[Ffmpeg-devel-irc] ffmpeg.log.20120412

[burek]

[00:00] <Kingsy> burek: the latest version says Unrecognized option 'f'
[00:00] <burek> o.O
[00:00] <burek> can you please use pastebin.com, to show your command line and its output?
[00:01] <Kingsy> sure
[00:01] <Kingsy> burek: http://pastie.org/3770857
[00:02] <burek> :/
[00:02] <burek> try this one https://sites.google.com/site/linuxencoding/builds
[00:02] <burek> if that also doesnt work.. f it..
[00:03] <Kingsy> 1 sec
[00:03] <Kingsy> burek: ok thats a little different http://pastie.org/3770866
[00:04] <burek> hm
[00:04] <burek> it doesnt support alsa, we can try with oss
[00:04] <burek> let me see
[00:04] <burek> ./ffmpeg -f video4linux2 -r 30 -i /dev/video0 -f oss -ac 1 -i /dev/dsp -acodec aac -strict experimental -ab 128k -ac 1 -ar 44100 -b 1000k out.mp4
[00:05] <Kingsy> ok 1 sec
[00:05] <burek> or try /dev/dsp1, /dev/dsp2, ...
[00:05] <Kingsy> burek: naaa just [oss @ 0x1ee6820] /dev/dsp: No such file or directory
[00:05] <Kingsy> tried 1 and 2 too
[00:05] <burek> just a sec
[00:06] <burek> try apt-cache search oss
[00:06] <burek> open sound system - something like that
[00:07] <Kingsy> oss4-base ?
[00:07] <burek> let me check
[00:08] <burek> yes
[00:08] <burek> try that
[00:08] <burek> when you install it, it should create /dev/dsp
[00:11] <Kingsy> naaa it didnt..
[00:11] <Kingsy> screw it
[00:11] <burek> oss-compat
[00:11] <burek> try that
[00:11] <Kingsy> it was installed with oss-base
[00:11] <Kingsy> btw
[00:11] <burek> is there /dev/dsp
[00:12] <Kingsy> isnt there a delay you can add to the audio?
[00:12] <Kingsy> burek: no
[00:12] <burek> well you can
[00:12] <Kingsy> those vsync and async commands do nothing as far as I can see
[00:12] <burek> -itsoffset 2
[00:12] <burek> will shift stream for 2 seconds
[00:12] <burek> now, I don't remember do you put it
[00:13] <burek> before -i or after
[00:13] <burek> so try
[00:13] <Kingsy> don't you think it would be easier to just stitch the two streams together after recording?
[00:13] <burek> or that :)
[00:14] <Kingsy> -itsoffset 2 doesnt do anything either
[00:14] <burek> well
[00:14] <Kingsy> perhaps I should just stitch them after..
[00:15] <Kingsy> do you know of a decent editor that I could use? preferably something with a GUI ?
[00:15] <burek> ffmpeg -f video4linux2 -r 30 -i /dev/video0 -itsoffset 2 -f oss -ac 1 -i /dev/dsp -ab 128k -b 1000k out.mp4
[00:15] <burek> or
[00:15] <burek> ffmpeg -f video4linux2 -r 30 -i /dev/video0 -f oss -ac 1 -i /dev/dsp -itsoffset 2 -ab 128k -b 1000k out.mp4
[00:15] <burek> try those two
[00:15] <burek> and you'll see the difference
[00:15] <Kingsy> well the first one doesnt work because I don't have oss
[00:15] <burek> well, you get the idea :)
[00:16] <burek> either put it after first input or after second
[00:16] <burek> and then you can also use -itsoffset -2
[00:16] <burek> if you want to go to another way
[00:16] <Kingsy> ah I see
[00:16] <Kingsy> ok I will mess arund with that
[00:16] <burek> ok :)
[00:16] <burek> have fun :)
[00:19] <Kingsy> burek: thanks so much for your help with that
[00:19] <burek> :beer: :)
[00:30] <Kingsy> damn right.. you earned a beer :)
[01:39] <echelon> what causes this.. av_interleaved_write_frame(): Operation not permitted
[01:40] <echelon> it doesn't happen when i use the lossless_ultrafast preset
[01:40] <echelon> just when i use hq
[01:40] <echelon> for libx264
[01:41] <burek> I would suggest you to ask in #libav since I was thinking it's just a library usage support
[01:41] <burek> but I found out today it's a fork of ffmpeg
[01:41] <echelon> oh
[01:41] <burek> so I don't know.. google? :)
[01:41] <echelon> i did
[01:42] <burek> did you get that error by running ffmpeg cmd line tool
[01:42] <echelon> yes
[01:42] <burek> or using it programmatically
[01:42] <echelon> the ffmpeg tool
[01:42] <burek> can you please use pastebin.com, to show your command line and its output?
[01:45] <echelon> http://pastebin.com/6wfivCw3
[01:46] <burek> hmh
[01:46] <burek> at first look
[01:47] <burek> your ffmpeg looks very old
[01:47] <burek> a lot of features/bug fixes are being made each day
[01:47] <burek> so 6 months old ffmpeg is really a red alert
[01:48] <echelon> hmm ok
[01:48] <burek> you can try
[01:48] <burek> https://sites.google.com/site/linuxencoding/builds
[01:48] <burek> just download and run the same command
[01:48] <burek> to see if it will work
[01:48] <burek> if it does, then update your ffmpeg
[01:51] <echelon> Option 'croptop' has been removed, use the crop filter instead
[01:51] <echelon> how do i use that
[01:52] <burek> yes, you type something like -vf 'crop=x:y:z' let me find manual
[01:52] <burek> http://ffmpeg.org/ffmpeg.html#crop
[02:14] <echelon> i don't understand it
[02:15] <burek> what exactly
[02:15] <echelon> let's say i wanted to crop top 44 and crop bottom 44 on a video whose original resolution is 936x616
[02:15] <echelon> what would be the parameters for -vf crop
[02:16] <burek> crop=in_w-88:in_h
[02:16] <burek> no
[02:16] <burek> crop=in_w:in_h-88
[02:17] <Freakshow> w and h being width and height respectively?
[02:17] <burek> as written
[02:17] <echelon> -vf crop=out_w:out_h:x:y
[02:17] <burek> -vf 'crop=in_w:in_h-88'
[02:18] <echelon> ok
[02:18] <echelon> how does it tell it to from where to crop
[02:18] <burek> it crops from center if you dont specify 3rd and 4th param
[02:18] <burek> because it is the majority usage
[02:18] <echelon> mm ok
[02:19] <echelon> [crop @ 0x8085420] Invalid too big or non positive size for width '936' or height '528'
[02:19] <echelon> Error opening filters!
[02:19] <burek> can you please use pastebin.com, to show your command line and its output?
[02:21] <echelon> http://pastebin.com/Xy45ATfr
[02:21] <burek> that's not what I wrote
[02:22] <echelon> w:936, h:616-88
[02:22] <burek> also, if you want to crop your input
[02:22] <burek> and not output
[02:22] <burek> then move -vf before first -i
[02:23] <burek> you don't have to calc those values
[02:23] <burek> ffmpeg -vf 'crop=in_w:in_h-88' -i WMCap\[2\].mpg -t 1284 -an -vcodec libx264 -s 800x500 -aspect 16:10 -preset ultrafast video1.mp4
[02:24] <echelon> ^_^
[02:24] <burzum> hi
[02:27] <burzum> when im converting audio from a file that has no extension does ffmpeg automatically get the source format and knows how to process it right? so in other words does it matter if i use a wave file for input or a flac to convert both to mp3?
[02:27] <burek> it doesn't look at extensions at all
[02:28] <burek> it analyzes data
[02:28] <burzum> good, because for the output i had to specify -f if i was writing an extensionless file
[02:29] <burek> well yes :) how else would it know what you want :)
[02:29] <burzum> i thought -acodec was enough ;)
[02:29] <burek> well yes, but the same -acodec can be put into various formats (muxers)
[02:29] <burek> mp4, mkv, flv, mp3, ...
[02:30] <burek> http://uk.answers.yahoo.com/question/index?qid=20100808145033AADpyVX
[02:32] <burzum> thanks im aware of that i just didnt know about that ffmpeg makes a difference here and was throwing errors when i have not used the -f param or whatever exactly it was doing, i dont remember
[02:32] <burek> yes, because -acodec mp3 can be put into mp3 container, or avi, or flv, etc..
[02:33] <burek> and when it's about input then ffmpeg loads first part of the file in the memory and analyzes it
[02:33] <burek> to see what is it actually
[02:34] <burzum> but two more things i could not really figure out (i havent had to deal with audio stuff before...): how can i set a certain audiobitate for ogg vorbis? mp3 is using ab 256k for example but -aq and a value from 0 to 100 is used for ogg. how can i make it accept also 256k for example? i read that the whole thing is using a dynamic bitrate but i would like to somehow figure out whats the best
[02:34] <burzum> quality/compression and preferable set it in like i do it for mp3s
[02:35] <burzum> my use case is that i have wav files, convert them to flac which become the master file from which i create ogg and mp3
[02:35] <burek> -ab 256k will generally set audio quality
[02:35] <burek> no matter is it mp3 or ogg
[02:35] <burek> unless ogg is a codec that doesn't deal with bitrates at all, just quality indicator
[02:35] <burek> but that's high unlikely
[02:35] <burek> highly
[02:36] <burek> burzum, did you try aac+ ? :)
[02:37] <burek> it compresses several times more than mp3 with better quality
[02:37] <burek> meaning you get smaller files with better quality than mp3
[02:37] <burzum> looks like you have the experience, so how would you create a mp3 file that has a very high quality? go for 256k (from what i read 192k should be already good enough) or use vbr? if vbr how do i get ffmpeg to write a vbr mp3?
