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February 2012

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ffmpeg-devel.log.20120228
by burek 29 Feb '12

29 Feb '12

[02:23] <darkfires> is there a way to force ffmpeg to this Stream #0.0: Video: mpeg4, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 25 fps, 25 tbr, 1200k tbn, 30 tbc? I'm having a problem with an IP camera, it's got a bug in the firmware and the manufact wont respond to my emails, basically the first time you connect the proper vol header is sent, any subsequent connect without a reboot the header is corrupt [02:23] <darkfires> i've been trying with various options but im getting no where [03:26] <Compn> darkfires : can you relay the stream so we can access it [03:26] <Compn> there is a program called socket2socket, part of desproxy project that can do it, if you cant figure it out in your router ... [03:27] <Compn> and whats ffmpeg say about it after the header is corrupt ? [03:27] <Compn> its possible to force it, depends how much data is coming [03:27] <Compn> and the format the camera streams [03:27] Action: Compn goes afk [03:53] <CIA-17> ffmpeg: 03Anton Khirnov 07master * rf5f5b154e7 10ffmpeg/doc/APIchanges: doc/APIchanges: fill in missing dates and hashes. [03:53] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r1d428004c6 10ffmpeg/libavcodec/avcodec.h: [03:53] <CIA-17> ffmpeg: lavc: reorder AVFrame fields. [03:53] <CIA-17> ffmpeg: Put related fields together, move the important ones closer to the [03:53] <CIA-17> ffmpeg: beginning, try to eliminate holes. [03:53] <CIA-17> ffmpeg: 03Anton Khirnov 07master * rb1edfce08a 10ffmpeg/libavcodec/ (avcodec.h pthread.c): [03:53] <CIA-17> ffmpeg: lavc: reorder AVCodecContext fields. [03:53] <CIA-17> ffmpeg: Put related fields together, move the important ones closer to the [03:53] <CIA-17> ffmpeg: beginning, try to eliminate holes. [03:53] <CIA-17> ffmpeg: 03Justin Ruggles 07master * rf234e8a32e 10ffmpeg/libavformat/movenc.c: [03:53] <CIA-17> ffmpeg: movenc: use timestamps instead of frame_size for samples-per-packet [03:53] <CIA-17> ffmpeg: For encoding, AVCodecContext.frame_size is the number of input samples to [03:53] <CIA-17> ffmpeg: send to the encoder and does not necessarily correspond directly to the [03:53] <CIA-17> ffmpeg: timestamps of the output packets. [03:53] <CIA-17> ffmpeg: 03Tommy Winther 07master * r1a6b9a98ce 10ffmpeg/libavformat/rtsp.c: (log message trimmed) [03:53] <CIA-17> ffmpeg: rtsp: Handle requests from server to client [03:53] <CIA-17> ffmpeg: This returns 200 OK for OPTIONS requests and 501 Not Implemented [03:53] <CIA-17> ffmpeg: for all other requests. [03:53] <CIA-17> ffmpeg: Even though this doesn't do much actual handling of the requests, [03:53] <CIA-17> ffmpeg: it makes the code properly identify server requests as such, instead [03:53] <CIA-17> ffmpeg: of interpreting it as a reply to the client's request as it did [03:54] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * re60bdb7e5c 10ffmpeg/libavformat/flvdec.c: [03:54] <CIA-17> ffmpeg: flvdec: Remove the now redundant check for known broken metadata creator [03:54] <CIA-17> ffmpeg: The index validation identifies these indexes as broken. [03:54] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [03:54] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r7e297a46db 10ffmpeg/libavformat/flvdec.c: (log message trimmed) [03:54] <CIA-17> ffmpeg: flvdec: Validate index entries added from metadata while reading [03:54] <CIA-17> ffmpeg: By validating the index entries while reading, we don't need to [03:54] <CIA-17> ffmpeg: seek at startup to validate the entries. If the error in the [03:54] <CIA-17> ffmpeg: index entries is not pointing to (our definition of) the start [03:54] <CIA-17> ffmpeg: of packets, and there is an index entry pointing at some of the [03:55] <CIA-17> (18 lines omitted) [04:03] <Compn> long commit messages :P [07:37] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8b1cd25ca7 10ffmpeg/libavformat/pmpdec.c: [07:37] <CIA-17> ffmpeg: pmpdec: Check for zero audio packets. [07:37] <CIA-17> ffmpeg: This fixes a division by 0. [07:37] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [07:37] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [12:15] <trupheenix> Hi. I need to know if there are any good sources where I can