[Ffmpeg-devel-irc] ffmpeg.log.20131013

[burek]

[00:42] <undercash> hello
[00:43] <undercash> did you had report with audio sync problems lately?
[00:43] <undercash> i m just encoding standart files, and most of the video are out of sync
[00:48] <beastd> undercash: See http://ffmpeg.org/bugreports.html
[00:49] <undercash> thx
[00:50] <undercash> but i guess if there was a major bug ..  you would know yourself
[00:50] <undercash> kinda curious..
[00:50] <beastd> you can search the ticket database if you do not find your problem and decide to report it provide commandline and full, uncut console output. if it is possible provide a small sample too.
[00:50] <undercash> i dont want to report something I am ensure
[00:50] <undercash> not sure*
[00:51] <undercash> just thought you might have heard of something
[00:51] <beastd> undercash: also it would be worth it too try with current ffmpeg git master
[00:51] <undercash> i updated on 09/20 and since then , i think must of the videos are out of sync
[00:52] <beastd> undercash: i do not know of any major regression on top of my head. but at any means update again and see if the problems you obeserve still exist
[00:52] <undercash> yep why not
[00:52] <undercash> thanks for your suggestions
[00:52] <beastd> good :)
[00:53] <beastd> also please understand that multimedia is a mean variety of all kinds of thinks chunked together in every conceivable way, so problems can appear in very isolated conditions :(
[00:54] <undercash> yes but I didnt had this problem before
[00:54] <undercash> it s either ffmpeg, either the remote rtmp server
[00:54] <undercash> hard to say, only downloading a copy of what I am encoding
[01:18] <denysonique> What is the safest format for an old TV to play?
[01:18] <denysonique> it plays some .avi's
[01:18] <denysonique> not sure about which codecs it understands
[01:25] <beastd> denysonique: there is usually no satisfying answer to your question. best is to search for the model and see what others succeded with if you find any. other than that i am afraid you will have to use trial and error.
[01:25] <denysonique> beastd: thanks. what codec should I begin with?
[01:26] <beastd> i can't possible tell you because i know nothing about your tv
[01:27] <denysonique> right, then what is the name of the codec which will play on a fresh install of Windows XP, without extra codecs being installed
[01:27] <beastd> if you mention the AVI i would guess there was a time: mpeg4 video and mp3 audio in avi was very popular so i would start that direction. but i am just guessing here
[01:28] <denysonique> yes, thats what I am thinking about as well, mpeg4
[01:32] <llogan> and provide a sample if possible so i can attempt to duplicate
[01:33] <undercash> hm ok
[01:34] <undercash> i dont use through terminal. but i can try
[01:34] <llogan> how do you use it?
[01:34] <undercash> webmin script
[01:34] <undercash> well.. bash script
[01:36] <undercash> -verbose?
[01:36] <undercash> sorry dont have in mind the debug option
[01:37] <llogan> no, just your command and the complete console output.
[01:37] <llogan> the debug stuff is usually more annoying than useful
[01:38] <undercash>  -loglevel verbose
[01:38] <undercash> ok
[01:38] <llogan> omit that
[01:38] <undercash> ah
[01:38] <undercash> ok
[01:47] <undercash> http://pastebin.com/pHWPGSnv
[01:49] <llogan> undercash: which player are you using? does ffplay play it normally?
[01:51] <undercash> i dont use ffplay
[01:51] <undercash> i can download it .. but this vid in particular might not be very interesting regarding dialogues..
[01:53] <llogan> what's with your version number? i think it should be 56548
[01:53] <llogan> are you using a shallow git repository?
[01:55] <llogan> maybe i should look in version.sh first but maybe it has issues with a shallow repo, or maybe I am confused
[01:57] <llogan> undercash: does using a different aac encoder show the same behavior?
[01:58] <beastd> hmm, the output seems to indicate that there is no audio stream at all...
[01:58] <undercash> i don't know
[01:59] <undercash> well i have audio for sure with my settings
[01:59] <undercash> since i broadcast 20 channels
[01:59] <undercash> but i notice this audio sync problem
[02:00] <beastd> undercash:  i am talking about your paste. can you make sure the input flv plays with audio. if it does can you test if ffplay plays it with audio too?