[02:37] <burzum> ive heard about aa+ but didnt know that
[02:38] <burek> well, are you on windows
[02:38] <burzum> yes
[02:38] <burzum> well, yes and no
[02:38] <burek> download NeroAACEnc
[02:38] <burek> and try
[02:38] <burzum> i develop on windows, the final app is going to run on a nix system
[02:38] <burek> read the help for parameters and try to convert one file to see the result
[02:38] <burzum> would that aa+ in an mp3 container play well on all hardware players to?
[02:38] <burek> but make it aac+ (a.k.a. HE-AAC v2)
[02:39] <burek> it depends on hardware
[02:39] <burek> old hardware will probably fail
[02:39] <burek> but all the new hardware is already aac+ aware
[02:40] <burzum> the goal is to produce files that are playable on most hardware players but also to not have to produce 10 formats because of keeping the storage low. i do not want to have thousands of files for a single song eating up hundreds of mb
[02:40] <burek> also, for vbr mp3, use quality selector in ffmpeg and not bitrate selector
[02:41] <burek> mp3 is mostly supported
[02:41] <burek> but is very old too
[02:41] <burzum> i know, thats why ive asked about the aa+ compatibility with hardware players
[02:41] <burek> so don't expect it to cover all your goals
[02:41] <burek> aac+
[02:42] <burzum> so far i plan to offer flac, ogg vorbis and mp3
[02:42] <burek> well, if you offer flac
[02:42] <burzum> this should satisfy all kinds of possible customers
[02:42] <burek> then your chances are better with aac+ because it's older than flac
[02:42] <burzum> but less known or?
[02:43] <burek> AAC is from 1997
[02:43] <burek> FLAC started around 2000 or 2001
[02:44] <burzum> yes, but its not about the year but about the public perception
[02:44] <burek> for that, I'm afraid I'm not competent :)
[02:44] <burzum> even if aac would be from 1900 but people and hardware know it less well than flac the choice is clear
[02:45] <burzum> i personally would always go for bleeding edge technology ;)
[02:45] <KaosTheory-> good evening
[02:45] <KaosTheory-> any devs here at all?
[02:46] <burzum> the way the code is written its not a problem to reprocess the audio later into a new format
[02:46] <KaosTheory-> or anyone who can help me with a problem with ffmpeg
[02:46] <burek> KaosTheory-, what's the problem you have?
[02:46] <burek> burzum, I agree
[02:47] <burzum> i think i simply put the aac+ stuff in, 5min coding task
[02:47] <KaosTheory-> I have a 1980x1080 avi file which i am trying to convert using ffmpeg i type the necessar command line (basic -i <file><File>)
[02:47] <KaosTheory-> however ffmpeg just hangs and no encoding commenses
[02:48] <burek> can you please use pastebin.com, to show your command line and its output?
[02:48] <KaosTheory-> two secs
[02:48] <burek> make it three :)
[02:48] <burek> it's on the house :)
[02:49] <KaosTheory-> :P
[02:50] <burek> can you please paste it here
[02:50] <burek> because the channel is being logged
[02:50] <burek> for others to read
[02:50] <KaosTheory-> ah nps
[02:50] <KaosTheory-> http://pastebin.com/n1FpfX0x
[02:50] <burek> hm
[02:50] <burek> and then ffmpeg just hangs there?
[02:50] <KaosTheory-> yeah
[02:51] <KaosTheory-> does nothing
[02:51] <burek> seems like a bug
[02:51] <KaosTheory-> get the win7 'program not responding' popup
[02:52] <KaosTheory-> so its not me doing something stupid then
[02:52] <burek> KaosTheory-, do you know how to use ftp
[02:52] <burek> and is that video some private video
[02:52] <burek> or you can upload it as a sample
[02:52] <burek> so that we can fix the bug
[02:52] <KaosTheory-> nah just a Tribes Ascend SunStar capping route
[02:52] <burek> is it big
[02:53] <KaosTheory-> 2.5gb
[02:53] <burek> hm
[02:53] <KaosTheory-> wonders of using DxTory and the Lagarith Lossless Codec
[02:53] <burek> would it be a problem for you to split the first 10 mb of the file
[02:53] <burek> and upload it?
[02:54] <KaosTheory-> i think you would get about 3 frames of it
[02:54] <KaosTheory-> lol
[02:54] <KaosTheory-> oh
[02:54] <burek> :)))
[02:54] <KaosTheory-> is it worth noting that if i run it through vDub the file works for encoding
[02:55] <KaosTheory-> well it did for a 50% down scale
[02:55] <burek> what did you do in vdub
[02:56] <KaosTheory-> just recoded it with zero codec
[02:56] <KaosTheory-> well resaved the avi
[02:56] <KaosTheory-> with no codec
[02:56] <KaosTheory-> with a 50% size filter on it
[02:56] <KaosTheory-> and try it now with 100% scale
[02:56] <burek> can you run just ffmpeg -i SunStar.avi
[02:56] <burek> to see what's the output
[02:56] <KaosTheory-> i can in a sec
[02:57] <KaosTheory-> saving a seperate avi incase its the Lagarith Codec causing the issue
[02:57] <KaosTheory-> ... for some stupid reason
[02:57] <KaosTheory-> 80%
[02:57] <burek> ok
[02:58] <KaosTheory-> well there goes my hdd space
[02:58] <KaosTheory-> 9.2gb so far
[02:58] <burek> :)
[02:58] <burzum> burzum do you know the acodec name for aac+ that ffmpeg is using?
[02:58] <burzum> burek do you know the acodec name for aac+ that ffmpeg is using?
[02:58] <KaosTheory-> ok that worked at full size
[02:59] <burek> -acodec libaacplus
[02:59] <KaosTheory-> perhaps codec problem?
[02:59] <burzum> thank you!
[02:59] <burek> burzum, http://ffmpeg.gusari.org/viewtopic.php?f=25&t=37
[03:00] <burek> KaosTheory-, just run ffmpeg -i SunStar.avi
[03:00] <burzum> thanks
[03:00] <burek> to see what ffmpeg detected
[03:00] <KaosTheory-> burek same thing as before
[03:00] <burek> hm
[03:01] <burek> it would be great if you could cut first and last 10mb of that file KaosTheory-
[03:01] <KaosTheory-> to note though: vdub re-coded with Uncompressed RGB/YCbCr allows the video to be encoded
[03:01] <burek> so that we can investigate what's wrong
[03:01] <KaosTheory-> ok let me see if i can do that with vDub
[03:01] <burek> I know, as soon as you change video stream, it gets fixed
[03:01] <burek> just a second
[03:01] <burek> not vdub
[03:02] <KaosTheory-> ok..
[03:02] <burek> I'm looking into copy dos command
[03:03] <burek> it has got some param
[03:03] <burek> that says how many bytes to copy
[03:03] <burek> just a sec
[03:03] <KaosTheory-> not sure how i will cut the first and last 10mb
[03:03] <burzum> burek so -aq vs -ab is that ab will get me a fixed bitrate while -aq will create a vbr file with a given average quality (0-100)?
[03:04] <burzum> if the codec supports it
[03:04] <burek> KaosTheory-, http://www.virtualobjectives.com.au/utilitiesprogs/partcopy.htm :)
[03:04] <burzum> and if i do not specify -ar the target file will get the same -ar as the input file?
[03:05] <burek> burzum, yes
[03:05] <burzum> yes for both? cool
[03:05] <burek> yes :)
[03:06] <burek> -aq says something like "I want constant quality"
[03:06] <burzum> \o/
[03:06] <burek> where -ab on the other hand says "I want constant bit rate"
[03:06] <burzum> yea, im going to validate that before i'll check the file into my storage system by parsing the ffmpeg info about that file
[03:07] <KaosTheory-> ok part copy made
[03:07] <KaosTheory-> well start bit anyway
[03:07] <burek> KaosTheory-, http://ffmpeg.org/bugreports.html read the bottom text
[03:07] <burzum> it would be a wet dream if ffmpeg would output json or xml
[03:07] <burek> so just upload SunStar.avi_begin and SunStar.avi_end
[03:08] <burek> with a third (text) file which will contain your pastebin text
[03:08] <burek> so that we know what is what
[03:08] <burek> ok ?
[03:08] <burek> burzum, why do you need json or xml
[03:09] <burzum> easier to parse than regexing the whole output
[03:09] <burek> searching for what
[03:09] <burzum> if you do not know ALL error messages you could still simply check if there are child elements in the <error> node for example and simply get its message
[03:10] <burek> well, when you start ffmpeg, you check it's return value
[03:10] <burek> if it is not 0 then there were errors :)
[03:10] <burek> if that's what you want :)
[03:11] <burzum> 2>&1
[03:11] <burek> and if there were errors, display the entire log to the user
[03:11] <burek> and let him decide what to do :)
[03:11] <burzum> i do not want to show an error log to a user of the web frontend :)
[03:11] <burek> then log to a file, for yourself
[03:11] <KaosTheory-> oh
[03:11] <burzum> just a simple message "Failed because blah blah"
[03:12] <KaosTheory-> do you defo want that format burek?
