[Ffmpeg-devel-irc] ffmpeg.log.20111130

[burek]

[00:10] <_micah> Trying to test a video encode intended for streaming to a mobile device.
[00:10] <_micah> This is with 0.7.8 and this cmd: ffmpeg -i Heir_HD.mov -vf scale=320:-1 -vcodec libx264 -profile baseline -x264opts vbv-maxrate=364:vbv-buffer=182:level=1.2 mobile.mp4
[00:11] <_micah> results in http://pastie.org/2941341
[00:11] <_micah> Anyone got a suggestion?
[00:12] <newl_> xz
[00:12] <newl_> sorry
[00:21] <Alasdairrr> Does anyone here know of an rtmp stream I can connect to to test some stuff? Google isn't finding much :-/
[00:24] <pasteeater> _micah: you can try -vf scale=320:-1 -vcodec libx264 -vprofile baseline -x264opts vbv-maxrate=364:vbv-bufsize=182:level=1.2 mobile.mp4
[00:25] <_micah> Nope
[00:25] <_micah> [NULL @ 0x7fa4a406f400] Unable to parse option value "baseline"
[00:25] <pasteeater> change vprofile to profile
[00:25] <pasteeater> see if it likes that instead
[00:26] <_micah> Aha!
[00:26] <_micah> vbv-bufsize != vbv-buffer
[00:26] <_micah> thanks pasteeater 
[00:26] <pasteeater> Mista-D: you didn't show your ffmpeg command and the complete terminal output on that post.
[00:29] <newl_> Alasdairrr: want an aussi channel or cnn?
[00:29] <Alasdairrr> newl_: cnn would be fine! I'm in london
[00:30] <newl_> all i have is working cli ... you want to do it yourself? it was a HUGE pain to do lol
[00:30] <newl_> /usr/local/bin/rtmpdump -v -r rtmp://cp44679.live.edgefcs.net/live -y cnn_stream2_low at 2787 -a live -W http://i.cdn.turner.com/cnn/.element/apps/CNNLive/2.1.5.7/assets/swfs/LivePlayer.swf -p http://www.cnn.com/video/flashLive/live.html?stream=2 | /usr/local/bin/mplayer - &> /dev/null &
[00:31] <newl_> i haven't tested it in a while tell me if it still works - man that rtmpdump is a pain
[00:32] Action: newl_ would have offered Alasdairrr a cspan one but doesn't want any more users clogging the stream :)
[00:46] <Xd7mT> i met another problem. its async of audion and video. i know that are PTS and DTS frames work with it, but something wrong and audio begins too early
[00:48] <Xd7mT> another problem: i have 2 files: audio and video. but audio is faster (after about 3 hours audio going to be early on 3 seconds). hoe to fix it?
[01:02] <Alasdairrr> newl_: thanks for that
[01:02] <Alasdairrr> I don't suppose you know of an SSL encrypted one?
[01:04] <newl_> how would that work?   you mean probably through a tunnel
[01:04] <nathan0n5ire> burek are you on?
[01:05] <Alasdairrr> rtmpdump supports SSL it seems
[01:06] <Alasdairrr> I'm basically adjusting rtmpdump to support stream timeouts on Solaris 10, which lacks SO_RCVTIMEO
[01:06] <Alasdairrr> just want to test my code
[01:09] <burek> nathan0n5ire, yes?
[01:09] <nathan0n5ire> burek I wanted to make a logo for this task http://www.google-melange.com/gci/task/view/google/gci2011/7170237
[01:09] <nathan0n5ire> but first I wanted what you expected from it
[01:10] <Alasdairrr> finding an ssl encrypted rtmp stream is going to be difficult i bet
[01:12] <burek> Xd7mT: https://ffmpeg.org/trac/ffmpeg/ticket/692
[01:13] <burek> nathan0n5ire
[01:13] <burek> we expect you just to be creative
[01:13] <burek> that's all :)
[01:13] <burek> if we would knew what we want, we would have already made it :)
[01:15] <nathan0n5ire> burek also is a raster image fine or does it need to be vector?
