s to aac and tell me that I'm stupid...
[02:55] <burek> you told yourself that you are stupid
[02:55] <burek> I didn't mention it ever
[02:56] <burek> just read the wikipedia and see what it is
[02:56] <wishie> http://pastebin.ca/2168922
[02:56] <burek> wishie, did you git clone both ffmpeg and x264
[03:02] <YellowOnion> burek, mp3 use to sound like shit before LAME came along, and for the same reasons faac sounds worse than lame, the scene has banned faac from their releases, and hydrogen audio don't even bother testing the quality in abx because it's obviously infeiroir to nero and itunes
[03:03] <burek> geez
[03:03] <burek> ok you're right :)))
[03:04] <burek> just let's drop it already, ok? :)
[03:13] <wishie> ffmpeg is from git
[03:13] <wishie> x264 is the latest snapshot, too
[03:13] <wishie> from yesterday
[03:14] <burek> so, both are latest
[03:15] <wishie> checking x264 now
[03:15] <wishie> doing a fresh git clone
[03:15] <wishie> as opposed to the snapshot from yesterday
[03:15] <burek> well, shouldn't make a big difference
[03:16] <wishie> http://pastebin.ca/2168928
[03:18] <burek> can you try to git pull some older version of ffmpeg, like from month ago or something
[03:18] <burek> because some major changes have occurred recently
[03:18] <burek> and it might affect x264 linkage
[03:18] <wishie> i'll give it a go
[03:19] <burek> <durandal_1707> burek: x264 compiles just fine here (lavf enabled)
[03:19] <burek> wishie, :)
[03:19] <burek> you did something wrong :)
[03:20] <wishie> hmm
[03:20] <burek> you might join #ffmpeg-devel and ask durandal_1707 what options did he use
[03:20] <burek> etc
[03:23] <heimlich> Hi Guys, I have a question.  I have a quicktime movie file that has a timecode track in it.  I am trying to figure out, how to extract a specific segment from it.  In this particular quicktime, the timecode starts at 34:33:22:15 and i want to extract 34:33:25:14 until 34:34:00:02.  I dont think i can do that with ffmbc/ffmpeg.  Is anyone aware of any other commandline tool ?  The importance...
[03:23] <heimlich> ...is that I don't always know what the starttimecode is of the quicktime, unless I were to open it and double check, which of course defeats the purpose.
[03:24] <burek> ffmpeg -i input.mov -ss 34:33:25.14 -t (calculate the output length) -c copy out.mov
[03:24] <burek> or
[03:24] <burek> ffmpeg -ss 34:33:25.14 -i input.mov -t (calculate the output length) -c copy out.mov
[03:24] <burek> which will be faster, but less accurate
[03:27] <burek> oh I see, timecodes.. :)
[03:27] <heimlich> yeah :)
[03:27] <burek> well, have fun with it :)
[03:27] <wishie> lol
[03:27] <heimlich> lol :)
[03:27] <burek> maybe -segments can help but I'm not sure
[03:28] <burek> http://ffmpeg.org/ffmpeg.html#segment_002c-stream_005fsegment_002c-ssegment
[03:29] <heimlich> the big issue with that, is that i don't know what the start timecode of the quicktime is.. just that the editor wants a specific range with starttimecode and endtimecode... but without opening the quicktime, i can't do it :(
[03:30] <burek> well, use ffprobe to parse the metadata
[03:30] <burek> where the start time code is written
[03:30] <burek> and use it in subsequent calculations
[03:30] <burek> (in a script for example)
[03:31] <heimlich> lemme see if ffprobe will give me that info
[03:32] <wishie> hmm
[03:32] <wishie> i just need a fresh ffmpeg checkout
[03:32] <wishie> (instead of using the 1 day old snapshot)
[03:32] <wishie> and now x264 says it will compile with lavf
[03:32] <burek> wishie, I wasn't aware that ffmpeg makes snapshots
[03:33] <wishie> hmm, wait, i might be confused
[03:33] <wishie> but it had 20120807 in the filename
[03:33] <wishie> which is.. yesterday (for me, anyway)
[03:33] <burek> where did you get it from
[03:34] <wishie> hmm, the first time i ran& svn export svn://svn.mplayerhq.hu/ffmpeg/trunk ffmpeg
[03:34] <heimlich> cool.. ffmpeg gives me the starttimecode
[03:34] <heimlich> sorry.. ffprobe i mean
[03:34] <wishie> ^^ didn't work
[03:34] <wishie> so i did& git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg
[03:34] <wishie> that works.
[03:35] <burek> wishie, svn is like..
[03:35] <burek> abandoned.. or something
[03:35] <burek> very long time ago
[03:36] <burek> :)
[03:37] <wishie> grr, now x264 is being silly
[03:39] <mfillpot>  I am attempting to write a script to allow screencasting through ffmpeg on slackware-13.37 and am unable to properly locate the mic and speaker output files to capture, could anyone assist me with this?
[03:41] <burek> mfillpot, did you use -i pulse
[03:42] <burek> also try reading this: http://ffmpeg.org/trac/ffmpeg/wiki/Capturing%20audio%20with%20FFmpeg%20and%20ALSA
[03:44] <mfillpot> burek, I am in slackware, so pulse is not used. However I am reading the link.. thank you for the link
[03:45] <burek> :beer: :)
[03:49] <mfillpot> burek, that wiki did prove that my attempt to capture the microphone was correct, however it does not reference capturing the system/speaker output. I have been searching all day to try to find a solution that works. Do you know where I can find information on that?