understand about the fundamentals behind video codecs? [12:45] <Compn> wikipedia has that stuff trupheenix [12:45] <av500> also the video from greg [12:45] <trupheenix> Compn, any good books i can read? or there are no books on this subject? [12:47] <av500> tried amazon? [12:47] <trupheenix> av500, no [12:47] <trupheenix> av500, what video from greg? [12:50] <av500> http://wiki.xiph.org/A_Digital_Media_Primer_For_Geeks/Episode_01 [13:11] <Compn> trupheenix : oh, you mean video coding and dct and stuff [13:11] <Compn> i thought you meant ... how codecs work, not the algorithms and such [13:11] <Compn> yeah there are books [13:11] <trupheenix> Compn, !!! yes!! :) finally! :) [13:11] <j-b> where is episode 2? [13:12] <Compn> trupheenix : http://codecs.multimedia.cx/?p=70 [13:12] <Compn> lists a couple good books [13:13] <Compn> as for the rest, i dunno, best ask someone else, maybe kostya :) [13:16] <trupheenix> Thanks Compn. I'm pretty bad at my calculus. Need to see these! :) [14:09] <kierank> oh he's gone [14:40] <funman> michaelni: you dont' want to use separate postproc repo because of fate? couldn't fate be changed to fetch postproc? [15:35] <Compn> cant git do like svn externals ? [15:36] <Compn> not that i care one way or the other [15:36] <JEEBsv> yes [15:36] <JEEBsv> git submodules [15:51] <michaelni> JEEBsv, submodules are very annoying [15:52] <av500> use repo [15:52] <michaelni> funman, ill just keep external and internal in sync [15:54] <funman> sounds like a plan [15:55] <michaelni> ive looked at submodules and subtree and a few other things but it all looked like it would be a bigger mess [15:55] <funman> in that case merging is free [15:59] <michaelni> Daemon404, ping [16:00] <michaelni> wth is this: Merge: 3a6e4e6 a792a83 8cf4ef5 59d8d9c 7876565 ??? [16:00] <michaelni> i mean yeah its a merge of 5 branches but [16:00] <michaelni> why is it there in libpostproc repo [16:03] <michaelni> also, whats "This file is part of FFmpeg/Libav." ?? [16:03] <michaelni> libpostproc was not developed in libav and isnt part of it either [16:03] <michaelni> its either part of FFmpeg or part of libpostproc [16:06] <michaelni> copyright (c) 2006 Michael Niedermayer <michaelni(a)gmx.at> [16:06] <michaelni> This file is part of Libav. [16:07] <Compn> its part of XXX or YYY , the line is only to provide where the file came from [16:07] <Compn> altho why it says libav when its from ffmpeg is another question [16:07] <Compn> similar things were done to files in mplayer that mplayer collected over the years [16:08] <Compn> i dont remember the reasoning for abandoning the original 'file is from' part, only so people dont copy over incorrect license headers i guess [16:08] <Daemon404> [10:07] <@Compn> altho why it says libav when its from ffmpeg is another question <-- it has stuff from both sides. [16:08] <Compn> definitely a question to ask dondiego [16:08] <Daemon404> :politics: [16:08] <michaelni> libav and 2006 ... [16:09] <Compn> Daemon404 : it should say @Copyright blah blah if that were the case [16:09] <Daemon404> it has code from both sides. [16:09] <Compn> the 'file is from' is not a copyright declaration [16:09] <Daemon404> what [16:09] <Daemon404> [10:00] < michaelni> wth is this: Merge: 3a6e4e6 a792a83 8cf4ef5 59d8d9c 7876565 <-- i wodnered this myself [16:09] <Daemon404> it seems to be a biproduct of filter-branch [16:10] <Daemon404> [10:03] < michaelni> libpostproc was not developed in libav and isnt part of it either <-- it has commits from them [16:10] <Compn> Daemon404 : i just mean, if libav has changed the file, whoever changed the file should put his copyright on there, assuming it was enough code to add copyright .. [16:10] <Daemon404> it is not a matter of copyright as far as im concerned [16:10] <Daemon404> it;s a matter of where it lived [16:10] <Daemon404> it lived in both ffmpeg and libav. [16:11] <Daemon404> and seriously, you gusy can take your politics and shove it. [16:11] <Daemon404> i will not involve myself in them. [16:11] <michaelni> <Daemon404> it has code from both sides. [16:11] <michaelni> then git blame isnt working [16:11] <Compn> i'm not trying to start a flame so i'll go away... [16:11] <Compn> michaelni : are you saying there are no changes from libav ? [16:11] <michaelni> Compn, not in x86_cpu.h [16:12] <michaelni> or blame -C -w is lying to