[01:17] <burek> nathan0n5ire, imagine you are creating some logo for your own company or business
[01:17] <burek> or something similar
[01:17] <burek> what would you do and use? :)
[01:20] <newl_> Alasdairrr: does it really support ssl ? rtmps ?
[01:20] <Alasdairrr> newl_: rtmpdump claims to support rtmps yeah
[01:20] <Alasdairrr> it has code to do it
[01:20] <Alasdairrr> with multiple different SSL/TLS libraries (gnutls/openssl)
[01:21] <newl_> when you find one for ssl paste it here ;)
[01:21] <newl_> the overhead would be terrible
[01:21] <Alasdairrr> ssl can be encrypted in realtime, its not a big deal
[01:22] <newl_> indeed it does ssl - i have the INC in the config script
[01:26] <burek> if someone really wants to read your stream, ssl would not stop him
[01:26] <burek> i personally think you are just wasting cpu cycles with it
[01:31] <newl_> what he is connecting to and from where would not be hidden by the ssl 
[01:44] <plethora> also, encrypting compressed data isn't very safe, http://software.imdea.org/~bkoepf/papers/esorics10.pdf
[01:58] <burek> yes, the stream is in a well known format, so it's easy to decypher it
[01:59] <burek> decipher*
[04:38] <nathan0n5ire> what does ffmpeg stand for?
[04:43] <newl_> if you saw ff on a remote control .. what would you guess it stood for?
[05:12] <kcm1700> fast forward?
[05:15] <relaxed> +motion picture expert group
[05:47] <nathan0n5ire> hey all, I'm making a icon for ffmpeg and am wondering how we want it to look?
[05:47] <nathan0n5ire> one possible idea is http://f.imgtmp.com/Btx89.png
[05:48] <Whitewizard> people! woot
[05:49] <Whitewizard> cf. x264 --fullhelp
[05:49] <Whitewizard> whats that mean?
[05:52] <Whitewizard> anywho, ping me when someone gets around to it :p
[05:53] <relaxed> Whitewizard: it means run the command `x264 --fullhelp`
[05:54] <Whitewizard> always get this  Unable to find a suitable output format for 'x264'
[06:00] <relaxed> Whitewizard: command?
[06:00] <Whitewizard> ffmpeg x264 --fullhelp
[06:00] <relaxed> x264 --fullhelp | less
[06:01] <nathan0n5ire> could some people give me some feedback for this ffmpeg icon I created? http://f.imgtmp.com/Btx89.png
[06:02] <Whitewizard> 'less' is not recognized as an internal or external command
[06:02] <Whitewizard> broken
[06:02] <relaxed> Whitewizard: Windows?
[06:02] <Whitewizard> vista
[06:02] <Whitewizard> yes
[06:03] <relaxed> what do you want to do?
[06:03] <Whitewizard> see what it says
[06:03] <relaxed> nathan0n5ire: It's fine. Why not use the logo at ffmpeg.org?
[06:03] <MeateaW> try more instead of less
[06:03] <Whitewizard> end goal is making .mp4s for a samsung reality
[06:03] <MeateaW> (the command is more in windows)
[06:04] <Whitewizard> back to this  Unable to find a suitable output format for 'x264' when i use more
[06:06] <relaxed> Whitewizard: An android phone? Try, ffmpeg.exe -i YOURINPUT -vcodec libx264 -profile:v baseline -threads 0 -crf 18 -acodec aac -strict experimental -ab 160k -ac 2 output.mp4
[06:07] <Whitewizard> baseline profile necessary? :s
[06:08] <relaxed> You can try main too. Sometimes it's easier to start with simple options and work your way up when targeting a device.
[06:12] <Whitewizard> failed!
[06:12] <Whitewizard> lets try one using.. mpeg4
[06:12] <Whitewizard> 2
[06:12] <relaxed> Whitewizard: http://pastebin.com/
[06:12] <Whitewizard> whats that for?
[06:12] <relaxed> playback failed?