[03:51] <burek> mfillpot, that is kinda specific
[03:51] <burek> use alsamixer to see if your audio card
[03:51] <burek> supports recording from an output device (like stereo output)
[03:52] <burek> also, I've read somewhere that if your audio card doesn't support that kind of thing, that you can still use "jack" audio to create a virtual device, from which you will record
[03:52] <burek> but I read it a long time ago, so I don't have any reference links :/
[03:52] <mfillpot> I will look into what alsamixer has available, I have been looking for a /dev/ file so I may have been misguided. FYI, if you are interested I am working on the script at http://pastebin.com/m2yA94CC
[03:54] <burek> mfillpot, it might be good that you read this too: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20(screen)%20with%20FFmpeg
[03:55] <burek> usually people grab their game play, so encoding during the game play will require some decent hardware
[03:55] <burek> so, you might consider grabbing it losslessly first
[03:55] <burek> and encoding after the grabbing is finished
[03:58] <mfillpot> I will look into that option, however this will not be for anything too intensive, mostly just following desktop navigation
[04:03] <wishie> grr, now gpac won't compile against the ffmpeg/libavcodec i built
[04:18] <mfillpot> I know that my output can be captured, so I just can't find the input file/device. I know that when it is found it will seem obvious.
[04:44] <heimlich> burek: thanks for the tip with -ss... so far, on the few quicktimes i tried.. works perfect
[05:06] <wishie> i am now getting the following when trying to compile ffmpeg
[05:07] <wishie>  /usr/bin/ld: libavutil/intfloat_readwrite.o: relocation R_X86_64_PC32 against undefined symbol `frexp@@GLIBC_2.2.5' can not be used when making a shared object; recompile with -fPIC
[05:18] <JoeHazzers> is there anyway to demux all streams from a file/container, and automatically give appropriate file names? i just want to extract all the streams and dump them somewhere, but i don't want to specify individual filenames for each stream. can i do this?
[06:20] <heimlich> anyone here ever converted a .mov to a dpx sequence ?
[07:09] <zap0> no.
[07:22] <heimlich> getting slight color shift between the quicktime and the resulting dpx frames :(
[07:33] <Sashmo> hey everyone, can anyone tell me why when using -f mpegts option that it adds a default service name and service provider "ffmpeg"?  Is there a way to remove it with out haveing to add it into the command line?
[08:14] <diegoviola> hi
[08:15] <diegoviola> is there a way for ffmpeg to tell me the codec information of a file/
[08:15] <diegoviola> ?
[08:15] <diegoviola> codec info, res, etc
[08:18] <spaam> yes
[08:18] <spaam> ffmpeg -i file
[08:18] <spaam> or ffprobe file
[08:19] <diegoviola> ty
[08:19] <spaam> you can about it as json and other formats with ffprobe.
[08:20] <spaam> http://ffmpeg.org/ffprobe.html
[08:20] <diegoviola> very nice, thanks a lot
[11:57] <pespin> hi, I'm trying to use libffmpeg in my C app to encode h264, however the output is aleways in mainline profile, despite I set profile=FF_PROFILE_H264_BASELINE in my code :S
[11:57] <pespin> does someone have any hint on what am I doing wrong? May I be missing something?
[12:03] <burek> what is libffmpeg?
[12:05] <pespin> burek, well I meant the C api for ffmpeg sorry :)
[12:06] <burek> you mean libav libs? :)
[12:06] <burek> well, h264 is a decoder in ffmpeg
[12:07] <burek> and for encoding, libx264 is used
[12:08] <pespin> burek, yep, I'm using libx264
[12:08] <pespin> but I only get videos with mainline profile :S
[12:10] <burek> take a look how does ffmpeg do it properly
[12:10] <burek> and copy the logic
[12:11] <pespin> burek, you mean the ffmpeg app cli?
[12:49] <Darkyen> Where can i find various error codes in ffmpeg ?
[12:49] <Darkyen> avcodec_audio_encode2(); returns error code -22 i wonder whats that supposed to mean ?
[12:51] <Darkyen> nvm found it
[13:35] <bindi> C:\Python27\stream\ffmpeg -f dshow -i video="UScreenCapture" -f dshow -i audio="Stereo Mix (ASUS Xonar DG Audio"  -r 30 -c:v libx264 -b:v 2000k -acodec libmp3lame -ab 128k -f flv http://bindi.shroudbox.net:8888/stream.ffm
[13:36] <bindi> scary.. 2 clients instantly opened that link
[13:36] <bindi> http://pastebin.com/k6x2nqRS here's the output
[13:37] <bindi> Mon Jul  9 11:33:40 2012 94.22.125.174 - - [POST] "/stream.ffm HTTP/1.1" 200 4096
[13:37] <bindi> I see this on the server
[13:49] <bindi> http://bindi.shroudbox.net/ffserver.conf here's my ffserver.conf
[13:51] <bindi> woop wrong codec there
[14:03] <bindi> nobody? :<
[14:16] <burek> bindi
[14:17] <bindi> hmm?
[14:19] <bindi> burek: what am I doing wrong
[14:20] <burek> I don't know, read the manual
[14:20] <burek> I don't even understand what is your question there
[14:20] <bindi> it doesn't work, what am I doing wrong is my question :-)
[14:20] <burek> I don't know
[14:20] <bindi> the command line is fine since it works locally
[14:21] <bindi> do I have to specify the video resolution?