[03:12] <burek> defo?
[03:12] <KaosTheory-> i have uploaded as SunStar-Start.avi
[03:12] <burek> oh, ok :)
[03:12] <burek> name doesn't matter that much
[03:12] <KaosTheory-> definetly = defo
[03:12] <burek> just make it so that we can understand :)
[03:12] <KaosTheory-> easy ;)
[03:13] <KaosTheory-> where do you want the link?
[03:13] <KaosTheory-> here im guessing
[03:13] <burek> if you want to report a bug yourself, go here: https://ffmpeg.org/trac/ffmpeg
[03:13] <burek> if not, just leave the link here
[03:13] <burek> I'll create a bug report
[03:13] <burek> :)
[03:14] <KaosTheory-> http://www.digitallyhazardous.com/kaos/ff/
[03:14] <burek> oh :)
[03:14] <burek> I thought you read the page
[03:14] <burek> that I gave you
[03:14] <burek> D
[03:15] <burek> :D
[03:15] <burek> "KaosTheory-, http://ffmpeg.org/bugreports.html read the bottom text"
[03:15] <burek> it should have the effect like: "Upload the sample to the MPlayer FTP server."
[03:15] <burek> :)
[03:15] <KaosTheory-> oh
[03:15] <KaosTheory-> lol
[03:16] <burek> ^^
[03:16] <burek> do you know how to do it :)
[03:18] <KaosTheory-> uploading now
[03:19] <burek> thanks :)
[03:19] <KaosTheory-> uploaded
[03:20] <burek> great :)
[03:20] <burek> now, do you want to submit a new bug report
[03:20] <burek> or would you like me to do it?
[03:20] <KaosTheory-> if you dont mind
[03:20] <burek> not at all
[03:20] <KaosTheory-> i have enough trouble remembering all the accounts and passwords i have as it is
[03:20] <burek> are the names the same
[03:21] <burek> file names
[03:21] <KaosTheory-> SunStar-Start.avi SunStar-End.avi Error.txt
[03:21] <burek> ok
[03:22] <burek> https://ffmpeg.org/trac/ffmpeg/ticket/1192
[03:22] <burek> bookmark it so you know when it is resolved if you like :)
[03:23] <KaosTheory-> will do
[03:23] <KaosTheory-> its probably something that the codec doesnt do
[03:23] <KaosTheory-> or misses out that ff isnt taking into account
[03:24] <KaosTheory-> as a fully uncompressed 1980x1080 file works fine
[03:24] <burek> well, as much as I can se
[03:24] <burek> see
[03:24] <burek> it can't even read file normally
[03:24] <burek> (container)
[03:25] <burek> so it didn't even come to the point to show what streams are inside the container
[03:25] <KaosTheory-> hmmm
[03:25] <burek> if vdub has an option
[03:25] <burek> to just copy codecs (audio/video)
[03:25] <burek> and just change the format (muxer)
[03:25] <burek> then that could help
[03:26] <KaosTheory-> you mean video compression audio compression?
[03:26] <KaosTheory-> i can select Largarith codec in vDub
[03:26] <burek> format is like an envelope in which are two other envelopes (audio and video)
[03:26] <burek> so, if you just change the format and leave av streams intact
[03:27] <burek> that might help ffmpeg recognize stuff
[03:27] <KaosTheory-> ok....
[03:27] <KaosTheory-> sorry its been a while since i did any video editing and stuff
[03:27] <KaosTheory-> so im a total noob to it all again
[03:28] <burek> it's ok :)
[03:28] <burek> noone has been born skilled :)
[03:28] <KaosTheory-> unless you happen to be my m8
[03:28] <KaosTheory-> the crap he does and programs just baffles me
[03:28] <burek> :)
[03:29] <KaosTheory-> on a side notw
[03:29] <KaosTheory-> note*
[03:29] <burek> does he know how to make beer? :)
[03:29] <KaosTheory-> umm
[03:29] <KaosTheory-> computer wizz
[03:30] <KaosTheory-> not so much alchemical wizz ;)
[03:30] <burek> mh :) ok :) it was worth to try asking :))
[03:30] <KaosTheory-> side note: ffmpeg did work on a 1270x720 recording using Lagarith Lossless
[03:31] <burzum> burek i know somebody who makes homebrewed beer in a good quality, need some advice? ;)
[03:31] <burek> well, the thing here is that
[03:31] <burek> ffmpeg didn't get to "unpack" the avi format, to read what streams are inside
[03:31] <burek> (one of those streams is your Lagarith)
[03:31] <burek> so I doubt the codec is a problem, it is more likely the damaged avi format or something
[03:31] <burek> burzum, I need m8 that makes beer :D
[03:31] <KaosTheory-> dunno
[03:32] <KaosTheory-> i can only go with what i see, have experienced
[03:32] <burzum> haha, who doesnt? :)
[03:32] <KaosTheory-> burek i have better. i have mates that buy me beer :D
[03:32] <burek> KaosTheory-, did you manage to play that file in vlc
[03:32] <burek> or that :D
[03:32] <KaosTheory-> the vid works fine in every player
[03:32] <KaosTheory-> well Windows Media at least
[03:33] <burek> can you open vlc and go to tools - messages
[03:33] <KaosTheory-> it gitters
[03:33] <burek> set the verbosity level to 2
[03:33] <burek> and then play that file
[03:33] <KaosTheory-> but im guessin thats down to it being 2.5gb
[03:33] <burek> just until it starts
[03:33] <burek> then stop it and copy the log
[03:33] <KaosTheory-> ok hang on
[03:33] <iive> KaosTheory-: you may have already said that, but I'd ask anyway.
[03:34] <KaosTheory-> downloading vlc
[03:34] <iive> did you make the file on your own, or have you gotten it from somewhere else?
[03:35] <KaosTheory-> iive the vid was grabbed using Dxtory
[03:36] <iive> is it dv pro capture?
[03:37] <KaosTheory-> burek http://pastebin.com/yyR8NDQT
[03:37] <KaosTheory-> iive please elaborate
[03:38] <iive> i see it is dx/opengl capture program, like fraps.
[03:38] <KaosTheory-> yeah
[03:38] <iive> do you know what codec does it use?
[03:38] <KaosTheory-> i am using the Lagarith Lossless Codec
[03:38] <KaosTheory-> for a compramise between quality and HDD space
[03:39] <KaosTheory-> not to mention to retain fps within game
[03:39] <iive> i think i've heard that one. wasn't it lossless?
[03:39] <burzum> burek if i use -aq 15 or -aq 85 it doesnt matter, the mp3 always comes out with 96kbps? why?
[03:39] <KaosTheory-> yeah as far as i understand it, and as my mate told me, its basically a huffmans codec
[03:40] <iive> huffman is also lossless :)
[03:40] <KaosTheory-> ;)
[03:40] <burek> KaosTheory-, I updated your bug report with that VLC output :) https://ffmpeg.org/trac/ffmpeg/ticket/1192
[03:40] <iive> can you try to make few more files.
[03:40] <KaosTheory-> i already have loads
[03:41] <KaosTheory-> none of the 1980x1080 files work
[03:41] <iive> what is the biggest file that works?
[03:41] <burek> just curious, are all those that don't work, bigger than 2 gb
[03:41] <KaosTheory-> been racking my brains on this for two days
[03:41] <KaosTheory-> ummm
[03:41] <KaosTheory-> sec
[03:43] <KaosTheory-> ok resolution it worked on 1280x720
[03:44] <KaosTheory-> and no just tested a 113,453kb file and it is the same as before
[03:45] <KaosTheory-> lol
[03:46] <KaosTheory-> not even a 1280x720 wants to work now :/
[03:46] <burek> same error?
[03:46] <KaosTheory-> same 'freezing'
[03:46] <iive> KaosTheory-: the resolution is not important. Focus on the file size
[03:46] <burek> with "non-interleaved AVI" message?
[03:46] <KaosTheory-> yeah
[03:47] <iive> can you try making even smaller files? 10mb one?
[03:47] <burek> try typing ffmpeg -i filename
[03:47] <KaosTheory-> same
[03:47] <burek> and if you find a small one (10 mb)
[03:47] <burek> can you upload it
[03:48] <iive> KaosTheory-: does the program have auto split/segment functionality?
[03:49] <KaosTheory-> lol i double tapped the record button. 53mb
[03:49] <KaosTheory-> lol
[03:49] <KaosTheory-> same error on that
[03:49] <iive> good enough :)
[03:49] <iive> even on that!
[03:49] <KaosTheory-> yep
[03:50] <iive> try smaller resolution.
[03:50] <burek> omg :)
[03:50] <KaosTheory-> that was using 1280x720
[03:50] <iive> computers should still be capable of 320x200 :P
[03:50] <burzum> if i use -aq 15 or -aq 85 it doesnt matter, the mp3 always comes out with 96kbps. am i using it wrong?
[03:50] <KaosTheory-> lolololol
[03:51] <burek> burzum,
[03:51] <burek> can you please use pastebin.com, to show your command line and its output?
[03:51] <burzum> burek ok
[03:51] <KaosTheory-> 800x600 smallest resolution. 36mb. same 'freeze'
[03:52] <KaosTheory-> need drink and food brb
[03:52] <iive> burzum: aq may be algorithmic quality. try -ab 128k
[03:53] <burek> he wants vbr
[03:53] <burzum> burek http://pastebin.com/uXce5VZ7
[03:54] <burek> burzum, try using -q instead of -aq
[03:55] <burzum> 'ffmpeg: unrecognized option \'-q\'',
[03:55] <burek> ok :D wait :)
[03:55] <burzum> ive already looked at the documentation :)
[03:56] <burek> burzum, your ffmpeg is really old..