[06:13] <Whitewizard> yeah phone doesnt like h264 i guess
[06:13] <Zeranoe> Has qscale been removed, or does it depend on GPL and therefor if FFmpeg is compiled for LGPL this feature will not be included?
[06:14] <relaxed> Zeranoe: I've never heard of that.
[06:15] <Zeranoe> relaxed: Really? i have a client asking this question, I'm just relaying it.
[06:16] <Zeranoe> http://ffmpeg-users.933282.n4.nabble.com/What-does-qscale-do-td941320.html
[06:18] <relaxed> where in that link does he say -qscale is missing?
[06:19] <Zeranoe> relaxed: It doesn't I thought you meant you haven't ever heard of -qscale. Should -qscale be supported in all versions of FFmpeg, including LGPL?
[06:20] <reflexrg> how do I list the audio or videos quality?  the details of the codecs playback specs or is this done with another commandline app?
[06:20] <relaxed> Zeranoe: Sorry, I meant that I've never heard anyone else say it was missing.
[06:21] <Zeranoe> relaxed: Thanks, I'll investigate then.
[06:52] <pasteeater> reflexrg: can you clarify your question? I don't understand what you are asking.
[06:53] <reflexrg> well I'd like to use only commandline apps for doing audio video converting. But I want it to always sound its best so I look at the bitrate so I know if I need to tone it down a little so its not higher than what I was converting from. 
[06:53] <reflexrg> the details of the audio or video. Bitrate, FPS...etc...
[06:54] <reflexrg> I just use the gui to see this but I'd like to see it in commandline
[06:55] <reflexrg> list the details I mean
[06:57] <pasteeater> you can see info with: ffmpeg -i input
[06:59] <pasteeater> or try mediainfo
[06:59] <reflexrg> thanks pasteeater
[07:00] <pasteeater> ...but not all encoders are equal, so comparing bitrates may be a waste of time.
[07:00] <reflexrg> I see
[07:00] <reflexrg> yes I thought probably so
[07:01] <reflexrg> because it did ahve a bitrate that did qaulity settings according to the codec
[07:01] <reflexrg> see
[07:01] <reflexrg> in the optins for ffmpeg
[07:01] <pasteeater> what format is your output?
[07:01] <reflexrg> mp3 
[07:01] <reflexrg> input is mp4
[07:02] <reflexrg> from youtube
[07:02] <pasteeater> then do: ffmpeg -i input -acodec libmp3lame -aq 4 output.mp3
[07:02] <pasteeater> adjust aq for quality.
[07:02] <pasteeater> it's mapped to lame's -V option.
[07:02] <reflexrg> aq is better than ab?
[07:02] <reflexrg> what's the difference?
[07:03] <pasteeater> read this: http://wiki.hydrogenaudio.org/index.php?title=LAME
[07:04] <pasteeater> you probably want vbr, which aq will give you.
[07:06] <reflexrg> bitrate is way higher than the bitrate for the audio in the mp4 ffacc
[07:06] <reflexrg> 125 kbs compared to 160 kbs
[07:07] <reflexrg> I'll have to read about the differences between bitrates and audio codecs 
[07:08] <pasteeater> with -aq the encoder will choose the proper bitrates to achieve your desired quality level (the value you give for -aq).
[07:09] <reflexrg> do you need to have ac option in there or is that not need with aq?
[07:09] <pasteeater> generally, either you encode for quality (and don't give a rat's ass about bitrate) or for size (where bitrate is important)
[07:10] <pasteeater> the output will inherit the ac of the input if possible, so you only usually use ac if you want to change the number of channels such as stereo to mono
[07:11] <reflexrg> I don't like distortions and compression artifacts in the audio
[07:11] <pasteeater> then use ac
[07:11] <pasteeater> duh...i mean aq
[07:11] <reflexrg> so I prefer to use best settings as possible 
[07:11] <pasteeater> everyone does
[07:11] <reflexrg> okay thangs buddy 
[07:12] <pasteeater> read the link i gave you. most people probably can't tell a difference between the original input and an output with -aq 4 for most inputs.