[14:31] <bindi> burek: look what I found http://ffmpeg.org/trac/ffmpeg/ticket/1275
[14:31] <bindi> I tried what that guy said works for him, didn't change anything
[14:33] <burek> bindi, use vlc
[14:33] <bindi> what for
[14:33] <burek> for the peace in the world...
[14:33] <burek> what for..
[14:33] <bindi> why do I need vlc here
[14:33] <burek> ok, then have fun with ffserver
[14:33] <bindi> no tell me
[14:33] <bindi> i used vlc before
[14:34] <burek> need to go, sorry
[14:34] <bindi> bullshit
[14:38] <proshot> hya everybody, i am trying to record my desktop and using ffmpeg for that, however when i use vlc to view the recording the contrast is too high
[14:38] <proshot> and the brightness also, is there a way to change the contrast and brightness of the recordings
[15:12] <Sashmo> hey everyone, can anyone tell me why when using -f mpegts option that it adds a default service name and service provider "ffmpeg"?  Is there a way to remove it with out haveing to add it into the command line?
[15:23] <rainmaker1> Anyone have experience with fragmented mp4? I am trying to make a fragmented mp4 but it can't be played in vlc, although ffmpeg does not drop any error. I use this command ffmpeg -i udp://230.11.0.3:5000  -c:v copy -c:a copy -map 0 -movflags frag_keyframe -frag_duration 60000000 -absf aac_adtstoasc out.mp4
[15:33] <burek> rainmaker1, did you try -ssegment
[15:33] <burek> -f ssegment or something
[15:36] <rainmaker1> burek: yes I tried but I need to stream this file and streaming server does not support segemnt muxer :(
[16:12] <swkide> rainmaker for http apple streaming?
[16:16] <rainmaker1> wiasw	sw	sw
[16:16] <rainmaker1> swkide: yes, for HLS and RTMP streaming
[16:19] <swkide> rainmaker1: I was told, that segmentation code is broken - I tried and found the correct parameters but the audio was clipping
[16:20] <rainmaker1> swkide: can you share a command you used to create fragmented mp4?
[16:20] <swkide> rainmaker1: sure will lokk them up - please hold
[16:22] <swkide> ffmpeg -v debug -i $input -c:v copy -c:a copy -vbsf h264_mp4toannexb \
[16:22] <swkide> 	-map 0 -f segment -segment_time 10 -segment_list out\test.m3u8 out\stream%03d.ts
[16:23] <rainmaker1> swkide: thats segment muxer, I saw audio issues also, but that was fixed in the latest version of ffmpeg
[16:23] <swkide> the m3u8 file is broken after that, but you can easily recreate it.
[16:23] <swkide> Now I remember I only had audio problems when segmenting from flv
[16:23] <function1> hi people, any sane h.264 codec inherently can decode all profiles, correct? only question is whether hardware can do the computation fast enough for smooth play?
[16:23] <rainmaker1> swkide: what I was looking is fragmented mp4 support, which means one file with multiple fragments inside that file
[16:24] <swkide> rainmaker1: sorry my fault then
[16:24] <rainmaker1> swkide: but tnx :)
[16:26] <burek> function1, what exactly is your question?
[16:31] <function1> burek: getting 720p high at L4.1 videos on my idevice. all the stuff online says just hexedit the header to whatever the max level supported by idevice is (main at l3.1 in my case). apparently this works, but i'm wondering how decoding of various profiles is different from eachother. does this trick work because the idevice hardware is actually fast enough, despite the tech specs, or does the decoder actually have support for high at l4.1? is the binary
[16:32] <function1> not really an ffmpeg question
[16:33] <burek> if I understand the problem correctly, you shouldn't be able to "just hexedit" the header
[16:33] <burek> because the header keeps the information on profiles/levels used to encode the stream
[16:34] <burek> and according to encoding techniques used (depending on level/profile), you need appropriate decoder capable of decoding such thing
[16:35] <burek> when you encode your video with the lower profile (less cpu needed) then your video was most probably encoded with less transformations
[16:35] <burek> which will require less cpu to decode it
[16:35] <burek> something like that
[16:36] <function1> right.. so why does this actually work? apple is lying and they actually wrote support for a higher profile than what is listed in the specs? or, the codec can still render the stream, though maybe it is not optimized to do so, or will do so with some distortion?
[16:36] <burek> and if it was encoded in high profile (using all the possible tricks/transformations to optimize bit rate, quality, file size, etc) then I don't know what the effect of hexeditting the header would be
[16:37] <burek> the only thing that comes to my mind is that those transformations do not depend on each other, meaning, you can apply only certain reverse transformations to get the image of your input more or less accurate
[16:37] <burek> and more reverse transformations you use, the more accurate input you get
[16:38] <burek> so probably what you did is to tell your decoder not to apply all those decoding transf. but only a subset, that the hardware can bare with
[16:39] <bindi> burek: would you like to assist me with vlc?
[16:40] <burek> bindi, did you try on #videolan?