[02:00] <undercash> gonna broadcast it in 5min live..
[02:00] <llogan> heh. i didn't even notice that
[02:00] <llogan> at this point there isn't enough info from you to duplicate the issue or even narrow it down
[02:01] <undercash> yes I don't know, really.. just noticing many documentaries have audio out of sync. and since it didnt happen before, I thought I might shoot a word here
[02:02] <undercash> problem is I dont have the files locally.. it s downloaded directly on the server so I can't say.. gotta download them first
[02:03] <undercash> the vid i paste must have audio.. it s from youtube
[02:03] <llogan> if you play it i'm fairly sure you'll hear nothing due to a lack of an audio stream
[02:07] <undercash> the converted file has no audio indeed, i guess i chose a bad example.. but i was trying to find a rather short file
[02:10] <undercash> yea really unlucky video choice..
[02:10] <undercash> pff
[02:10] <undercash> no audio either on original file
[02:10] <llogan> problem solved. next!
[02:15] <undercash> not really
[02:16] <undercash> http://fr.justin.tv/xxxproducer1x#/w/7136049952/4
[02:16] <undercash> watch yourself
[02:16] <undercash> if u understand, tell me if the mouth act as the sound
[02:23] <undercash> you based your opinion on an unlucky example.. i didn't come here without reasons ;)
[02:50] <llogan> undercash: looks ok to me
[02:50] <undercash> no ;)
[02:50] <llogan> but it might be harder for me since I don't speak Russian.
[02:50] <undercash> slight delay
[02:50] <llogan> just kidding. french
[02:50] <undercash> ahah
[02:51] <undercash> but yea , i ll make some testing locally with a random file, then see how the converted file behave
[02:51] <undercash> i just thought it could have been a reported bug lately
[02:51] <llogan> not that i know of
[02:51] <undercash> i guess not
[02:52] <llogan> but try a local file and also see if the problem is from faac (you can try -acodec aac -strict experimental)
[02:54] <lakitu> i converted a file from avi to hv-whatever mp4, & it's got a bunch of green pixelation on it
[02:55] <undercash> yea i know this one, never used it on "production" though
[02:55] <lakitu> how to improve my command line fu to get rid of it?
[02:55] <lakitu> i have five 4gig avi files i'm trying to convert
[02:55] <undercash> i ll just recompile i guess
[02:55] <undercash> but it s boring.. need to shutdown a lot of stuff
[02:56] <llogan> if you recompile consider using fdk-aac instead of faac
[02:56] <llogan> https://trac.ffmpeg.org/wiki/AACEncodingGuide
[02:56] <undercash> interesting
[02:56] <undercash> i have fdk-aac but i guess my command doesnt use it
[02:57] <undercash> i kept using my libfaac command
[02:57] <lakitu> http://pastebin.ca/2466232
[02:57] <llogan> it should provide better quality per bitrate compared to faac
[02:57] <lakitu> ^ command
[02:58] <lakitu> i just compiled with fdk-aac
[02:58] <llogan> where's the complete console output?
[02:58] <lakitu> lost out of the buffer
[02:59] <lakitu> want me to run it again?
[02:59] <llogan> yes, but add "-t 10" so you don't have to encode the whole thing
[02:59] <undercash> gotta investigate this llogan
[03:00] <undercash> a proper fdk command
[03:00] <llogan> undercash: right now we don't know if: 1) it's faac's fault 2) it's ffmpeg fault 3) if it is RTMP or justin.tv fault 4) it is flash players fault 5) a combination
[03:00] <undercash> i think ffmpeg changed quite a bit since february..
[03:01] <undercash> can't recognize much in what i see
[03:01] <undercash> input.mp4 -c:v copy -c:a libfdk_aac -b:a 384k
[03:01] <undercash> geez..
[03:01] <llogan> the old names still work too if you prefer them
[03:01] <llogan> -vcodec -acodec -ab
[03:01] <undercash> oh good
[03:02] <undercash> -acodec fdk-aac  is fine then?