[03:56] <burek> and i mean really old
[03:56] <burzum> ok
[03:56] <burek> like 3 years ago
[03:56] <burek> or more
[03:56] <burzum> where do i get a new windows build that is reliable and has liblame, flac and all the stuff i already use?
[03:57] <burzum> ok just found that
[03:57] <burzum> hmm looks like the page ive got the current one from
[03:57] <KaosTheory-> right im back
[03:57] <burek> http://ffmpeg.zeranoe.com/builds/
[03:58] <iive> try with "compression_level"
[03:59] <iive> that one is for algorithmic quality.
[03:59] <KaosTheory-> iive is that to me or burzum?
[03:59] <iive> burzum:
[03:59] <KaosTheory-> k
[03:59] <burzum> ok
[03:59] <KaosTheory-> just checking
[04:00] <iive> burzum:  try also "global_quality", this one must be for VBR.
[04:01] <burzum> compression_level doenst work, 64k, always
[04:03] <iive> burzum: add qscale to the mix. it sets the flag.
[04:04] <burek> burzum, also try -aq 2
[04:04] <iive> so, you need qscale, to enable it, global_quality to set the level.
[04:05] <burek> http://ffmpeg.org/pipermail/ffmpeg-devel/2011-April/110717.html
[04:05] <burek> ffmpeg -i a.wav -aq 2 b.mp3 "corresponds to" lame -V2 a.wav b.m3
[04:05] <burek> and if you take a look here: http://wiki.hydrogenaudio.org/index.php?title=LAME
[04:05] <burek> -V0 (~245 kbps), -V1 (~225 kbps), -V2 (~190 kbps) or -V3 (~175 kbps)
[04:06] <burek> so -aq 0 will give 245 kbps, -aq 1 = 225, -aq 2 = 190 ...
[04:07] <iive> KaosTheory-: btw, first I asked you about DV codec, because avi handles DV in a special way. But that's not the case.
[04:07] <KaosTheory-> ok
[04:08] <iive> KaosTheory-: then I asked you about size, because avi files above 1GB (or max 2GB) need a special index to work. something called ODML2.0
[04:08] <iive> but with 53MB file, this doesn't seem to be case either.
[04:08] <KaosTheory-> ok
[04:08] <KaosTheory-> yeah still 'froze'
[04:08] <KaosTheory-> exact same log as before
[04:09] <iive> so now, you just have to make the smallest files you can do, and upload it to the bug tracker, so active developers can try to reproduce the problem on their own
[04:09] <KaosTheory-> tried serveral resolutions of the game, same issue
[04:09] <KaosTheory-> i have uploaded a start and end of the vid
[04:09] <burek> burzum, use this table for your -aq <param> http://wiki.hydrogenaudio.org/index.php?title=LAME#VBR_.28variable_bitrate.29_settings
[04:09] <KaosTheory-> 10mb in size
[04:10] <KaosTheory-> with error log
[04:10] <KaosTheory-> its on the ftp already
[04:10] <iive> KaosTheory-: it may be better to have a whole file.
[04:10] <burek> KaosTheory-, if you could upload a complete that would be the best, that's why we are asking you if you can produce ~10mb file :)
[04:10] <burek> complete file* :)
[04:11] <KaosTheory-> lol
[04:11] <KaosTheory-> i couldnt
[04:11] <burzum> burek thanks again :) already looking at it but now ive got confused by ARB vs VBR... im trying to figure out what will give me the best audio quality while keeping the file small.
[04:11] <KaosTheory-> smallest i could get with a doublt tap of the record button was 53mb
[04:12] <burek> burzum, those are always the opposite requirements.. you want it to be the smallest size, the best quality and the most widely supported :)
[04:12] <burek> those 3 never go along :)
[04:12] <iive> KaosTheory-: try the same, with even smaller resolution
[04:13] <burzum> i prefer better audio quality over filesize but im trying to figure out an economic value
[04:13] <iive> KaosTheory-: but if you can't 53MB is ok for an ftp.
[04:13] <iive> well.
[04:13] <iive> have fun. n8 ppl
[04:14] <burek> o/
[04:14] <burzum> burek i guess given my requirements i would go for CBR and need to decide which one i want to use
[04:14] <burek> why cbr
[04:14] <burzum> constant bitrate
[04:14] <burek> i just gave you the table for vbr
[04:14] <burek> did you look at it?
[04:15] <burzum> well but you said: you want it to be the smallest size, the best quality and the most widely supported :)
[04:15] <burzum> yes
[04:15] <burek> why are you talking about cbr then :)
[04:16] <burzum> as I said im trying to figure out the best compromise of filesize, quality
[04:16] <burzum> crap, its 04:14 here... i had the hope this would be easy to figure out
[04:16] <burzum> i need to get some sleep :s the night is going to be a short one
[04:16] <burek> ok :)
[04:17] <burek> have a good sleep
[04:17] <burek> tomorrow might give you the answers :)
[04:17] <burzum> burek thanks a lot for your great support
[04:17] <burek> :beer: :)
[04:17] <burzum> coffee! i dont drink alcohol during the work week :)
[04:17] <KaosTheory-> perfect
[04:17] <KaosTheory-> managed to get a 5mb file
[04:17] <burek> :)
[04:18] <KaosTheory-> my finger hurts now. :/
[04:18] <KaosTheory-> thats was a quick dbl tap. lol
[04:18] <burek> :D
[04:19] <KaosTheory-> DryDock.avi
[04:19] <KaosTheory-> uploading to ftp now
[04:19] <burek> thanks :)
[04:20] <KaosTheory-> done
[04:20] <burek> does it also give "non-interleaved AVI"
[04:22] <KaosTheory-> lol
[04:22] <KaosTheory-> that worked
[04:22] <burek> what
[04:22] <KaosTheory-> it encoded it
[04:22] <burek> well we need the smallest that still produces that err msg
[04:22] <burek> :)
[04:22] <KaosTheory-> urgh
[04:23] <KaosTheory-> man thats gonna be an ass to find
[04:23] <KaosTheory-> ok hang on
[04:23] <burek> :)
[04:23] <burek> just try "ffmpeg -i filename" on some small files that you already have
[04:24] <KaosTheory-> LOLOLOLOL
[04:24] <KaosTheory-> that would be why
[04:25] <KaosTheory-> wasnt using the right codec
[04:25] <KaosTheory-> hang on ;)
[04:26] <KaosTheory-> nope still cant pull it lower than that 53mb
[04:26] <KaosTheory-> and still wont encode it
[04:27] <burek> well, just check if
[04:27] <KaosTheory-> it must be something to do with the Lagarith Codec
[04:27] <burek> ffmpeg -i
[04:27] <burek> still gives that error
[04:27] <KaosTheory-> the Cinepack worked fine
[04:27] <burek> if it does, upload that 53mb file
[04:27] <burek> we'll note that we couldn't produce any smaller file
[04:27] <burek> :)
[04:27] <KaosTheory-> nope same
[04:28] <KaosTheory-> deleted the 53mb one
[04:28] <KaosTheory-> smallest i got now is the 113mb one
[04:28] <KaosTheory-> is that ok?
[04:28] <KaosTheory-> oh sweet
[04:28] <KaosTheory-> got a 59mb one
[04:28] <burek> :)
[04:29] <KaosTheory-> lol
[04:29] <KaosTheory-> thats 2 seconds worth of video
[04:29] <burek> they will look into binary data
[04:29] <burek> not the images itself
[04:29] <KaosTheory-> i know ;)
[04:30] <KaosTheory-> ok
[04:30] <KaosTheory-> TribesAscend.avi
[04:30] <KaosTheory-> 59mb being uploaded
[04:30] <KaosTheory-> the DryDock.avi can be scrapped
[04:30] <KaosTheory-> unless you want it for evidence that its not DxTory causing the issue
[04:30] <burek> ok, so that is the smallest that produces that non-interleaved AVI error msg
[04:30] <KaosTheory-> yeah
[04:30] <burek> ok
[04:30] <KaosTheory-> smallest i can get
[04:31] <KaosTheory-> want to try and keep the 1920x1080 resolution incase that has some random effect on it
[04:31] <KaosTheory-> i know the likely hood is very very slim
[04:31] <KaosTheory-> but just to ensure continuity
[04:31] <burek> is this ok? https://ffmpeg.org/trac/ffmpeg/ticket/1192
[04:32] <KaosTheory-> yeah perfect
[04:32] <burek> ok :) you have rss link at the bottom
[04:32] <burek> so you can get instant info
[04:32] <burek> when it is resolved :)
[04:33] <KaosTheory-> i will just check daily ;)
[04:33] <burek> or that :)
[04:33] <KaosTheory-> many thanks to you for your help burek
[04:33] <burek> :beer: :)
[04:34] <KaosTheory-> i dont mind even if its the codec
[04:34] <KaosTheory-> just so long as i know
[04:34] <KaosTheory-> can then try and find a codec that works ok
[04:34] <burek> well, I think it's a glitch in matrix, which will be fixed :)
[04:34] <KaosTheory-> ok thats awesome
[13:22] <JamesJJ> OK... still struggling with a flv with G711 audio. the problem is that ffmpeg is seeing the audio track but doesn't reconise the codec - I can force it to use the right codec by adding -acodec pcm_mulaw, but it is also getting the frequency wrong - Stream #0.1: Audio: pcm_mulaw, 5512 Hz, mono, s16, 44 kb/s - is there a way to change the input frequency, as I as the input data is 8000 not 5500?