[16:40] <function1> that seems to make sense
[16:40] <bindi> I tried mpegts and flv over udp and rtp, none of these combinations work
[16:40] <bindi> burek: sure did :p
[16:40] <bindi> thought you had some experience in this
[16:40] <burek> bindi, I guess nobody replied :)
[16:40] <bindi> indeed
[16:40] <burek> I was just looking at their forum.. man.. 90% of posts are unanswered
[16:40] <burek> and a lot of them are alike, meaning that one answer could have covered many posts
[16:41] <burek> bindi, what is your use case
[16:41] <burek> what do you need to get done
[16:41] <bindi> burek: user streams his desktop to my server and my server re-streams it for flowplayer
[16:41] <bindi> or rather, other users watching it through flowplayer
[16:41] <burek> hmh
[16:42] <burek> do you need it to be real-time-ish or it can have big delays
[16:42] <bindi> real-time-ish :p
[16:42] <burek> i was afraid you're gonna say that
[16:42] <burek> well, you need udp then at least from the originator of the stream to your vlc
[16:42] <bindi> I already had a working solution, but it went the other way around - vlc (serverside) connected to the user (needs open ports, pain in the ass) but it worked flawlessly
[16:42] <burek> and from vlc, you only need http, which flowplayer will connect to
[16:42] <bindi> yes
[16:43] <burek> so, at the originator, you can use something like: vlc screen:// --sout='#transcode{...}:std{access=udp,mux=ts,dst=ip.of.the.vlc:port}'
[16:44] <bindi> can't I use ffmpeg for that?
[16:44] <burek> or ffmpeg -f x11grab -i 0.0 -vcodec .. -f mpegts udp://ip.of.the.vlc:port
[16:44] <bindi> hm
[16:44] <bindi> already doing that
[16:44] <burek> ok
[16:44] <bindi> ffmpeg -f dshow -i video="UScreenCapture" -f dshow -i audio="Stereo Mix (ASUS Xonar DG Audio"  -r 30 -c:v libx264 -b:v 2000k -acodec libmp3lame -ab 128k -f mpegts udp://bindi.shroudbox.net:8888
[16:44] <bindi> here's my full command line
[16:45] <burek> then start vlc with: vlc -I dummy -V dummy udp://@:port --sout='#std{access=http,mux=,dst=0.0.0.0:port2/live.flv}'
[16:45] <burek> vlc -I dummy -V dummy udp://@:port --sout='#std{access=http,mux=,dst=0.0.0.0:port2/live.flv}' --sout-mux-caching=10000
[16:45] <burek> caching because of udp
[16:45] <burek> yes that's ok too
[16:45] <bindi> mux= <- nothing?
[16:46] <bindi> should it be mux=flv
[16:46] <burek> it is auto-detected from dst
[16:46] <burek> i'm not sure if vlc has its own flv muxer
[16:46] <burek> most probably mux=ffmpeg{mux=flv}
[16:46] <bindi> that's exactly what I've tried
[16:46] <burek> but i always leave it blank and it works
[16:46] <bindi> let me try
[16:46] <bindi> but how will flowplayer like mpegts?
[16:47] <burek> dst=0.0.0.0:port2/live.flv
[16:47] <bindi> but i'm sending it in mpegts am I not
[16:47] <burek> and vlc remuxes it to flv
[16:47] <bindi> alright let's give it a try
[16:49] <bindi> no video, no sound
[16:49] <bindi> but it plays something
[16:49] <bindi> libdvbpsi error (PSI decoder): TS discontinuity (received 1, expected 11) for PID 17
[16:49] <bindi> bunch of these
[16:50] <burek> tools - codecs information
[16:51] <bindi> actually
[16:52] <bindi> second
[16:52] <bindi> it looks horrible when I open it directly in VLC
[16:52] <bindi> H264 - MPEG-4 AVC (part 10) (avc1)
[16:53] <bindi> can I try flv on ffmpeg?
[16:53] <burek> well, ffmpeg's ts muxer is broken :(
[16:54] <bindi> :E
[16:54] <burek> so, use vlc, you'll have less trouble
[16:54] <burek> I gave you the command above
[16:54] <bindi> yeah I'll give it a go
[16:56] <bindi> works in vlc, no picture in flowplayer
[16:56] <bindi> i'll add mux=flv to the server one
[16:57] <bindi> nothing
[16:58] <bindi> also the quality is horirble for some reason
[16:59] <bindi> horrible*
[16:59] <burek> try mux=ffmpeg{mux=flv}
[16:59] <burek> and add -vvv to vlc cmd line
[16:59] <burek> vlc -vvv -I dummy ...
[16:59] <burek> to see errors
[17:00] <burek> if any
[17:00] <bindi> [0xda44c8] avformat mux warning: Missing frame rate, assuming 25fps
[17:00] <bindi> it's 30 :D
[17:00] <bindi> D: even
[17:00] <burek> it's a warning
[17:00] <bindi> yeah yeah
[17:00] <bindi> hmm, no picture
[17:00] <burek> then you didn't set your #transcode well
[17:00] <bindi> client-side?
[17:01] <burek> well yes, there is no transcode on server-side
[17:01] <bindi> http://pastebin.com/6MwF5UCF
[17:03] <bindi> [0xdb07c8] ts demux error: MPEG-4 descriptor not found
[17:03] <bindi> the same command line with http access had good quality
[17:03] <bindi> though it was a flv container
[17:04] <burek> remove --sout-x264-preset=fast
[17:04] <burek> and change "venc=x264" to "venc=x264{preset=fast}"
[17:04] <burek> same for profile
[17:05] <burek> preset=fast,profile=baseline
[17:05] <bindi> yes
[17:05] <burek> well transcode seems fine
[17:05] <bindi> what difference does this make
[17:05] <burek> btw MPEG-4 is h264/avc
[17:05] <bindi> still horrible quality
[17:06] <burek> in that message above
[17:06] <burek> set preset=medium,crf=20,profile=baseline
[17:07] <burek> and remove vb param
[17:07] <burek> should be better then
[17:07] <bindi> how will users define max bandwidth then
[17:07] <bindi> vb=2000 worked fine when using http and server connecting to client
[17:07] <burek> using crf and preset will take care of that
[17:08] <burek> it will go far below 1M
[17:08] <burek> so don't worry
[17:08] <JEEB> if you are dealing with streaming or other bandwidth-limited scenarios just setting a bitrate doesn't limit anything
[17:08] <JEEB> if you have limits you have to use (vbv-) maxrate and bufsize
[17:08] <burek> also increase crf (up to 50) to lower your bandwidth down and decrease quality
[17:08] <bindi> ok well
[17:08] <bindi> now it broke
[17:08] <JEEB> burek, do you see this what I'm writing?