[07:12] <reflexrg> I see
[07:13] <pasteeater> also remember that the aq values will be different if you use other external encoders such as libfaac.
[07:14] <reflexrg> when I do the commandline options you suggested it doesn't tell audacious the correct playing time it has
[07:14] <reflexrg> other the other options I used from online
[07:14] <reflexrg> over*
[07:14] <pasteeater> i tihnk that's an old bug that was fixed in newer ffmpeg.
[07:15] <pasteeater> you can pipe to lame instead. that might work in audacious
[07:15] <pasteeater> ffmpeg -i input -f wav - | lame -V4 - output.mp3
[07:15] <reflexrg> V is like aq?
[07:16] <pasteeater> yes. consider it the same thing.
[07:16] <reflexrg> okay
[07:16] <reflexrg> thanks buddy
[07:17] <reflexrg> wav would be mp4 or acc?
[07:17] <reflexrg> in my case
[07:17] <pasteeater> i don't understand
[07:18] <reflexrg> -f aac
[07:18] <reflexrg> ?
[07:18] <pasteeater> you wanted mp3
[07:18] <pasteeater> oh, the -f wav...
[07:18] <reflexrg> ffmpeg -i input -f aac - | lame -V4 - output.mp3
[07:18] <pasteeater> you want it to be -f wav.
[07:18] <reflexrg> why wav?
[07:18] <reflexrg> its not in wave
[07:18] <reflexrg> wav*
[07:19] <pasteeater> that command is telling ffmpeg to output wav and feed it to lame.
[07:19] <reflexrg> ic
[07:19] <pasteeater> lame can't use aac as an input, as far as i know
[07:22] <reflexrg> thanks buddy that works
[07:24] <reflexrg> with audio I don't really agree people want the best you can see this in how Well HD for movies is doing compared to high quality Audio Discs for music like super audio CD.
[07:25] <reflexrg> at least HD is big so its easy for compatibility compared to SACD 
[07:25] <reflexrg> just HDTV and some kind of HDplayer 
[07:26] <pasteeater> people like convience. the masses are just fine listening to crappy music on lossy formats from their crappy earbuds from their phones and other devices
[07:27] <reflexrg> yeah your right
[07:27] <reflexrg> but there is inexpensive options out there for great sounding music
[07:30] <pasteeater> yeah. starting with your free encoder, lame.
[07:30] <reflexrg> haha
[07:30] <reflexrg> play on words
[07:30] <reflexrg> I thought about that too
[07:30] <reflexrg> mp3 is lame
[07:30] <reflexrg> I use ogg
[07:31] <reflexrg> oggconvert can't strip the video. would doing ogg from this mp4 sound okay?
[07:31] <pasteeater> but if you're concerned with quality then dont encode music from youtube videos if you can help it.
[07:31] <reflexrg> I don't like the sound quailty that winff makes the ogg files compared to oggconvert
[07:33] <reflexrg> I use jbuds lowest costing model from amazon 10 dollars and sandisk sansa clip plus 8gb. Best bang for the buck here at least from reviews says that sandisk uses one of its best soundcard in this device. well I am very happy with it. although the player doesn't last very long if I play flac files so I still with using ogg when I can or mp3 when I have to.
[07:33] <pasteeater> with youtube: input from who knows where that's probably been previously encoded several times with lossy encoders -> user who makes youtube video (encoded again) -> youtube (yes, they encode it again)
[07:33] <reflexrg> jbuds has very crisp great sounding bass and they are extremely light, small, comfortable to wear in all situations 
[07:35] <pasteeater> winff preset might be using crappy settings. if you want ogg vorbis: ffmpeg -i input -acodec libvorbis -aq 5 -vn output.ogg
[07:35] <reflexrg> the only thing I don't like about them is the paint that tells you which is left or right comes off very easily if you sleep with them
[07:36] <reflexrg> I use aq 4 with ogg when I convert from my flac or wav cds
[07:37] <reflexrg> of course for cd players and on my pc I use highest quality 
[07:38] <reflexrg> media devices don't last very long in battery life with highest quality
[07:38] <reflexrg> mp3 players
[07:39] <reflexrg> thanks bud
[07:39] <pasteeater> no problem. have fun with that.