[13:24] <burek> can you please use pastebin.com, to show your command line and its output?
[13:31] <burek> JamesJJ
[13:48] <hjt> :-&
[14:34] <darkb0e> hello :)
[14:36] <hjt> LOL
[14:36] <hjt> :-&
[14:36] <hjt> LOL
[14:45] <fling> hjt: :3
[14:58] <darkb0e> do you know any tutorial for streaming h264/mp3 to a flash video container?
[14:58] <darkb0e> anyone/
[14:58] <darkb0e> ?
[14:58] <darkb0e> i am trying with ffmpeg -> ffserver but with no luck
[14:59] <burek> streaming using what protocol
[15:00] <darkb0e> i am trying the 3 default that are in /etc/ffserver.conf
[15:00] <burek> well what is your goal
[15:00] <darkb0e> mpeg1 swf, ans asf
[15:00] <darkb0e> just for testing
[15:01] <burek> ffmpeg -i input.avi -vcodec libx264 -acodec libmp3lame -f mpegts udp://remoteserver:port
[15:01] <darkb0e> i want to transcode with ffmpeg  vcodec x264 and acodec mp3 to the ffserver, then ffserver will stream to http flv
[15:02] <burek> oh
[15:02] <burek> then use ffmpeg -i input.avi http://localhost:8090/feed1.ffm
[15:03] <burek> and all the rest, just setup in the conf file
[15:03] <darkb0e> w8 to paste bin my command and config
[15:04] <burek> and also, read this thoroughly http://ffmpeg.org/ffserver.html
[15:06] <darkb0e> http://pastebin.com/CfusGqRM
[15:07] <darkb0e> i read the documentation
[15:07] <burek> read my lines above
[15:09] <burek> also, using udp input, you'll have a lot of trouble with sync-ing
[15:10] <darkb0e> i was using vlc
[15:10] <darkb0e> but, it crashes like hell
[15:10] <darkb0e> and now i am trying to find alternative
[15:15] <darkb0e> burek: you said that i need to run ffmpeg twice? or you gave me 2 examples of ffmpeg/
[15:15] <darkb0e> ?
[15:16] <darkb0e> i am running first the ffserver with the config mentioned in the paste bin, and then i run ffmpeg with the arguments mentioned in the pastebin
[15:28] <darkb0e> burek: are you here?
[15:54] <fenrig> Hi I want to know how much I B and P frames there are in a certain video file, so I was wondering if I could do this with ffmpeg
[15:57] Last message repeated 1 time(s).
[16:02] <JamesJJ> Sorry for the delay - pasetbin on trying to use FLV with g711
[16:02] <JamesJJ> http://pastebin.com/3PTK5YA6
[16:03] <JamesJJ> I guess the important bit is from the flv demuxer - flv @ 0x148ac00]Unsupported audio codec (8)
[16:06] <JamesJJ> G711 has been supported by flash and FLV for a while now, if there was a way to override the frequency of the input stream I would be able to get round this
[16:10] <burek> darkb0e, no, just the 2nd ffmpeg line
[16:11] <burek> fenrig, use ffprobe?
[16:11] <darkb0e> ok, i managed to start it, but now i run into trouble with x264 presets :/
[16:12] <darkb0e> anyone with any link? with valid video sizes etc?
[16:12] <fenrig> burek: THANK YOU i'll look into it
[16:12] <darkb0e> thank you burek
[16:12] <burek> JamesJJ, your ffmpeg is way to old
[16:12] <burek> 3 years or more..
[16:12] <darkb0e> burek: thank you
[16:12] <burek> :beer: :)
[16:12] <burek> darkb0e,
[16:12] <burek> can you please use pastebin.com, to show your command line and its output?
[16:13] <JamesJJ> ah ok... more fool me for trying it on windows - I'll try a copy from the repo
[16:13] <darkb0e> ofc
[16:13] <burek> JamesJJ, if you use ffmpeg on windows
[16:16] <darkb0e> burek: http://pastebin.com/WaSuDyTQ
[16:16] <burek> where is the error msg?
[16:17] <darkb0e> no error, just doesn't start to show video on vlc
[16:17] <darkb0e> seems like buffering forever
[16:17] <burek> what url did you used to start the stream
[16:17] <darkb0e> http://<server>:8090/test.flv
[16:18] <darkb0e> it shows up in the stats page
[16:18] <darkb0e> but it says wait_feed
[16:18] <darkb0e> although the counter up'ing
[16:18] <burek> are you using latest vlc/ffmpeg
[16:20] <darkb0e> yeah compiled from wiki's guide
[16:20] <darkb0e> 4 times today :)
[16:20] <burek> which wiki
[16:20] <darkb0e> ffmpeg's wiki
[16:20] <darkb0e> ubuntu compilation
[16:20] <burek> do you happen to have url for that
[16:20] <darkb0e> ofc
[16:21] <darkb0e> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[16:21] <ckb> okay guys. 3 renditions from source: 1200Kb/s, 700Kb/s, 400Kb/s. Would it make a difference encoding in a linear route? (Encoding the 700 from the 1200 rather than src, and encoding the 400 from 700 rather than src)
[16:22] <burek> darkb0e, ok, that's ok
[16:22] <burek> now try opening another console
[16:22] <burek> and type ffmpeg -i http://localhost:8090/test.flv
[16:22] <Tjoppen> ckb: you'll get worse quality then doing it properly
[16:22] <ckb> That's what I was assuming
[16:22] <ckb> Tjoppen: is it significant?
[16:23] <ckb> or can I get away with it?
[16:23] <burek> ckb, why doing that if you can do it all in one cmd
[16:24] <burek> from 1 source
[16:24] <ckb> burek
[16:24] <darkb0e> burek: http://pastebin.com/FzEeivbn
[16:25] <Tjoppen> indeed. it'll be faster to encode everything in one go
[16:25] <burek> ffmpeg -i -vcodec ... -acodec ... -b 1200k out1.avi -b 700k out2.avi -b 400k out3.avi
[16:25] <ckb> seriously?
[16:25] <ckb> it tells me -b is ambiguous
[16:25] <burek> try -b:v
[16:25] <ckb> this works with dual pass as well?
[16:25] <Tjoppen> he might have to specify the stream. the syntax changed a bit recently
[16:26] <burek> darkb0e, your udp input might be poor
[16:26] <burek> you could use some more frequent gop's
[16:26] <burek> to increase seekability
[16:27] <ckb> I didn't realize you could have multiple outputs with one pass
[16:27] <burek> ckb, you can set the 1st pass logfile
[16:27] <burek> for each output
[16:27] <ckb> burke, I'm not worried about the logfile
[16:27] <darkb0e> its 3 mbits input
[16:27] <ckb> burek**
[16:27] <darkb0e> 720x576
[16:27] <burek> darkb0e, from what?
[16:27] <ckb> how do I specify frame size for multiple outputs
[16:27] <burek> ffmpeg also?
[16:27] <ckb> here's my syntax: one sec
[16:27] <darkb0e> gop 20?
[16:27] <burek> ckb, frame size?
[16:28] <ckb> yeah burke.. one sec I'll show you my cmds
[16:28] <burek> darkb0e, how do you generate udp input
[16:28] <ckb> ugh typos :(
[16:28] <darkb0e> don't know, service provider sends me
[16:30] <burek> it seems ffmpeg dropped support for -analyzeduration
[16:30] <burek> too bad
[16:31] <burek> now you are left behind :S
[16:31] <darkb0e> burek: YEAH :) thanks man, gopsize 200 works
[16:31] <burek> :)
[16:32] <ckb> burek: http://pastebin.com/bpdTArvp this is a single 2 pass encoding of 700Kb/s& If I wanted to make 2 more outputs how would I specify the frame size/bitrates/etc
[16:32] <ckb> right now. I'm doing this 2 other times for 1200 and 400 (all with 2 different frame sizes)
[16:32] <burek> ckb what is a "frame size" ?
[16:33] <burek> resolution?
[16:33] <ckb> resolution?
[16:33] <ckb> yes
[16:33] <burek> :)
[16:33] <burek> well ffmpeg's syntax is: ffmpeg -i input <options1> <output1> <options2> <output2> ...
[16:33] <ckb> & lol
[16:33] <ckb> that's too easy
[16:34] <burek> so -s ... -b ... out1.avi -s ... -b ... out2.avi ...
[16:34] <ckb> burek: thanks
[16:34] <burek> :beer: :)
[16:34] <ckb> :cheers:
[16:44] <woshty> is -c:v or -vcodec the newer syntax?
[16:44] <burek> yes
[16:45] <burek> c:v
[16:45] <woshty> since what version?
[16:45] <burek> check out changelog
[16:48] <woshty> burek: http://ffmpeg.org/changelog.html ends with 0.6
[16:49] <burek> http://ffmpeg.org/releases/
[16:51] <woshty> hmm, i am running 0.10.2 but -c:v doesnt work like -vcodec does
[16:51] <ckb> burek: why is my bitrate so damn slow when having 3 output files?
[16:51] <burek> woshty, that should be a bug then
[16:51] <burek> ckb, ?