[17:09] <burek> yes
[17:09] <JEEB> ok
[17:09] <bindi> burek: can you open http://bindi.shroudbox.net:4444/live.flv in your vlc to see how messed up this is :p
[17:09] <bindi> or should I just take a screenshot
[17:09] <JEEB> then do note that crf doesn't limit bitrate in any way and neither does bitrate otherwise than for the average bit rate on the length of the whole clip
[17:10] <JEEB> vbv-maxrate and bufsize are to set the bandwidth limitations
[17:10] <burek> yes I see the video
[17:11] <bindi> what's up with that? :p
[17:12] <burek> bindi, try with crf and preset first
[17:12] <burek> and see if it works better
[17:12] <burek> when you use vb, vlc ignores crf and preset
[17:12] <bindi> i don't have vb
[17:12] <bindi> i used preset=medium,profile=baseline,crf=20
[17:12] <burek> try crf=30
[17:13] <bindi> burek: same problems
[17:13] <bindi> and it's green
[17:13] <burek> what is scale=1 used for
[17:14] <burek> btw, your stream is around 200k now :)
[17:14] <bindi> this will be used for gaming so quality will be respected :p
[17:14] <bindi> uh, i don't know
[17:14] <burek> try removing it and setting the preset=fast
[17:14] <bindi> my friend and I have been fiddling around for this and I started working on his original command line that used the clients vlc as the http server
[17:16] <bindi> broken still
[17:18] <bindi> lol
[17:18] <bindi> that looks weird
[17:18] <bindi> you watching it? :p
[17:19] <bindi> parts of it are not updating
[17:19] <burek> I think the cpu cannot cope with it
[17:19] <bindi> whose?
[17:20] <burek> try preset=ultrafast
[17:20] <bindi> better quality has been seen as I said before, over http
[17:20] <bindi> don't see how this would break
[17:21] <burek> just try for a test
[17:21] <bindi> sure
[17:21] <bindi> actually the server might be running low on resources
[17:22] <bindi> one moment
[17:22] <bindi> but i don't think that would affect this much
[17:22] <burek> server is just doing remuxing so it's cpu usage is really low
[17:22] <burek> 1-2%
[17:22] <burek> client, that is encoding the video, should be checked
[17:23] <bindi> well there
[17:23] <bindi> it looks normal but quality is horribad :D
[17:24] <bindi> actually i'm not sure if the quality is bad or if vlc is being a bad player :p
[17:24] <bindi> gotta get it working in flowplayer
[17:25] <bindi> flowplayer plays sounds, vlc doens't
[17:25] <bindi> doesn't
[17:25] <burek> are you using ffmpeg as a source or vlc
[17:25] <bindi> vlc
[17:25] <burek> the latest version?
[17:26] <pespin> burek, (baseline h264 encoder using libx264) I've been looking at the libx264.c file in libavcodec and it seems the only way to change the profile is using the command line option -profile.
[17:26] <bindi> let's see
[17:26] <bindi> actually no it's 1.1.9 because 2 bugged something
[17:26] <bindi> i can try it with 2
[17:26] <burek> pespin, that's when you use x264 as a cmd line tool, but if you use libx264 as a library, then there is a way through its api/abi
[17:26] <pespin> burek, yeah but not directly from libavformat
[17:27] <pespin> libavcodec sorry
[17:27] <burek> no no, no need for libavformat
[17:27] <burek> wait, what exactly do you want to do?
[17:27] <burek> use libx264 to encode your stream programmatically
[17:27] <burek> right?
[17:28] <pespin> burek, I have a bunch of code already passed from other people which uses av_codec to encode frames to h264, but I need to change it in order to get frames in baseline profile
[17:28] <pespin> burek, yeah
[17:28] <burek> take a look at libavcodec/libx264.c
[17:28] <pespin> I did
[17:28] <bindi> burek: yeah vlc 2 doesn't work
[17:28] <burek> that's the ffmpeg's wrapper for libx264
[17:28] <bindi> avcodec encoder error: cannot open encoder
[17:28] <burek> bindi, pastebin
[17:28] <bindi> sec
[17:28] <bindi> I can't hilight this
[17:28] <pespin> burek, I did look at the code there and the key is the char* profile in the private structure
[17:29] <bindi> screenshot will have to do
[17:29] <bindi> http://bindi.shroudbox.net/leiska/dl/4ffaf827e8f5f.png
[17:29] <burek> pespin, well ffmpeg uses x264's native -preset -profile -tune and -crf
[17:29] <burek> so it's possible it's only passed as a char* param to the lib too
[17:29] <pespin> burek, the only way to set that string is throught the options from cmd line, and that string needs to be set (non null) in order to set the profile inside that string
[17:30] <burek> pespin, I don't understand how does it make things not work for you?