[03:02] <llogan> libfdk_aac as shown in the link i gave you
[03:02] <undercash> ;)
[03:02] <undercash> thank you
[03:02] <undercash> i ll try asap
[03:03] <undercash> ok ok -c:v -c:a
[03:03] <undercash> donno why they always changed those stuff
[03:03] <llogan> -codec:v/-c:v/-vcodec all do the same (ffplay only uses the old names still, AFAIK)
[03:04] <lakitu> what frame should this get to
[03:04] <llogan> it should create a 10 second output duration
[03:04] <lakitu> it's on the 56 thousandth frame
[03:04] <lakitu> seems much
[03:05] <undercash> yes i understand, except in the manuals that have been updated.. i guess i just need to accept hollidays are over :P
[03:05] <undercash> and work a bit
[03:05] <llogan> i would expect it would end on frame 250 if the input is PAL
[03:05] <lakitu> ok
[03:05] <lakitu> can i truncate it with a ctrl+c break?
[03:05] <llogan> or press "q" probably
[03:05] <lakitu> there
[03:05] <lakitu> it finished
[03:07] <lakitu> i ran out of buffer;
[03:07] <lakitu> extended my buffer & redoing it
[03:07] <lakitu> thanks for the help - that was probably like my 5th compile ever
[03:08] <llogan> you added -t 10? or you could use -vframes 100 instead
[03:08] <lakitu> i'll try vframes so i don't have to wait
[03:08] <lakitu> -t 10 is letting it run this long
[03:08] <lakitu> altho it's huge:
[03:08] <lakitu> 4gigs for 37minutes
[03:08] <lakitu> each
[03:08] <lakitu> i've got 5 of them
[03:08] <lakitu> 5 4gig files of 37minutes each
[03:08] <lakitu> which is why i need to compress them, obviously
[03:10] <lakitu> did the arguments have to be placed somehwere special?
[03:10] <llogan> yes. as output options they should be placed after "-i input"
[03:11] <lakitu> lemme try again
[03:11] <lakitu> ok
[03:11] <lakitu> it was a placement issue - copying
[03:12] <lakitu> http://pastebin.ca/2466234
[03:13] <lakitu> the no pixel format specified, is that it?
[03:13] <lakitu> someone in linux gave me the command
[03:13] <lakitu> #linux
[03:13] <llogan> Use -pix_fmt yuv420p for compatibility with outdated media players.
[03:13] <lakitu> so that?
[03:14] <lakitu> had i enough experience with linux, i would've known to look at the output for errors
[03:14] <llogan> it says that in the console output. so add -pix_fmt yuv420p as an output option
[03:14] <lakitu> ok
[03:14] <llogan> also see https://trac.ffmpeg.org/wiki/x264EncodingGuide
[03:14] <lakitu> ok
[03:15] <llogan> but if the output looks good enough then you don't need to do anything since the defaults are decent (except for -pix_fmt for general users)
[03:15] <lakitu> ok
[03:15] <llogan> what player was it looking shitty in?
[03:15] <lakitu> what's a good linux video editor, to sharpen/etc my personal lectures?
[03:16] <lakitu> these aren't just movies you know
[03:16] <llogan> kdenlive seems ok
[03:16] <lakitu> i made some video lectures - i'm not sure, the simplest one for fedora
[03:16] <lakitu> ok
[03:16] <llogan> but you should tweak the ffmpeg encoding presets in kdenlive
[03:16] <lakitu> can you separate off the audio track, edit it, & then put it back on?
[03:17] <lakitu> not sure what you mean
[03:17] <llogan> i haven't looked in a long time, but the ffmpeg settings may not be that great in kdenlive
[03:18] <llogan> as for the audio: ffmpeg -i video.mp4 -i audio.wav -map 0:v -map 1:a -codec:v copy -codec:a libfdk_aac output.mp4
[03:19] <llogan> this will take the video from video.mp4 and the audio from audio.wav. it will stream copy the video (no encoding) and encode the audio using libfdk_aac
[03:19] <llogan> http://ffmpeg.org/ffmpeg.html#Stream-copy
[03:21] <llogan> to get the audio: ffmpeg -i input output.wav
[03:32] <buhman> is libswresample considered higher quality and/or faster than libasound's built-in resampler?