[07:40] Action: pasteeater peels self off of computer chair
[07:46] <reflexrg> looks like no piping required for this one very happy with the sound quality and it displays the correct playtime
[07:46] <reflexrg> :)
[07:46] <reflexrg> with ogg
[07:51] <Whitewizard> when using -vcodec mpeg4 how can it to not look like.. crap?
[07:58] <relaxed> add -qscale 3
[08:58] <Whitewizard> relaxed: thanks, seems to have cleared up some of the cubist art, now theres horizontal lines of shadows in areas without detail.. is there anything i can do about them?
[09:04] <Whitewizard> nvm.. meebie im seeing things..
[10:37] <giany> hello
[10:37] <giany> any idea why if I run : 
[10:37] <giany> /usr/bin/ffmpeg -debug 40 -level 40  -i 'sites/default/files/videos/original/kb1_0.mp4' -s 658x362 -vf "pad=658:490:0:64" -r 15 -b 5k -ar 22050 -ab 5000 'sites/default/files/videos/converted/kb1_0_0.flv'
[10:38] <giany> I get : Floating point exception
[10:38] <giany> I'm using ffmpeg version 0.8.6
[14:46] <zap0> how do i use a single input png  and have a video output of, say 10 frames?
[14:46] <MadWoodworker> Is this the right place to find solutions to problems I am having with ffmpeg?
[14:47] <osearth> no
[14:48] <osearth> gotta ask Q first
[14:48] <zap0> i did
[14:48] <osearth> MadWoodworker: then will be ;]
[14:48] <MadWoodworker> I have recorded programs using an HDHomeRun3 and mythtv.  I try to convert them to MKV using ffmpeg, but can't get decent quality.
[14:48] <osearth> zap0: oops i just like to seem smart, doh
[14:48] <osearth> zap0 you don't want any audio chan?
[14:48] <zap0> no
[14:49] <MadWoodworker> Using DivX Converter works fine.
[14:50] <MadWoodworker> Can you suggest the right command line for me?  File size doesn't matter to me (right now anyway).
[14:50] <osearth> i would literally need the help file in front of me but you can set output file type and length
[14:50] <osearth> 10 frames at 10 frames per sec. might make more sense to specify total time.
[14:50] <JEEB> MadWoodworker, so basically input to H.264 video in matroska container?
[14:50] <osearth> and i'm outside my exp already
[14:50] <MadWoodworker> Yes.
[14:51] <JEEB> alrighty
[14:51] <osearth> youtube ripping i got you LOCKED ;]
[14:51] <JEEB> go grab the newest ffmpeg with libx264
[14:51] <MadWoodworker> From 'git'?
[14:51] <MadWoodworker> I did.
[14:51] <JEEB> are you *nix or win?
[14:51] <MadWoodworker> Linux - Ubuntu.
[14:51] <JEEB> with win you usually pick someone's builds
[14:52] <JEEB> on *nix you usually build yourself
[14:52] <JEEB> ok, so you have current libx264 and ffmpeg compiled and installed?
[14:53] <MadWoodworker> Yes.
[14:53] <JEEB> ok
[14:53] <JEEB> what audio do those things contain btw?
[14:53] <JEEB> I think you'll most probably want to just copy audio tracks over if they're not crazy in some way
[14:54] <MadWoodworker> I just want to copy audio.  They are 5.1.
[14:54] <JEEB> k
[14:54] <MadWoodworker> My problem, btw, is with the video.  There is far too much 'pixelation'.
[14:54] <JEEB> you were most probably using it wrong then, I'm making you a command line atm :P
[14:55] <MadWoodworker> Thanks!