[17:30] <bindi> burek: different audio codec for me? :p
[17:30] <pespin> burek, how can I set that string to "baseline" ?
[17:30] <pespin> it's not accessible from the public api
[17:30] <burek> just set char * bla ="baseline";
[17:30] <burek> and pass it
[17:31] <burek> type x264 --help to see other possibilities
[17:31] <burek> bindi, what? :)
[17:31] <pespin> burek, I need to change libavformat code then...
[17:31] <bindi> burek: did you see the screenshot?
[17:31] <bindi> it whines about audio codecs
[17:31] <bindi> i'm using acodec=mp4a
[17:32] <burek> oh man
[17:32] <pespin> burek, libx264.c line 439 from last git
[17:32] <burek> you need to compile your vlc..
[17:32] <burek> that vlc is compiled with I don't know what
[17:32] <bindi> burek: downloaded it off the website
[17:32] <pespin> burek, that string can't be set from the C api afaik
[17:32] <burek> bindi, use -vvv to get more info
[17:32] <bindi> riight
[17:33] <pespin> which tells x264 the profile to use
[17:33] <pespin> the string pointer is only available to the cmd line options and from inside the C file.
[17:34] <bindi> burek: it's spamming me a lot :-D
[17:34] <bindi> main mux warning: late buffer for mux input
[17:34] <pespin> or can I access context->priv_data->profile field?
[17:34] <burek> pespin, x264_param_apply_profile(&x4->params, x4->profile)
[17:34] <burek> did you take a look how does x264_param_apply_profile() apply it?
[17:34] <pespin> burek, yes
[17:35] <pespin> burek, ok, then I need to use the x264 API directly for this :)
[17:35] <pespin> ah wait, no way, same problem
[17:35] <bindi> burek: it wont play anything in vlc, but when I quit vlc (on my desktop) I see some video
[17:35] <pespin> x4 structure is not public
[17:36] <pespin> burek, it's only available to the .c file
[17:37] <burek> http://git.videolan.org/?p=x264.git;a=blob;f=common/common.c;h=3f40e66f11c205259ed9cc9226960bcb404043fb;hb=999b753ff0f4dc872077f4fa90d465e948cbe656
[17:37] <burek> ctrl+F x264_param_apply_profile
[17:38] <burek> and above it
[17:38] <burek> static int profile_string_to_int( const char *str )
[17:38] <burek> so you pass the char* to that int func to set your desired profile
[17:38] <burek> not a big deal
[17:38] <burek> so where is the problem?
[17:39] <burek> bindi, vlc ... > log.txt 2>&1
[17:39] <bindi> burek: i got it working by restarting everything
[17:39] <bindi> but crf=20 is still bugging
[17:39] <burek> nice :D
[17:39] <bindi> and crf=30 looks bad
[17:39] <burek> are you using Windows ? :)))
[17:39] <bindi> hah
[17:39] <bindi> yes
[17:39] <burek> figures :)
[17:39] <bindi> but I restarted vlc server-side too
[17:40] <bindi> by everythinf I don't mean my OS though
[17:40] <bindi> just the streamer and the re-streamer
[17:40] <burek> ok :)
[17:40] <bindi> so err
[17:40] <burek> so, put a crf in between 20-30
[17:40] <burek> and fine tune it
[17:40] <bindi> how can I get this to display in flowplayer
[17:40] <bindi> and why does it look like arse
[17:40] <burek> it should play in flowplayer too
[17:40] <bindi> it doesn't
[17:40] <burek> because of preset=ultrafast
[17:40] <bindi> it's still mpeg-4 or something
[17:40] <burek> and too high crf
[17:41] <burek> mpeg-4/avc is h264
[17:41] <bindi> right
[17:41] <bindi> well
[17:41] <bindi> I used mux=ffmpeg{mux=flv} in the server command line
[17:41] <bindi> won't play
[17:41] <bindi> http://bindi.shroudbox.net/stream/
[17:41] <burek> it needs that
[17:41] <burek> without that it won't work in flowplayer
[17:41] <bindi> yes
[17:41] <bindi> as I said
[17:41] <bindi> I used that
[17:41] <bindi> still am using
[17:41] <burek> try in #videolan to see why doesn't it work
[17:41] <pespin> burek, yeah, and how do I get x264_param_t    params from the libavformat I'm using?
[17:42] <bindi> i'll try vlc 1 on the server instead :D
[17:42] <pespin> burek, the x264 specific stuff is hidden from libavformat API
[17:42] <pespin> I can't access those fields
[17:42] <pespin> look at libx264.c
[17:43] <pespin> libavcodec sorry
[17:43] <burek> pespin, take a look at x264_param_default_preset
[17:43] <burek> you'll figure it out from there
[17:44] <bindi> burek: is there no way to transfer flv from the user to the server?
[17:45] <burek> you can try :)
[17:45] <pespin> burek, how can I pass it to the libavcodec encoder once I use it?