[03:37] <lakitu> sweet, thanks logan. your advice worked, the pixel artifacts are gone
[03:38] <lakitu> saving your commands for how to edit the audio
[03:39] <lakitu> last question - i've got 1.5gigs for a 37 minute file - can i compress it (well) anymore?
[03:39] <lakitu> good compression means meaningfully smaller but still enough quality to make out whiteboard writing
[03:40] <lakitu> actually it's only 845 for some reason this time
[03:40] <lakitu> 845mb
[03:40] <lakitu> if that is optimal for quality being prioritized, what is optimal for size being prioritzied
[03:41] <lakitu> what would be a good small format, that is
[03:43] <undercash> its funny, been using -c:a libfdk_aac  and the stream didnt work
[03:43] <undercash> guess i dont have ffmpeg compiled with that library
[03:43] <undercash> though it s installed
[03:45] <lakitu> flv?
[03:50] <lakitu> or limit the kb/s
[05:05] <raytiley> Does anyone know the status of using directshow crossbar devices with ffmpeg. From googling seems like there was talk of support but I can't find anything less than 8 months old.
[05:28] <llogan> buhman: i'm not sure. there is also libsoxr which ffmpeg supports
[05:28] <llogan> https://trac.ffmpeg.org/wiki/FFmpeg%20and%20the%20SoX%20Resampler
[05:28] <buu> Hey, how do you control how many threads ffmpeg uses?
[05:29] <llogan> -threads <int>
[05:29] <buhman> llogan: well, I was looking more for comparisions rather that yet more alternatives ;p
[05:30] <buu> llogan: Well, ok, that's obvious, but why isn't it in man ffmpeg
[05:30] <buhman> buu: because the ffmpeg manual is split
[05:31] <buu> Oh my god
[05:31] <llogan> see "man ffmpeg-all" for the monolithic man page
[05:32] <buu> Don't have it =[
[05:32] <llogan> maybe your ffmpeg is old
[05:32] <buhman> llogan: ffmpeg-codecs(1) line 805
[05:32] <buu> llogan: 1.1.3
[05:33] <buhman> buu: that's post-man-split
[05:33] <llogan> i have no idea
[05:33] <buhman> though I think ffmpeg-all came later
[05:33] <buu> So I noticed =]
[05:33] <buhman> buu: no doubt you have ffmpeg-codecs though
[05:33] <buu> I do! I read it!
[05:33] <buhman> :D
[05:34] <buu> It says "threads: controls the number of threads"
[05:34] <buu> High quality documentation!
[05:34] <buu> =]
[05:34] <llogan> lakitu: 845mb is the input or the output?
[05:35] <llogan> buu: patches welcome
[05:35] <buu> Hrm. Why does my cpu only have 4 threads =[
[05:35] <buhman> well it makes sense to me, it's organized mostly by library obviously
[05:39] <llogan> buhman: i recall a resampler comparison with graphs 'n junk but i can't seem to find it
[05:40] <buhman> but specifically one that compares libasound's resampler to libswresample
[05:40] <buu> Does -threads default to something?
[05:40] <llogan> buhman: i can't remember. useful, huh?
[05:40] <buhman> buu: auto
[05:41] <buu> Oh
[05:43] <llogan> default 1, unless you're using en external encoding library that changes it, such as libx264 which is auto
[05:43] <llogan> frame based threads: 1.5 * logical processors, rounded down; slice based threads: 1 * logical processors
[05:43] <llogan> ^ for libx264
[05:44] <llogan> and you're probably doing frame based
[05:47] <buu> llogan: so its a h264, so that invokes libx264, which invokes 6 threads?
[05:47] <buu> 4 logical processors
[06:19] <llogan> buu: you can see the console output to show the number of utilized threads
[06:19] <buu> Oh.
[06:19] <buu> Cool.
[06:26] <llogan> buhman: old, but might be interesting and no mention of libasound: http://www.hydrogenaudio.org/forums/index.php?showtopic=99286
[10:06] <lakitu> some code i wrote from help i got earlier isn't quite working right - maybe osmeone can help me
[10:06] <lakitu> i'm putting it on pastebin now
[10:07] <lakitu> here's the bash shell code: http://pastebin.com/8nuZg3am
[10:08] <lakitu>  & then here's one result (it's a batch script). the goal was to lay wav audio tracks into mp4 files http://pastebin.com/HthUJ6nK
[10:11] <lakitu> i have that library, it works in another script
[10:11] <lakitu> or another command
[10:11] <lakitu> i just compiled specifically with that library
[10:27] <buu> What specifically isn't working?