[14:55] <JEEB> ffmpeg -i input.derp -acodec copy -vcodec libx264 -crf <the highest rate factor that still gives you good quality> -preset <a compression/speed setting> -tune <set a tuning for a certain kind of source> out.mkv
[14:55] <JEEB> generally like this
[14:56] <JEEB> you can leave the -preset and -tune alone for the time being (preset will be set to 'medium' by default and tunings are just extra)
[14:56] <JEEB> try encoding one of the clips you have with some crf, say, 24
[14:56] <JEEB> if it looks good
[14:56] <JEEB> you raise it
[14:56] <JEEB> and try again
[14:56] <JEEB> then when it stops looking good
[14:56] <JEEB> you go one lower
[14:57] <JEEB> and thus you have the rate factor that gives you the quality you need :)
[14:57] <JEEB> if it looks bad, you go down from 24
[14:57] <JEEB> until it looks good
[14:57] <MadWoodworker> How low can you go?
[14:57] <JEEB> quite low, but you usually don't need lower than 13-16, and these are quite special cases already
[14:57] <MadWoodworker> Perfect - I'll try that.
[14:57] <JEEB> don't
[14:57] <MadWoodworker> Thanks.
[14:58] <JEEB> if you set it way too low
[14:58] <JEEB> you won't get compression :P
[14:58] <MadWoodworker> Sorry - mean't I'll try your command line suggestion.
[14:58] <JEEB> not to mention that 13-16 isn't promising you good quality, it's just the lowest area I've gone so far if not for lossless (-crf 0 is lossless with 8bit H.264)
[14:59] <JEEB> if your source is super special you might have to go lower
[14:59] <JEEB> if it isn't special at all, you might be fine with 21-23
[14:59] <JEEB> the whole idea is to start up high
[14:59] <MadWoodworker> What does the 8-bit mean?
[15:00] <JEEB> precision
[15:00] <JEEB> don't care too much about it
[15:00] <JEEB> if you wanted 10bit H.264 you would've known about it and built a special libx264
[15:00] <MadWoodworker> Okay - thanks.
[15:00] <JEEB> so if you have good eyes, it might have a chance of looking bad -> 24 , if it looks good on the other hand you'd be getting rather alright'ish compression :)
[15:01] <MadWoodworker> Will the crf value affect the encoding speed?
[15:01] <JEEB> lower crf values can be slower because more data will be needed to be coded, but generally no -- it's not a switch that controls speed
[15:01] <JEEB> -preset controls speed
[15:01] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset
[15:02] <JEEB> basically I'm just having you find your sweet spot with crf first
[15:02] <JEEB> after that you can try different presets and select the slowest you can take
[15:03] <BobLfoot> I recently switched from Fedora14{ffmpeg-0.6} to Centos6{ffmpeg0.8} can found that my favorite libx264-slow.ffpreset no longer exists in /usr/share/ffmpeg.  I tried copying the 0.6 preset into my 0.8 directory but ffmpeg responds it can't find the preset, yet it finds libx264-lossless_slow.ffpreset just fine.  What am I missing the docs said it should search /usr/share/ffmpeg for preset files, and the variables listed in both are the same names, if not nearl
[15:03] <JEEB> BobLfoot, if the ffmpeg is new enough it should now use libx264's internal presets
[15:03] <JEEB> thus, the preset files are no longer needed
[15:04] <JEEB> -preset should take in the values mentioned here: http://mewiki.project357.com/wiki/X264_Settings#preset
[15:04] <BobLfoot> ffmpeg-0.8.7-50.el6 is what rpm/yum tells me I have not sure how new that is
[15:04] <JEEB> well, most probably new enough :)
[15:05] <JEEB> you can try by setting -preset to one of those values :)
[15:05] <JEEB> it shouldn't throw a warning/error at your face
[15:06] <BobLfoot> JEEB - didn't expect such a quick response, I've been up 20+ am gonna grab a nap and mark your link to visit when I wake thanks
[15:06] <JEEB> np
[15:12] <cpglsn> hi
[15:16] <cpglsn> hey guys, do you know what's going on here ? http://pastebin.mozilla.org/1389394
[15:22] <Mavrik> cpglsn, obviously the data in the .flv file isn't mp3
[15:22] <Mavrik> so if you just copy it to the mp3 file it won't work
[15:22] <cpglsn> Mavrik: isn't that the way to extract audio from video file ?