[17:45] <bindi> well vlc only supports ts over udp
[17:45] <burek> pespin, I believe you just initialize a default x264_param_default()
[17:46] <burek> and pass it to x264_param_apply_profile(param, "baseline")
[17:46] <burek> because that param object is what holds all the encoding options
[17:46] <pespin> burek, yeah, but then I need to pass those params to the libavformat encoder so it uses those params
[17:46] <burek> and that will be changed according to chosen profile
[17:47] <burek> pespin, that's why you need to take a look how did ffmpeg do that :)
[17:47] <pespin> burek, I did, cmd line option is the only eway
[17:47] <burek> :)
[17:47] <burek> ok
[17:47] <pespin> so, that should be changed I guess
[17:48] <burek> no
[17:48] <burek> you just need to read and understand it more thoroughly
[17:48] <pespin> burek, yeah, libx264.c is not making use of the context->profile field, it's not passing it to x264, that's missing I guess
[17:49] <bindi> burek: when I try to use ffmpeg as the origin, vlc goes mad
[17:49] <bindi> [0x1be84a8] ps demux warning: garbage at input, trying to resync...
[17:49] <bindi> [0x1be84a8] ps demux warning: found sync code
[17:49] <bindi> a lot of this.. this is killing me
[17:51] <burek> bindi, mpegts is broken, remember?
[17:51] <bindi> oh yes
[17:51] <bindi> but I used flv
[17:51] <burek> pespin, I don't understand
[17:51] <burek> flv is not segmentable
[17:51] <bindi> nngh
[17:51] <bindi> back to vlc it is
[17:52] <burek> pespin, initialize an empty x264 param
[17:52] <pespin> burek, when I do calls to the libavcodec API
[17:52] <burek> and pass that param and char* "baseline" to the func
[17:52] <burek> that's all it is
[17:52] <pespin> it internally has handles to x264 structures
[17:53] <burek> if ffmpeg can do it, you can do it too :)
[17:53] <pespin> burek, ffmpeg does it throught cmd line options
[17:53] <burek> well make it a constant
[17:54] <pespin> burek, libx264.c line 551
[17:55] <pespin> burek, that's the only door to the internal x264 structure  outside from libx264.c
[17:55] <burek> just analize that file and you'll figure it out, I'm sure :)
[17:55] <burek> ffmpeg has the same access rights through the x264 api/abi, just like you do
[17:55] <burek> so there is nothing special about it
[17:56] <burek> initialize an empty x264 param, set the bit rates, frame rates, etc and pass it to the function to set the profile
[17:56] <bindi> burek: http://pastebin.com/JkK3cUJk what's up with this
[17:56] <burek> I have 4 more minutes here before I go :)
[17:57] <burek> bindi, I really don't know.. ask in #videolan
[17:57] <bindi> ugh
[17:57] <bindi> fine
[17:57] <burek> most probably its ffmpeg's ts
[17:59] <bindi> i might just go back to http and flv from the user
[17:59] <bindi> and connecting to the user instead
[18:01] <bindi> sometimes it errors like hell
[18:01] <bindi> sometimes it works (picture only)
[18:01] <bindi> yep, back to original setup
[18:03] <pespin> burek, I do need to pass those parameters when I decode a frame right?
[18:03] <pespin> if I call the x264 decode frame funcs throught libavcodec, it passes the internal libavcodec fields
[19:18] <Ionic`> hi all
[19:19] <Ionic`> does ffmpeg have problems writing to a fifo?
[19:19] <Ionic`> I can read from stdin and write into a regular file without any problems, but writing to a fifo seems to fail
[19:19] <Ionic`> writes a few KB and halts
[19:36] <Mjiig> how can i turn a single .jpeg into an 11 second video file? all the documentation i can find is for multiple image files being stitched together
[19:48] <Mjiig> ugh, sorry to bother you, found it now
[20:47] <Fire_Eater64> If I have yasm installed in a custom directory, how can I point ./configure ffmpeg at it?
[20:51] <burek> Fire_Eater64, type ./configure --help
[20:53] <eduardosouza> hey guys, I am using ffmpeg -i 2.MOV -vcodec mpeg4 -s 640x480 test1.mp4 but it gives me an error Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[20:54] <eduardosouza> I am trying to make the videos smaller in size, keeping the best quality as possible.
[20:54] <burek> ok, but without the output how can we help you
[20:57] <eduardosouza> https://gist.github.com/ede94d5fad2f5e72a28d
[20:58] <eduardosouza> here it is
[20:59] <burek> why --disable-yasm
[20:59] <burek> you loose a lot of optimizations that way
[21:00] <burek> Stream #0:1(und): Audio: none
[21:00] <burek> this is the problem I guess
[21:00] <burek> try: ffmpeg -i 2.MOV -vcodec mpeg4 -s 640x480 -acodec copy test1.mp4
[21:01] <eduardosouza> it was giving me a trouble when installing on my mac os
[21:02] <burek> well, you could take a look at static binaries, if compilation is troublesome for you
[21:02] <burek> those 2 urls, try both of them
[21:02] <burek> and it would be wise to use libx264 instead of mpeg4
[21:02] <burek> it's much better in quality
[21:04] <eduardosouza> yeah, well the command above worked. But the quality of the video is really low now...
[21:04] <burek> that's because you didn't specify any encoding params
[21:04] <burek> try: ffmpeg -i 2.MOV -vcodec mpeg4 -q 2 -s 640x480 -acodec copy test1.mp4
[21:06] <eduardosouza> quality about the same but much bigger file
[21:06] <eduardosouza> also video is rotating 90 degres
[21:06] <burek> then you need to compile your ffmpeg with libx264 support
[21:06] <eduardosouza> got it
[21:08] <burek> http://ffmpeg.org/trac/ffmpeg/wiki
[21:08] <burek> there you have some tutorials how to do it easily
[21:08] <eduardosouza> thanks
[21:26] <eduardosouza> hey guys I am trying to reinstall ffmpeg using livx264 but it gives me an error that yasm not found. I am on a mac os. Where can I get that yasm?