[16:51] <ckb> like 11/Kbits/s
[16:52] <burek> well your cpu encodes it at the same time (all of them)
[16:52] <burek> so wait until it finishes
[16:52] <ckb> so that's why it's so slow?
[16:52] <burek> :)
[16:52] <woshty> burek: might be debian specific, will check that, thx
[16:53] <ckb> burek: http://pastebin.com/K6eRqDsG
[16:53] <ckb> burek: does that look right?'
[16:53] <burek> ckb, look at Stream #1:0
[16:53] <burek> 700 kb/s
[16:53] <burek> it's ok
[16:54] <burek> the overall kbps is 11kbps
[16:54] <burek> because you have a slow cpu
[16:54] <burek> but when it finishes
[16:54] <burek> they'll all be ok
[16:54] <ckb> ah
[16:54] <burek> also you can use -t
[16:54] <burek> to speed up tests :)
[16:54] <ckb> tests?
[16:54] <ckb> Not testing ;)
[16:55] <burek> 20_test-cactus-creek_1200k.mp4
[16:55] <ckb> eh
[16:55] <ckb> that's just the title of the video
[16:55] <ckb> that I put in
[16:55] <burek> ok then :)
[16:55] <burek> wait until it finishes :)
[16:55] <ckb> how can I upgrade the overall speed it encodes?
[16:55] <ckb> better hardware?
[16:55] <burek> usually :)
[16:55] <ckb> more ram? more cores? higher speed?
[16:56] <burek> more cpu :)
[16:56] <ckb> ah
[16:56] <burek> encoding is all about cpu :)
[16:57] <ckb> now generally speaking
[16:57] <ckb> if ONE output fails, will all others?
[16:58] <burek> why would it fail?
[17:00] <ckb> no idea
[17:01] <ckb> I just want to ensure 100% that it's going to encode
[17:02] <burek> then check the return value from ffmpeg process
[17:03] <burek> if it is 0 everything is ok
[17:03] <burek> KaosTheory-, check the trac :)
[17:03] <KaosTheory-> mr. burek
[17:03] <KaosTheory-> did
[17:03] <KaosTheory-> same problem
[17:03] <burek> I think it would be better
[17:03] <burek> to register some dummy account
[17:03] <burek> just to resolve that
[17:03] <KaosTheory-> http://ffmpeg.zeranoe.com/builds/win64/static/ffmpeg-20120409-git-6bfb304-win64-static.7z
[17:04] <KaosTheory-> downloaded that revision of the encoder
[17:04] <ckb> burek: the return value?
[17:04] <ckb> burek: not the output?
[17:04] <burek> ok
[17:04] <KaosTheory-> exactly as before
[17:04] <KaosTheory-> hangs
[17:05] <burek> KaosTheory-, can you try downloading the latest build
[17:05] <burek> (today's)
[17:05] <burek> just to see if it works
[17:05] <burek> ckb, when you start the ffmpeg process
[17:05] <burek> you can check its return value
[17:05] <KaosTheory-> umm
[17:05] <ckb> well...
[17:05] <ckb> I'm using PHP to execute ffmpeg
[17:05] <burek> even better
[17:05] <KaosTheory-> where can i find the 'todays' build?
[17:05] <burek> which function
[17:05] <burek> exec?
[17:05] <ckb> shell_exec
[17:06] <ckb> shall I use exec?
[17:06] <burek> yes
[17:06] <ckb> the return value of exec is the return value of the command?
[17:06] <burek> and check out return_var
[17:06] <burek> no
[17:06] <burek> return_var
[17:06] <burek> http://www.php.net/manual/en/function.exec.php
[17:06] <ckb> ah
[17:07] <ckb> so I can have OUTPUT and return
[17:07] <burek> KaosTheory-, are there any today's build at zeranoes?
[17:07] <ckb> burek: thanks for all your help, so much
[17:07] <burek> ckb, output will be in $output
[17:07] <burek> if you provide it
[17:07] <burek> :beer: :)
[17:08] <KaosTheory-> to confirm link to builds is: http://ffmpeg.zeranoe.com/builds/
[17:08] <ckb> exec($passone, $output, $encoded); -> $output to log, and $encoded == 0 if all is OK?
[17:08] <burek> http://www.php.net/manual/en/function.exec.php
[17:09] <ckb> right
[17:09] <ckb> I'm looking at it :D
[17:12] <KaosTheory-> burek if that link above is the correct one. then the lastest build is 2012-04-09
[17:12] <KaosTheory-> thats the one i have just downloaded
[17:12] <KaosTheory-> and it didnt work
[17:12] <ckb> burek: you said if ffmpeg's return is 0, all is ok?
[17:13] <burek> KaosTheory-, hmh..
[17:13] <burek> well ok
[17:13] <burek> do you have link to trac
[17:14] <burek> ckb yes
[17:14] <KaosTheory-> http://ffmpeg.zeranoe.com/builds/win64/static/ffmpeg-20120409-git-6bfb304-win64-static.7z
[17:14] <burek> KaosTheory-,  trac, not build
[17:14] <KaosTheory-> thats the revision i downloaded
[17:14] <KaosTheory-> oh
[17:14] <KaosTheory-> trac?
[17:14] <burek> bug report
[17:14] <KaosTheory-> oh
[17:14] <KaosTheory-> hang on
[17:14] <JamesJJ> OK... tried with a copy of ffmpeg from svn. Still doesn't like G711 in an flv - Stream #0:1: Audio: pcm_mulaw, 5512 Hz, 1 channels, s16, 44 kb/s - http://pastebin.com/zfSxYaRE
[17:15] <JamesJJ> looks from that it is the flv demuxer at issue
[17:15] <burek> JamesJJ, svn has been abandoned a long time ago
[17:15] <burek> git is used
[17:15] <JamesJJ> sorry 0 from git
[17:15] <KaosTheory-> http://pastebin.com/SYub7kF2
[17:18] <burek> KaosTheory-, I was looking for this :) https://ffmpeg.org/trac/ffmpeg/ticket/1192
[17:18] <burek> JamesJJ, if you are using the latest git
[17:18] <burek> can you please use pastebin.com, to show your command line and its output?
[17:19] <KaosTheory-> oh
[17:19] <KaosTheory-> sorry burek
[17:19] <KaosTheory-> lol
[17:20] <KaosTheory-> burek do you think its worth mentioning about the Lagarith Lossless Codec being used?
[17:20] <JamesJJ> burek - http://pastebin.com/zfSxYaRE
[17:21] <burek> KaosTheory-, if you've read that bug report you'd see that ffmpeg of the developer has detected that encoder
[17:22] <burek> JamesJJ, can you do just ffmpeg -i camera.stream.flv
[17:23] <KaosTheory-> so it has ;)
[17:24] <JamesJJ> http://pastebin.com/u0hEtJsG
[17:25] <JamesJJ> flv @ 0x24c4040] Could not find codec parameters (Audio: none ([8][0][0][0] / 0x0008), 5512 Hz, 1 channels <-- the audio is g711 at 8000
[17:26] <burek> hm
[17:26] <burek> how did you install your ffmpeg
[17:26] <JamesJJ> clean ubuntu server / got from git / make & make install
[17:26] <burek> this is not from git
[17:27] <burek> git versions dont look like 0.10.2.git-8274b21
[17:27] <burek> they look like N-8274b21
[17:27] <burek> something
[17:27] <JamesJJ> is ffmpeg-8274b21
[17:28] <JamesJJ> oh, was snapshot... OK back in a mo
[17:28] <burek> http://ffmpeg.org/download.html
[17:29] <burek> just do git clone...
[17:29] <burek> and make it
[17:29] <JamesJJ> git://source.ffmpeg.org/ffmpeg.git?
[17:29] <burek> I don't know
[17:29] <burek> it says there on that page
[17:29] <burek> check it out
[17:30] <burek> git clone git://source.ffmpeg.org/ffmpeg.git
[17:30] <JamesJJ> yup, doing that
[17:33] <steinesel> I am trying to use ffmpeg to convert an avi format to divx can someone help me with the parameters for this?
[17:36] <DTH0> hello, I'm trying to get a video out a big packed file which I don't know the format (.DAT, not VCD), ffplay does find a video stream though there are more than one. The thing is that I can't figure out where is the sound. ffplay is printing this : http://paste2.org/p/1976754
[17:36] <DTH0> can someone help ?
[17:37] <burek> steinesel,
[17:37] <burek> can you please use pastebin.com, to show your command line and its output?
[17:38] <taliaraven> Are any of you mpegts experts? :) I want to modify how often PES packets are added to a stream, to increase the granularity of PTS timestamps in it. Is this possible?
[17:39] <burek> DTH0, the format is Input #0, mpeg, from 'ZOE.DAT':
[17:40] <burek> also, you have 5 audio streams
[17:40] <burek> but your ffmpeg is too old
[17:40] <burek> and it might be the reason
[17:40] <burek> why it doesn't read the file correctly
[17:41] <burek> try updating your ffmpeg
[17:41] <JamesJJ> seems changing my ffmpeg gives the same output - http://pastebin.com/KLuXhNA5
[17:41] <burek> taliaraven, are you using ffmpeg cmd line tool or libav library
[17:41] <DTH0> The thing is that the .DAT file was not meant to be read by ffmpeg
[17:41] <taliaraven> burek: cli.