[15:23] <cpglsn> ffmpeg -i video.extension -acodec copy out.mp3
[15:23] <Mavrik> yes
[15:23] <Mavrik> but you're saving your audio to an mp3 container when it isn't mp3.
[15:23] <cpglsn> it does not convert it as well ?
[15:23] <Mavrik> no, because you said "COPY"
[15:24] <cpglsn> oh, you're right ... bad me
[15:24] <Mavrik> also it's written quite clearly:     Stream #0.1: Audio: aac, 44100 Hz, stereo, s16, 125 kb/s
[15:24] <Mavrik> try ffmpeg -i video.extension -acodec copy out.m4a
[15:24] <cpglsn> Mavrik: yes, sometime i'm really stupid, and the real problem is that some os those sometimes i understand it =(
[15:25] <cpglsn> Mavrik: yes thanks, but i'd like to use it in my mp3 reader, so i'll remove -acodec copy
[15:25] <Mavrik> yea
[15:25] <Mavrik> cpglsn, use something like ffmpeg -i video.extension -acodec libmp3lame -ab 128k out.mp3 instead
[15:25] <Mavrik> that'll convert audio from AAC to MP3
[15:26] <cpglsn> mmm, do i really need to specify the codec or it'll use libmp3lame by default? 
[15:27] <Mavrik> cpglsn, hmm... it should choose libmp3lame by default... however ffmpeg behaviour changess in between versions and can be sometimes erratic, so it's better to explicitly set bitrate and codec
[15:28] <cpglsn> understood, i'll keep in mind for nex conversions too. Thanks a lot Mavrik!
[15:30] <Mavrik> cpglsn, oh, and explicitly setting -vn should help as well (that's disable video)
[15:31] <cpglsn> cool, thanks
[15:46] <Nagy> how big is the erformance wise differance between 32 and 64 bit builds of ffmpeg?
[15:48] <Mavrik> meager iirc
[15:52] <sjuxax> Hello. It seems ffmpeg doesn't include presets for x264 anymore?
[15:53] <kcm1700> depends on the configuration.
[15:53] <kcm1700> ah
[15:53] <kcm1700> sorry, never mind my words
[15:53] <sjuxax> Nevermind, found the new dir in git. It doesn't seem to install with the package from my distro though. -vpre main complains that main can't be found.
[15:53] <kcm1700> misunderstood your sentence.
[15:53] <sjuxax> derp, actually, those are all vpx presets
[15:54] <Ave> the syntax has changed
[15:54] <Ave> whta worked before, doesnt anymore
[15:54] <sjuxax> Ave: OK. Do you have a doc or something that explains the changes I need to make?
[15:54] <Ave> nope but I had example commands somewhere ..
[15:54] <Mavrik> sjuxax, it's "-preset" now
[15:55] <Mavrik> sjuxax, and .ffprofile files aren't needed anymore, profiles are part of x264 now
[15:55] <Ave> but vpre still has its use!
[15:55] <Mavrik> sjuxax, http://mewiki.project357.com/wiki/X264_Settings#preset
[15:57] <sjuxax> Thanks Mavrik. How do I pass those options now?
[15:57] <JEEB> or more like, instead of using its own presets, ffmpeg now uses libx264's internal stuff :3
[15:57] <JEEB> sjuxax, -preset presetname
[15:57] <sjuxax> Just tried, got a complaint :\
[15:58] <JEEB> old ffmpeg?
[15:58] <JEEB> although... if it no longer has its own preset files it should be new enough...
[15:58] <JEEB> WhatIsThisIDontEven
[15:59] <sjuxax> JEEB, no, I think I got it sort of figure out now. :) Thanks
[16:02] <sjuxax> Heh, did audio opts change too? Can't seem to use libvorbis anymore...