[21:29] <burek> http://yasm.tortall.net/
[21:29] <burek> :)
[21:32] <Fire_Eater64> Is there an ffmpeg static build with libfaac installed?
[21:32] <newl> yasm is one of the best investment you can do to speed up libx264
[21:33] <burek> Fire_Eater64, I'm not sure, but why would you need it anyway.. ffmpeg has got his own internal aac encoder
[21:33] <burek> and there is libvo_aacenc
[21:33] <burek> newl, I agree :) there should be like big red warning whoever types --disable-yasm :)
[21:39] <Fire_Eater64> burek, when I replace -acodec libfaac with aac, I get "Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[21:39] <Fire_Eater64> "
[21:41] <Fire_Eater64> Actually, don't worry - the source audio was already aac :)
[22:08] <Ionic`> burek: actually, I've found information on SO, seems like ffmpeg does indeed NOT support named pipes
[22:08] <Ionic`> or fifos
[22:12] <Ionic`> hmmm
[22:12] <Ionic`> I wonder if tee would work
[22:41] <TheM4ch1n3> hello, How do I select one audio channel, from a source file, when I just want sterio in the output, and the inout has around 5 sterio channels ?
[22:41] <TheM4ch1n3> I tried, map, but its not working ... ?
[22:48] <burek> Ionic`, what are you talking about :)))
[22:48] <burek> of course it supports
[22:48] <burek> I've used it several times
[22:49] <DelphiWorld> Hello
[22:49] <DelphiWorld> michaelni, hello
[22:49] <burek> TheM4ch1n3, can you explain a little bit more?
[22:49] <DelphiWorld> hi, burek ;)
[22:50] <burek> hi :)
[22:50] <DelphiWorld> in the arab world there's a FOOD called burek !
[22:50] <TheM4ch1n3> where in a manual should I look, I looked into audio options, and -map might be it, should I carry on playing with -map ?
[22:50] <burek> yes, I know :)
[22:50] <DelphiWorld> burek: aguin a issue blowing up on my face rtmp://ns8.indexforce.com/ann
[22:50] <burek> that's exactly what my nick means :)
[22:51] <DelphiWorld> burek: you lie, realy ,
[22:51] <burek> TheM4ch1n3, are you looking for 5.1 to stereo conversion?
[22:51] <burek> DelphiWorld, the meat pie
[22:51] <DelphiWorld> burek: :-P
[22:51] <DelphiWorld> burek: so so ... rtmp://ns8.indexforce.com/ann :(
[22:52] <TheM4ch1n3> burek: no, just choosing the right audio track, has id's like 0.1, 0.2, 0.3 now what ?
[22:52] <burek> who is ann :)
[22:52] <DelphiWorld> burek: http://www.anntv.tv/
[22:52] <burek> TheM4ch1n3, audio channels are inside one audio stream
[22:52] <burek> you can't select a single audio channel with -map
[22:52] <burek> you can select entire audio stream, though
[22:52] <TheM4ch1n3> oh, well how do I select an audio stream
[22:53] <burek> -map :)
[22:53] <TheM4ch1n3> ok, thanks
[22:53] <burek> if you need to convert that stream from 5.1 to stereo
[22:53] <burek> use "pan" audio filter
[22:53] <burek> DelphiWorld, what's up with that url?
[22:54] <DelphiWorld> burek: like i say the rtmp stream is geving only a generic error msh
[22:54] <DelphiWorld> msg*
[22:54] <burek> well, send a report to the site admin then
[22:55] <DelphiWorld> burek: WTF?
[22:55] <DelphiWorld> burek: i mean the rtmp url!
[22:55] <DelphiWorld> rtmp://ns8.indexforce.com/ann
[22:55] <burek> my god.. asp...
[22:55] <burek> go to view source
[22:55] <burek> and ctrl+f "__VIEWSTATE"
[22:55] <burek> :D
[22:55] <burek> microsoft all over :)
[22:56] <burek> instead of keeping the session id, like all normal people would do, they keep entire session in a string :)
[22:56] <DelphiWorld> burek: wtf...
[22:56] <burek> so morbid..
[22:57] <burek> DelphiWorld, what exactly is the problem?
[22:57] <DelphiWorld> burek, let me see it
[22:58] <DelphiWorld> burek
[22:58] <DelphiWorld> Closing connection: NetStream.Play.StreamNotFound
[22:58] <DelphiWorld> rtmp://ns8.indexforce.com/ann: Unknown error occurred
[22:58] <burek> the stream is down, not a big deal
[22:59] <DelphiWorld> burek: is down? you sure?
[22:59] <burek> that's what the message says
[23:05] <DelphiWorld> ok burek mms://38.96.175.165/live
[23:05] <DelphiWorld> burek: if i have this stream
[23:06] <DelphiWorld> would i be able to mcast it?
[23:06] <DelphiWorld> it's asf i seen
[23:11] <DelphiWorld> burek: also the rtmp stream work on the web site;(
[23:14] <TheM4ch1n3> ok, its working properly using -map , and I will use your advise if I encounter 5.1, thanks
[23:14] <burek> DelphiWorld, I don't know, really, I don't use rtmp
[23:15] <DelphiWorld> burek, you SHOULD
[23:15] <DelphiWorld> otherwise i will eat burek !
[23:15] <burek> because..?
[23:15] <burek> well eat me :D
[23:26] <DelphiWorld> burek: FACEPALM
[00:00] --- Tue Jul 10 2012