[17:41] <DTH0> it's a file extracted from a game DVD for PS2
[17:42] <burek> JamesJJ, can you try this static build
[17:42] <DTH0> so it's a bit hacking to try to read it with ffmpeg
[17:42] <burek> https://sites.google.com/site/linuxencoding/builds
[17:42] <burek> just download-and-run with ./ffmpeg
[17:42] <DTH0> or something that's not a Playstation
[17:43] <burek> DTH0, it's not hacking, ffmpeg recognized it without any problems
[17:43] <burek> you can do something like ffmpeg -i input.dat -vn out.wav
[17:44] <burek> to see if it extracts audio
[17:44] <burek> if not, you might update your ffmpeg first
[17:44] <burek> and then try again
[17:44] <taliaraven> burek, in the interest of being proactive, here's the CLI output if it helps. http://pastebin.com/YdXZ19pe
[17:44] <burek> taliaraven, thanks :)
[17:44] <DTH0> "Output file #0 does not contain any stream"
[17:44] <taliaraven> My issue is getting this ts stream to play onto an STB, and the stb seems to choke on the audio if it doesn't recieve enough PTS timestamps.
[17:45] <burek> taliaraven, you might take a look at x264 options
[17:45] <burek> because it's not ffmpeg specific
[17:45] <burek> but rather x264
[17:45] <JamesJJ> burek - the static version gives the same - http://pastebin.com/PfMXcp3R
[17:45] <taliaraven> mpeg transport stream is, burek? That doesn't make sense.
[17:45] <burek> JamesJJ, you might contact zeranoe to resolve that?
[17:45] <burek> taliaraven, that's what ffmpeg recognized
[17:46] <JamesJJ> oh? how would I do that?
[17:46] <burek> click on contact? :)
[17:47] <taliaraven> burek, thank you for being both delicious and helpful. Can I ask you to clarify why an mpegts issue is resolved in x264?
[17:47] <JamesJJ> I'm lost - sorry, I don't know who  zeranoe  is or who to contact him/her
[17:48] <taliaraven> Because the PTS I need should actually be on the audio stream, which is AAC. The video plays back fine, it's the audio timing that appears to be off. THe PTS in my output stream is only delivered every 300ms, where my reference stream shows it every 20ms. I need that amount of granularity to make an STB work.
[17:50] <taliaraven> sorry, I know this is a lot. Thanks for looking any way. :)
[17:52] <burek> taliaraven, did you try using native aac encoder instead of faac
[17:53] <burek> JamesJJ, sorry my mistake
[17:53] <burek> JamesJJ, can you try: ffmpeg -i camera.stream.flv -vn out.wav
[17:53] <burek> and then ffmpeg -i out.wav
[17:55] <JamesJJ> http://pastebin.com/dHUTq3b8 - doesn't go far, just get a 0k file
[17:59] <taliaraven> burek, I wasn't using aac native because it was listed as experimental. The issue still seems to be with my mux, rather than my encode, though.
[18:00] <burek> JamesJJ, it seems your ffmpeg doesn't have linked some library (audio encoder/decoder) needed to recognize that audio stream
[18:01] <burek> taliaraven, can you try changing your muxer to mp4 or flv
[18:01] <burek> and see if the problem persists
[18:01] <taliaraven> No, I can't, burek. MPEGTS is necessary in this case
[18:01] <taliaraven> the ts container is the only one that allows me to stream across a network correctly.
[18:02] <burek> because ffmpeg's mpegts implementation has usually been criticized as not stable and such
[18:02] <burek> so I'm afraid you might want to try with vlc or some other ts muxer
[18:03] <taliaraven> Maybe. I was hoping there's some more configuration options I wasn't aware of.
[18:03] <taliaraven> brb.
[18:03] <JamesJJ> Hmmm, I don't think so - the audio G711 is supported, it looks to me like the FLV demuxer needs to be updated for other than mp3 audio. It gets a 0x008 as the code for the codec and doesn't know what that codec is
[18:04] <burek> JamesJJ, try typing: ffmpeg -codecs
[18:04] <burek> and see is there g711 in that list
[18:04] <burek> if not, then you didn't compile your ffmpeg with that lib
[18:05] <DTH0> where can I fing libavutil ?
[18:05] <DTH0> *d
[18:05] <burek> try #libav
[18:05] <DTH0> thanks
[18:07] <JamesJJ> I have the codec I think becouse:
[18:07] <JamesJJ> ffmpeg.exe -acodec pcm_mulaw -i camera.stream.flv -vcodec copy -ar 44100 -ac 2 -ab 192 -f mp3 test.mp3
[18:08] <JamesJJ> works, kinda...
[18:08] <JamesJJ> but becouse the demuxer is getting the input frequency wrong, I get a audio track 30% longer than it should be
[18:10] <JamesJJ> oops - that should be ffmpeg.exe -acodec pcm_mulaw -i camera.stream.flv -vn -ar 44100 -ac 2 -ab 192 -f mp3 test.mp3
[18:39] <fenrig> Hi I'm using ffprobe to count the I and P frames, so I'm using a python script to count the output of the "ffprobe -show_frames <file>" command output but I can't seem how to see if a frame is a B frame :s
[18:42] <burek> fenrig, you can always use a video editor :)
[18:42] <burek> JamesJJ, try ffmpeg -acodec ... -ar 8000 -i camera..
[18:45] <fenrig> burek: I'd rather find out how I could count all the B frames
[18:46] <Durmush> fenrig: mux to mkv then use use mkvinfo -s
[18:47] <Mista_D> Gents, I'm testing "lossless" 1080p `ffmpeg -i 1.mpg -sameq -vcodec mpeg4 -an 2.mpg` Getting a lot of warnings "{mpeg @ 0xd6d9940} packet too large, ignoring buffer limits to mux it...buffer underflow". Any advice?
[18:47] <taliaraven> Fenrig, you didn't encode the video originally with ffmpeg, did you?
[18:47] <fenrig> taliaraven: No I did not, that's not even the purpose.
[18:47] <burek> Mista_D,
[18:47] <burek> can you please use pastebin.com, to show your command line and its output?
[18:48] <burek> taliaraven, you can try with tsmuxer tool
[18:48] <taliaraven> I'm considering it burek, that's on the list of things to try next.
[18:49] <burek> fenrig, try also mediainfo
[18:49] <fenrig> taliaraven: I actually just wanna analyze a video file, it's for my multimedia course. We have to analyze different containers and video codecs
[18:50] <fenrig> burek: Tried it, doesn't decently work on linux so I'd rather not bother :D XD
[18:54] <taliaraven> what about mediainfo?
[18:55] <taliaraven> no, sorry, that's just metadata and types. won't give a framecount
[18:56] <fenrig> actually it's kinda funny
[18:57] <taliaraven> Mmm, fenrig. Mediainfo will actually tell you if b-frames are enabled in the encoding if you pass it the -f flag
[18:57] <taliaraven> Just checked
[18:57] <JEEB> I think using the ffmpeg's APIs you could get the frame listing and frame types. Just don't ask me how :P
[18:57] <taliaraven> but I can't get a count from it. I'll poke at probe some more cause I'm curious now :)
[18:57] <fenrig> does anyone know a file with B frames in them
[18:58] <fenrig> maybe i'm testing files without B frames, could be it (I know I'm not that clever -.-)
[18:58] <taliaraven> sintel trailer in mp4 does.
[18:59] <taliaraven> I've been using the open movie project stuff as test assets.
[19:00] <fenrig> okay no my bad, I was using files without B frames, damn I'm stupid bothering you guys with this XD
[19:01] <fenrig> taliaraven: I'm using them too for my analysis: big buck bunny, Elephants Dream and Sintel
[19:01] <taliaraven> excellent. :)
[19:02] <DTH0> hmm one think : Is x264 is warning me about some libav stuff while configuring but... do i need to compile libav in order to get x264 working so that I can compile ffmpeg ? I just found out that libav and ffmpeg aren't apparently meant to be installed together... what should I do ? (I want ffmpeg).
[19:03] <ckb> burek: this is so much easier! :D
[19:04] <ckb> DTH0 if you have another audio codec you plan on using (libfaac) then libav isn't required.. since you're using libx264
[19:04] <DTH0> libfaac can't be enabled in ffmpeg ?
[19:04] <ckb> huh?
[19:05] <DTH0> ah sorry misunderstood what you were saying
[19:05] <ckb> libav is like a generic audio/video codec
[19:05] <ckb> but if you know specifically what scope you're using ffmpeg for, then you're all good
[19:06] <ckb> speaking of all good. Kind of pissed it's in Indiana this year
[19:06] <DTH0> ok
[19:39] <level09> can ffmpeg convert quick time reference files ?
[19:52] <zap0> probably.
[19:53] <zap0> level09, ffmpeg -formats      or   -codecs
[20:06] <level09> thnx will check
[20:31] <ckb> wow
[20:32] <ckb> this 2 hour movie is taking 2 hours to encode.
[20:32] <ckb> :(
[20:33] <silverrocker> questions about using ffmpeg as a library in my project can be asked here?
[20:37] <relaxed> silverrocker: sure
[20:39] <silverrocker> I can compile the example decode_encode.c provided in doc/examples/ but this only works for mpg1 files. I would like to extend this so that it detects the file format and loads the appropeate codec for that file. I have used libavformat to find the codecid of the codec in question for the first video stream of an arbitrary file.