[16:03] <Mavrik> hmm, check ffmpeg -vodecs
[16:03] <Mavrik> *codecs
[16:03] <Mavrik> it might be just "vorbis" now
[16:04] <sjuxax> Mavrik: libvorbis is still there
[16:04] <sjuxax> ffmpeg -threads 0 -i 00001.MTS -s 1024x576 -vcodec libx264 -profile main -preset slow -crf 25 -c:a libvorbis -aq 5 -ar 48000 out.mkv
[16:04] <sjuxax> that's my line
[16:15] <JEEB> sjuxax, any specific reason you set main profile?
[16:16] <sjuxax> JEEB: no
[16:16] <JEEB> then you don't need to set it, since it should be auto-set to the needed one depending on your other settings (preset here)
[16:17] <JEEB> profile limits the features you can use
[16:23] <sjuxax> OK, thanks JEEB. That also seemed to fix the complain about the audio thing? Weird. Now I get a segfault though :D
[16:23] <JEEB> lol
[16:24] <JEEB> oh right
[16:24] <JEEB> you specified threads 0 before -i
[16:24] <JEEB> before -i -> decoding threads (zero is for lulz, prolly crashes) , after -i -> encoding threads
[16:25] <sjuxax> hmm, ok, thanks, I'll try moving it
[16:25] <sjuxax> FYI argument order has been one of the things that has really tripped me up in the past when trying to use ffmpeg.
[16:40] <JEEB> sjuxax, I will guess it worked?
[16:40] <sjuxax> JEEB, Yes, that did work, thank you very much :)
[16:40] <osearth> arguments order never been an issue! ffmpeg shes perfect!!
[18:15] <kanzure> i used -ss and -t to get a clip from a file, but the last two seconds of the generated video sorta lag/pause, what am i doing wrong?
[18:16] <kanzure> i was using vcopy and acopy for video/audio codecs
[20:20] <pasteeater> kanzure: hard to say without knowing your command and the complete console output (use a pastebin service)
[20:35] <kanzure> pasteeater: ffmpeg -vcodec copy -acodec copy -i "$file" -ss 7 -t 15 "previews/$file"
[20:35] <kanzure> http://pastie.org/2945652
[20:36] <kanzure> actually, this doesn't matter i think
[20:36] <kanzure> the output file is incompatible with my phone anyway
[20:36] <kanzure> somehow ffmpeg is breaking the encoding
[20:39] <kanzure> huh vivienschilis is actually the person i should be talking with. doesn't vivien run pandastream?
[21:07] <pasteeater> kanzure: options before -i are generally applied to the input/decoder
[21:07] <pasteeater> move vcodec copy and acodec copy after -i "$file" 
[21:23] <kanzure> pasteeater: thanks, i'll try that
[21:23] <kanzure> does anyone know what video_tmp_noqt.mp4\nqt-faststart is?
[21:34] <kanzure> pasteeater: the problems at the end of the videos are gone, thanks
[21:53] <pasteeater> kanzure: qt-faststart is generally used if you put your video online. it moves some data to the beginning of a video file so the video will play before it's completely downloaded.
[22:11] <BobLfoot> {/part
[22:15] <kanzure> pasteeater: what would video_tmp_noqt.mp4 be?
[22:25] <pasteeater> kanzure: a name of a video file? i have no context, so i don't know.
[22:27] <kanzure> ok, guess it's not something standard
[23:01] <giany> any idea what this error means : [aac @ 0x9b27b00] Number of bands (54) exceeds limit (46).3 bitrate= 241.6kbits/s dup=0 drop=18100 
[23:01] <giany> i'm trying to convert from mp4 to flv
[23:03] <sacarasc> kanzure: I'd guess it's the video before qtfaststart has been run on it.
[23:10] <kanzure> sacarasc: two passes of encoding?
[23:11] <sacarasc> No, the file name just sounds like it's the end of encoding, but before qtfaststart was run on it.
[23:35] <giany> so no one has any idea what these messages mean : 
[23:35] <giany> Number of bands (54) exceeds limit (46)
[23:35] <giany> [aac @ 0x9f98000] TNS filter order 13 is greater than maximum 12.
[23:59] <Toux1> Hi all
[00:00] --- Thu Dec  1 2011