[10:28] <lakitu> it says can't find the library
[10:28] <lakitu> like in the paste
[10:29] <lakitu> but i have that library, & was just usin git
[10:29] <lakitu> i'm pretty sure
[10:29] <lakitu> yes. i was
[10:29] <buu> Oh there's two pastes
[10:29] <lakitu> yeah
[10:29] <lakitu> i forgot i had it in a separate text file
[10:29] <buu> TRICKY
[10:29] <lakitu> gotcha.
[10:30] Action: lakitu reveals the candid cameras
[10:31] <buu> Are you sure you're using the same instance of ffmpeg in both places?
[10:32] <lakitu> hm, how would i - oh you're right
[10:32] <lakitu> i think!
[10:32] <lakitu> good catch...
[10:32] <lakitu> well
[10:32] <lakitu> wait
[10:32] <lakitu> i compiled it
[10:32] <lakitu> does that mean it's in a folder somewhere
[10:32] <lakitu> (directory =P)
[10:32] <buu> lakitu: Unless you installed it somewhere else..
[10:32] <lakitu> or it's the default 'ffmpeg'
[10:32] <lakitu> no
[10:33] <lakitu> because i was just calling 'ffmpeg {arguments}' & it was working, with that library
[10:33] <buu> lakitu: run this: ffmpeg -encoders
[10:33] <lakitu> there's a lot
[10:33] <lakitu> aac is x'd
[10:34] <buu> lakitu: So wait, you compiled your own version of ffmpeg? Where did you do it? How did you do it?
[10:35] <lakitu> https://trac.ffmpeg.org/wiki/CentosCompilationGuide
[10:35] <lakitu> in some folder, if that's what you mean
[10:36] <buu> lakitu: so you did the previous steps then ran ./configure ...; make; make install?
[10:37] <lakitu> yes, with the pre-./configure steps of getting the plugins
[10:37] <lakitu> or some of them
[10:37] <lakitu> like it shows
[10:37] <buu> lakitu: ok, what does find $HOME/ffmpeg_* | grep bin; show you?
[10:38] <buu> In otherwords, where did you install ffmpeg?
[10:38] <buu> And what does which ffmpeg; show you?
[10:38] <lakitu> i just copied-n-pasted, i have no idea
[10:38] <lakitu> the first says Is a directory
[10:38] <lakitu> the second says /usr/bin/ffmpeg
[10:39] <buu> lakitu: wait, what? it says is a directory?
[10:40] <buu> lakitu: Let me rephrase that, what files are in $HOME/bin; ?
[10:40] <lakitu> bash: /home/me/ffmpeg_build: Is a directory
[10:40] <buu> lakitu: You're supposed to put 'find' in front of that!
[10:41] <lakitu> oh =P
[10:41] <lakitu> good catch again
[10:41] <buu> lakitu: But I think you should have $HOME/bin/ffmpeg
[10:41] <lakitu> a bunch of stuff
[10:41] <buu> And that's the one you compiled
[10:41] <lakitu> must've
[10:42] <buu> Do you see what I'm getting at here?
[10:42] <lakitu> but it worked when i just used ffmpeg with that library
[10:42] <lakitu> do i have to specifically call that ffmpeg?
[10:42] <buu> Yes
[10:42] <buu> I have no idea what you did before
[10:42] <lakitu> but i think i wasn't
[10:43] <buu> Try it with $HOME/bin/ffmpeg instead
[10:43] <buu> and see what happens
[10:43] <lakitu> ok
[10:45] <lakitu> doing
[10:45] <lakitu> working. weird
[10:45] <lakitu> not sure, but i need asian soup bowl now. thanks
[10:46] Action: buu shrugs
[14:17] <khali> are there options I can pass to ffmpeg to improve the quality when encoding audio to AAC?