[20:39] <silverrocker> but when I try to decode anything else then a mpg1 file, I get errors form the corresponding codecs
[20:42] <fenrig> I'm having difficulties using ffprobe, I want ffprobe to report the audio bitraete
[20:43] <JEEB> extract the audio track, divide size with length :P
[20:44] <JEEB> and you will get the average bit rate
[20:47] <fenrig> JEEB: that's a real crappy way to do it, isn't ffprobe made to analyze mediafiles. Meanwhile everyone is extracting audio to get the average bit rate :P that doens't sound too clever
[20:48] <JEEB> it is, but if it isn't doing what you want :P
[20:49] <JEEB> also you'd need to go thorugh the whole audio track in any case to get the whole average bit rate
[20:49] <JEEB> if you can't go through it with code, it's just easier to extract it and count it like that >_>
[20:50] <fenrig> JEEB: besides I need to know if it's CBR, VBR too
[20:50] <JEEB> it's better to just assume VBR
[20:51] <fenrig> JEEB: I have to analyze files for school
[20:52] <JEEB> also, for CBR/VBR you'd have to then check the sizes of every packet, and check if the length of every packet is the same in that given format
[20:52] <JEEB> if the length for every packet is the same, and the file size for every packet is the same -> CBR
[20:52] <JEEB> it gets messy if the length of every packet is _not_ the same
[21:00] <Durmush> Hi, i'm trying to build ffmpeg in msys-mingw, but whatever I tell it, it refuses to link to pthreads staticly. here is my configure line: --enable-gpl --enable-version3 --enable-nonfree --enable-static --extra-ldflags=-static --enable-libmp3lame --enable-libxvid --enable-pthreads --disable-doc --disable-shared
[21:21] <Mista_D> Problem extracting s320m audio. http://pastebin.com/7ZYk9AtA
[21:23] <Mista_D> *s302m
[21:28] <bcoudurier> don't put s302m in .wav
[21:29] <Mista_D> bcoudurier: it was just a test.
[21:30] <bcoudurier> that test won't work
[21:31] <Mista_D> bcoudurier: any way to preserve s320m in any other container?
[21:36] <bcoudurier> no, ts only
[21:58] <burek> Durmush, what does your config.log says at the end
[21:58] <burek> say*
[22:00] <burek> Mista_D, did you try ffmpeg102 -i orig.mpg -vn -acodec copy -aframes 10 out.mp4
[22:07] <iive> burek: what happened with the hanging avi?
[22:08] <burek> I've updated trac pst
[22:08] <burek> post
[22:09] <burek> https://ffmpeg.org/trac/ffmpeg/ticket/1192
[22:10] <Durmush> burek: the problem has gone. a rename of certain file did the trick.
[22:10] <burek> ok
[22:10] <Mista_D> burek: tried s302m with mp4 - "[mp4 @ 0x113d1920] track 0: could not find tag, codec not currently supported in container"
[22:10] <burek> I see
[22:11] <burek> what is s302m anyway :)
[22:29] <Mista_D> burek: some codec for high quailty audio...
[22:31] <Mista_D> burek: aka BSSD "Audio data as specified in SMPTE Standard SMPTE 302M-1998 for Television - Mapping of AES3 Data into MPEG-2 Transport Stream shall use this format_identifier." so its AES3 codec in MPEG2.
[22:33] <DTH0> hey, I'm following this guide : "https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide". The compilation and the installation of x264 seems to go well but when trying to compile ffmpeg, it's stating that there's no libx264, though I'm using last version from git
[22:33] <Mista_D> burek: s302m = BSSD = 2 PCM channels wrapped in AES3, then in MPEG2-TS.
[22:35] <DTH0> I'm also getting a weird version from the version.sh file ("#define X264_POINTVER "0.122.2184 5c85e0a") which doesn't look too promising...
[22:36] <DTH0> I strictly followed the guide, is the mistake mine ?
[22:36] <DTH0> the version.sh script is from gitting x264
[22:38] <NateW> Anyone around?
[22:38] <taliaraven> plenty of people around.
[22:38] <NateW> Just checking. Is it possible to use more than one audio input when using ffmpeg.
[22:38] <NateW> From either pulseaudio or alse?
[22:38] <NateW> *alsa
[22:40] <NateW> Currently, to do a proper screen capture, I have to create a loopback of my mic and record the desktop sound, I was hoping I can record both the mic and the desktop sound without having to setup a loopback.
[22:41] <NateW> It's really annoying having to listen to an echo of what I just said 100ms ago.
[22:43] <NateW> I have found many forum posts about recording screencasts, but none cover recording from more than one input, so is it an ffmpeg limitation?
[22:46] <NateW> taliaraven: do you have any idea?
[22:48] <pasteeater> DTH0: what version of ubuntu?
[22:55] <taliaraven> You should be able to combine a couple inputs; depends on how you want to do it. If you want them as seperate channels, check out this: http://ffmpeg.org/ffmpeg.html#amerge
[22:56] <taliaraven> And the pan filter should let you combine two sources into stream. In either case, you're going to have to do some more complex things and build out a filter graph
[22:57] <taliaraven> It explains the syntax for how to do that here: http://ffmpeg.org/ffmpeg.html#Filtergraph-description
[22:57] <NateW> As I said above, I'm doing screencasts. I can set my mic level to match the audio level, I just need to be able to record from both the mic and desktop audio (They can be combined) at the same time.
[22:58] <NateW> now with this: ffmpeg -f pulse -i default, could I specify multiple inputs like: ffmpeg -f pulse -i mic,desktop
[22:59] <NateW> obviously I'd have to properly name the pulse inputs.
[23:00] <taliaraven> Beyond what I just pointed out, I won't be much help. I've built filtergraphs in gstreamer, but not ffmpeg, and I don't have the time right now to research, sorry :(
[23:00] <NateW> No probs. I was hoping someone would have the knowledge, if it is just research, I can do that.
[23:01] <taliaraven> The links I posted should be a good place to start.
[23:01] <NateW> Yep.
[23:09] <DTH0> pasteeater, 11.10
[23:18] <DTH0> anyone have an idea ? (I was AFK, sorry)
[23:26] <burek> DTH0, you are compiling ffmpeg with libx264?
[23:26] <DTH0> yes
[23:26] <burek> that's easy
[23:27] <burek> http://ffmpeg.gusari.org/viewtopic.php?f=25&t=36
[23:27] <burek> that's for x264
[23:27] <burek> http://ffmpeg.gusari.org/viewtopic.php?f=25&t=14
[23:27] <burek> that's all together (just ignore faad and other stuff you don't need)
[23:28] <DTH0> ok
[23:28] <burek> or, if you prefer apt
[23:28] <DTH0> I was just avoiding the make install stuff
[23:28] <burek> (but that will install some old version of ffmpeg)
[23:28] <burek> you can install libx264-dev
[23:28] <burek> prior to compiling ffmpeg
[23:28] <DTH0> I followed the ffmpeg guide
[23:28] <burek> but my advice is to always compile multimedia projects
[23:28] <burek> they change rapidly
[23:29] <DTH0> "https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide"
[23:29] <burek> and repos cannot follow them quickly enough
[23:29] <burek> I don't have time to read that, but you get the general idea, ok? :)
[23:29] <DTH0> yes
[23:29] <DTH0> but
[23:29] <DTH0> libx264 is properly installed
[23:29] <burek> libx264-dev
[23:29] <burek> not libx264
[23:30] <DTH0> I'm compiling from git
[23:30] <burek> I don't know, I got to go
[23:30] <DTH0> so libx264-dev files should be there as well right ?
[23:30] <burek> read those links
[23:30] <burek> and decide for yourself
[23:31] <DTH0> It's almost the same thing that what i've done. Compilation and installation of x264 libraries go well. Problem is ffmpeg won't detect it
[23:31] <DTH0> when configuring
[23:31] <DTH0> before compilation
[23:31] <burek> what does your config.log say
[23:31] <burek> can you please use pastebin.com, to show your command line and its output?
[23:37] <DTH0> http://paste2.org/p/1977085
[23:39] <burek> add --enable-shared
[23:39] <burek> anyway
[23:39] <burek> can you copy/paste config.log from FFmpeg's ./configure ?
[23:41] <DTH0> http://paste2.org/p/1977090
[23:43] <burek> well it seems that you didn't properly install your compiled x264
[23:43] <burek> maybe checkinstall made some errors or something
[23:43] <burek> or
[23:43] <burek> you didn't type ldconfig
[23:43] <burek> to update your libs
[23:43] <burek> after checkinstall
[23:43] <DTH0> i didn't entered ldconfig
[23:43] <burek> try typing ldconfig
[23:44] <burek> and retry ./configure for ffmpeg
[23:45] <DTH0> but the output of  version.sh from git is strange anyway ("#define X264_POINTVER "0.122.2184 5c85e0a")
[23:46] <burek> I really don't know, I'm just trying to help you make it work.. I don't use checkinstall
[23:46] <burek> I just use what I gave you on those likns
[23:46] <burek> and it works everywhere and always
[23:47] <DTH0> ok
[23:48] <DTH0> ldconfig did the trick
[23:48] <burek> cool :)
[23:48] <DTH0> should be added to the guide then since it's the official one :)
[23:48] <DTH0> thanks
[23:49] <burek> :beer: :)
[23:51] <burek> no
[23:51] <burek> oops, wrong channel
[23:51] <burek> :)
[23:52] <DTH0> ><
[23:55] <DTH0> wow ##beer exist but it's invite only ...
[23:55] <burek> :D
[00:00] --- Fri Apr 13 2012