[14:17] <khali> I see suggestions for high quality video in the FAQ but nothing for audio
[14:52] <LithosLaptop2> khali: you can use a higher VBR setting  of 5 when using '-c:a libfdk_aac' by add '-vbr 5' instead of specifying the bitrate
[14:57] <LithosLaptop2> khali: when using libfaac try instead of libfdk_aac' then try to keep the bitrate higher than 192Kbit/s. IMO libfaac doesn't wotk well below 192Kbps. libmp3lame would be a better choice
[14:58] <LithosLaptop2> khali: for libfaac I would use something like: -c:a libfaac -q:a 330 -cutoff 15000
[15:00] <LithosLaptop2> khali: for the internal aac encoder try to keep the bitrate around/above 240Kbps: -c:a aac -b:a 240k -strict -2   For older ffmpeg builds you might need to add -cutoff 15000
[15:01] <LithosLaptop2> khali: never use libvo_aacenc
[15:01] <LithosLaptop2> thats about it for aac
[15:46] <khali> 192 kbps seems a lot for a movie audio track
[15:47] <khali> the whole point of AAC is that it compresses better than MP3
[15:47] <khali> and I wouldn't even consider 192 kbps for MP3
[15:47] <khali> at 192 kbps even MP2 is good enough...
[15:59] <sacarasc> khali: https://trac.ffmpeg.org/wiki/AACEncodingGuide
[16:16] <iive> 192kbps is the minimum for mp2/3
[16:16] <khayyam> hello, I'm having a linking error with the current svn: libavcodec/libavcodec.so: undefined reference to `ff_idct_xvid_mmxext_put' (and other ff_idct_xvid_* refs). I was previously able to build the tree, the last being Sept 29th, but something seems to have changed. The build.log: http://bpaste.net/show/140133 ... any ideas?
[16:50] <khali> iive: minimum for what?
[17:20] <iive> for quality
[17:20] <iive> and with mp2 i think it is the lowest possible bitrate
[17:32] <who0> Hi All, I would like to convert into "profile High, level 4.0" with 700k but ffmpeg says " profile High 4:4:4 Predictive, level 4.0, 4:2:0 8-bit" lossless :( -- Have you got some clue ?
[17:46] <khali> iive: mp2 128 kbps does exist
[17:47] <iive> mono?
[17:48] <khali> some DVB-T broadcaster use that for secondary languages here in France
[17:48] <khali> iive: no, stereo
[17:48] <khali> iive: but 128 kbps mp2 doesn't sound good
[17:48] <iive> maybe I should look up the standard.
[17:48] <khali> they really shouldn't be doing that
[17:48] <iive> 128kbps mp3 doesn't sound good.
[17:51] <khali> iive: I don't get your point on that
[17:51] <sacarasc> 128kbps MP3 is poo.
[17:51] <khali> iive: 128kbps mp3 used to be the standard for audio CD ripping
[17:51] <khali> so for a movie it seems good enough to me
[17:51] <who0> I found ... how to force rc=abr instead of rc=cpq with libx264 (it sounds good ?)
[17:52] <sacarasc> khali: The operative part there is "used to be". Not any more.
[17:53] <khali> sacarasc: isn't it odd given that encoder quality improved meanwhile?
[17:53] <khali> 128 kbps today is better quality than it was in the early 2000s
[17:53] <sacarasc> So has storage capacity.
[17:53] <sacarasc> 5GB in 2000 to 5TB now.
[17:54] <khali> sacarasc: that's a very good point, I admit
[17:55] <khali> but I still don't think it makes sense to put too much bitrate on the audio when it brings no listening benefit
[17:55] <khali> the audio streams come to me as 192 kbps or 256 kbps mp2, that's not CD audio quality
[17:56] <khali> I doubt it makes sense to reencode 192 kbps mp2 to 192 kbps AAC, as LithosLaptop2 was suggesting above
[18:00] <sacarasc> Reencoding lossy to anything is pointless, unless it's needed because of some hardware incapability.
[18:07] <iive> khali: it is rare, but once I had an anime with such poor audio quality that I had to find another release...
[18:08] <zap0> oh NO!
[18:08] <iive> saving few kbps when your video is mbps is really pointless.
[18:09] <LithosLaptop> try saying that to the people broadcasting our DVB-T2 channels
[18:10] <LithosLaptop> every channel has crappy 64Kbps audio
[18:32] <who0>  it dosen't work :( I have = high profile doesn't support lossless ... and I don't know why (google didn't help me)
[19:13] <iive> LithosLaptop: I hope thay are using at least aac :|
[19:14] <LithosLaptop> yeah AAC-HE
[19:16] <iive> well, aac-he have been developed for low bandwidth communication, like phones...
[19:59] <khali> sacarasc: that's exactly the problem... I'd leave mp2 audio streams as is, but most hardware decoders can't cope with mp2 audio streams in mp4 containers
[20:00] <khali> (192 kbps I'd leave as is... 256 kbps I'd certainly reencode anyway)
[20:03] <khali> iive: my video streams are in the 600-900 kbps range
[20:03] <khali> iive: so 128 vs. 192 vs. 256 kbps audio isn't that pointless
[20:03] <khali> or even 112 kbps, as this is what I use at the moment in most cases
[20:37] <Apic> Yoo hoo.
[20:37] <Apic> (UGT)
[21:13] <francogrex> is this for real? about ffmpeg: "*** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead."
[21:15] <JEEB> that is regarding the ffmpeg BINARY within the libav project
[21:15] <JEEB> not the ffmpeg project
[21:15] <francogrex> ah ok, because all I did was apt-get install ffmpeg and then got this message
[21:15] <JEEB> basically elenril rewrote much of the ffmpeg cli application over at libav, and the rewrite was called avconv
[21:15] <JEEB> yes, if you use a distro that uses libav, you use avconv
[21:15] <JEEB> because that's the more updated binary
[21:16] <JEEB> (within that project)
[21:16] <francogrex> ok
[21:16] <JEEB> ffmpeg (project) still has its ffmpeg (tool)
[21:17] <JEEB> the ffmpeg tool was then removed with the next libav release, IIRC. So it only has the avconv tool.
[21:24] <llogan> francogrex: for more info http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html
[22:11] <DrSlony> Hi, what is the best way to deshake video using ffmpeg-2.0? Is the following line still valid?
[22:11] <DrSlony> ffmpeg -i INPUT.mp4 -an -vcodec libx264 -preset superfast -crf 18 -vf "deshake=-1:-1:-1:-1:16:16:0:8:125:0:" -threads 0 out.mp4
[22:15] <durandal11707> DrSlony: it should still work
[22:16] <DrSlony> is there any better method? or better parameters?
[22:19] <llogan> DrSlony: also see http://ffmpeg.org/ffmpeg-filters.html#vidstabtransform
[22:19] <llogan> (ive never tried it)
[22:23] <ubitux> you need a git version of vid.stab installed if you don't want any surprise
[22:50] <DrSlony> I have a MJPEG video which I would like to transcode to h.264 and also cut off the first 3 seconds, is this the most simple way to do it?ffmpeg -i
[22:50] <DrSlony> oops
[22:51] <DrSlony> ffmpeg -i foo.avi -ss 00:00:03 -vcodec libx264 -preset slower -crf 18 -threads 0 out.mp4
[22:52] <llogan> that will work
[22:52] <llogan> may need to add -pix_fmt yuv420p for "dumb" players; depending on your input and ffmpeg version
[22:53] <DrSlony> does it matter where i put that?
[22:53] <llogan> as an output option
[22:53] <llogan> but the console output will confirm if ffmpeg is outputting to yuv420p or not
[22:54] <llogan> but if you're not going to use any dumb players then don't worry about it
[23:10] <DrSlony> What audio format and library is recommended when transcoding typical home video (stereo) or screencasts intended to be emailed to friends and played on typical computers?
[23:18] <DrSlony> I used "-vcodec libx264" and it worked, but http://ffmpeg.org/ffmpeg.html shows "-c:v libx264". Which oldest version of ffmpeg does that apply to?
[23:18] <llogan> either works
[23:19] <llogan> h264 video and aac audio in mp4 container should be fine (as long as you include -pix_fmt yuv420p if needed)
[23:20] <llogan> some devices may require "-profile:v baseline"
[23:20] <DrSlony> should I use fdk or other?
[23:20] <llogan> fdk
[23:29] <DrSlony> thank you very much
[00:00] --- Mon Oct 14 2013