me [16:12] <Daemon404> then change it [16:12] <Daemon404> i seriously do not give a shit about this libav/ffmpeg political bullshit anymore. [16:12] <Daemon404> do as you wish. [16:13] <michaelni> Daemon404, what about the 5 head merge ? [16:13] <Compn> Daemon404 : you sound upset :( [16:13] <Daemon404> on the technical side there, i dont know what to do to fix that per se, michaelni [16:14] <Daemon404> do you want my git filter branch commandss? [16:14] <michaelni> i could just clone ffmpeg and throw all bit libpostproc out and it wont have such monsters [16:14] <Daemon404> perhaps you could spot it [16:14] <Daemon404> Compn, i merely no longer wish to ever hear any of this politics crap again. too many things i do get plagued with it. [16:14] <Daemon404> i prefer ignorant bliss now. [16:15] <michaelni> if libpostproc is a true clone i also could do a true git merge to keep them in sync ... [16:17] <Daemon404> it's not a clone [16:17] <Daemon404> it's a filtered branch. [16:17] <michaelni> i know [16:17] <Daemon404> my intention was to filter out the noise of all the otehr unrelated commits [16:17] <michaelni> i understand that [16:17] <michaelni> i dont understand why it has 5 headed monsters in there [16:19] <Daemon404> i think it may be teh 2nd command i ran... maybe [16:19] <Daemon404> sec [16:19] <Daemon404> (it has to do with the merges you did from libav) [16:20] <Daemon404> http://pastebin.com/raw.php?i=gr9SUP3M [16:20] <Daemon404> ^ what i used [16:20] <Daemon404> just a ... bit ... ugly [16:22] <michaelni> whats your oppinon on using a true clone, its kinda ... less work ? [16:22] <Daemon404> far less work [16:22] <Daemon404> less nice lookign history wise [16:23] <Daemon404> it's up to you. [16:23] <michaelni> if its up to me, ill pick a clone [16:23] <Daemon404> ok. [16:42] <bgmarete> Hello. I am looking for interlaced video sequences to test the yadif filter. Any down-loadable ones from the web? With varying field parity? Or how can I use FFmpeg to create such? Thanks! [16:46] <cbsrobot> bgmarete: what is varying field parity ? [16:47] <bgmarete> cbsrobot: I mean I would like some samples to be tff and others bff. [16:47] <bgmarete> (If possible) [16:48] <michaelni> i remember generating such (half framerate) sequences with mencoder long ago from progessive input [16:48] <michaelni> i dont remember how exactly i did it though ... [16:49] <michaelni> Tjoppen, there are mxf patches on ffmpeg-dev [16:56] <Compn> there are some interlaced mpeg2's on sample repo [16:56] <Compn> but hes gone [17:01] <mystica555_> heh [17:02] <mystica555_> at one point i came up with a very good resizing interlace preserving commandline for 1080i to 720x480i dvds.. but, ive modified the script in the meantime and its not the one that actually worked [17:03] <mystica555_> why i didn't keep using the commandline and up-enter, i'll smack myself for.. i store those logs <.< [17:19] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r1ebc5e71a0 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/pp): [17:19] <CIA-17> ffmpeg: tests: add libpostproc test [17:19] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:19] <CIA-17> ffmpeg: 03Philip de Nier 07master * r539a8469ae 10ffmpeg/libavformat/utils.c: [17:19] <CIA-17> ffmpeg: add v210 to intra only list to allow "-vcodec copy" [17:19] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:23] <mystica555_> apparently -vf yadif=1,scale=720:480,tinterlace=4 works well for mencoder; i was seeing audio sync issues so i just reencoded the audio on my test instead of copying, and it worked. heh. [17:24] <mystica555_> and michaelni that basically does your half framerate (tinterlace=4) from full framerate video (as given by yadif=1) [17:33] <ubitux> michaelni: is it normal to have ./presets and ./ffpresets directories? (i think i already asked, but i can't remember any answer) [18:03] <michaelni> ubitux, i dont remember the awnser either :( [18:06] <ubitux> the option system is quite complicated btw :p [18:06] <ubitux> for instance if you want to use libx264 profile [18:07] <ubitux> you can use -profile, -vprofile, profile:v [18:07] <ubitux> not sure how that work :p [18:07] <ubitux> some presets reference vprofile, some profile [19:05] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rddd86a2924 10ffmpeg/ffmpeg.c: [19:05] <CIA-17> ffmpeg: ffmpeg: Fix division by 0 due to invalid timebase [19:05] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [19:05] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:05] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r4f5d1468f5 10ffmpeg/libavformat/omadec.c: [19:05] <CIA-17> ffmpeg: omadec: Fix wrong number of array elements. [19:05] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [19:05] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:24] <michaelni> ubitux, -profile is a bit risky due to audio vs video [19:43] <bcoudurier> kierank, kostya can join the ml as well ;) [19:45] <Compn> did maxsim work on the encoder or just decoder ? [19:46] <Compn> unrelated... [19:46] <Compn> is there an easy way to diff two trunks ? [19:46] <Compn> like diff two whole dirs ? [19:49] <Compn> oh, diff does it already [19:50] <Compn> not subdirs tho :\ crappy [19:52] <Compn> ack CR vs LF diff [19:52] <Compn> with git diff [19:58] <Compn> ok , fix a few line endings and git diff works great [19:58] <Compn> yay for subdirs [19:59] <nevcairiel> you can also make it ignore whitespace differences [19:59] <Compn> reading git diff manual , i must have missed it [19:59] <nevcairiel> i would hope that also counts for line ending differences [20:00] <nevcairiel> its -w [20:00] <nevcairiel> or --ignore-all-space if you like verbose [20:00] <Compn> oh yeah its there [20:00] <Compn> i'm blind i guess [20:00] <Compn> diff -wb is old thing too, should have tried it :) [20:37] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rbf72c0410d 10ffmpeg/libavcodec/cook.c: [20:37] <CIA-17> ffmpeg: cook: add const to quant_index_table of categorize() [20:37] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:37] <CIA-17> ffmpeg: 03Philip de Nier 07master * r627cd3841b 10ffmpeg/libavformat/mxf.c: [20:37] <CIA-17> ffmpeg: mxf: fix v210 label matching len [20:37] <CIA-17> ffmpeg: Must check all 16 bytes because there is a planar 10-bit format [20:37] <CIA-17> ffmpeg: label that has equal first 15 bytes [20:37] <CIA-17> ffmpeg: Review-by: Baptiste Coudurier <baptiste.coudurier(a)gmail.com> [20:37] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:37] <CIA-17> ffmpeg: 03Philip de Nier 07master * r3592bd509c 10ffmpeg/libavformat/mxfdec.c: [20:37] <CIA-17> ffmpeg: mxfdec: set track edit rate num/den in expected order [20:37] <CIA-17> ffmpeg: This matches the order used for the index table edit rate. [20:37] <CIA-17> ffmpeg: Reviewed-by: Baptiste Coudurier <baptiste.coudurier(a)gmail.com> [20:37] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:37] <CIA-17> ffmpeg: 03Philip de Nier 07master * rc2d11a8b86 10ffmpeg/libavformat/mxfdec.c: [20:37] <CIA-17> ffmpeg: mxfdec: detect uncomp pictures using essence container ul [20:37] <CIA-17> ffmpeg: This supports detection of uncompressed picture in files that [20:37] <CIA-17> ffmpeg: The properties of the CDCI Descriptor are insufficient to specify [20:37] <CIA-17> ffmpeg: the pixel format for uncompressed picture data. SMPTE 377-1 and [20:37] <CIA-17> ffmpeg: RP224v10 have defined a set of picture coding labels to indicate what [20:37] <CIA-17> ffmpeg: formatting was used. [20:37] <CIA-17> ffmpeg: This patch uses 2 labels to detect UYVY422 or YUYV422 pixel formats. [20:37] <CIA-17> ffmpeg: 03Philip de Nier 07master * r51d632cc8f 10ffmpeg/libavformat/mxf.c: [20:37] <CIA-17> ffmpeg: mxf: add uncompressed 422 8-bit rawvideo label [20:37] <CIA-17> ffmpeg: Reviewed-by: Baptiste Coudurier <baptiste.coudurier(a)gmail.com> [20:37] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:54] <burek> if somebody is watching #ffmeg [22:54] <burek> this is the perfect example [22:54] <burek> why do we need static ffmpeg build [22:54] <burek> :) [23:12] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rf929abd0c3 10ffmpeg/libavcodec/adpcm.c: [23:12] <CIA-17> ffmpeg: adpcm_xa: Check filter validity. [23:12] <CIA-17> ffmpeg: Fixes out of global array read. [23:12] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [23:12] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Wed Feb 29 2012

1 0

ffmpeg.log.20120228
by burek 29 Feb '12

29 Feb '12

[00:45] <aixenv> hey gang - whats wrong with this line it worked a week ago, now im seeing unknown -vf ; whats the substitute? [00:45] <aixenv> . /usr/bin/ffmpeg -y -i "$file" -vcodec libx264 -r 30000/1001 -vf "yadif,scale=1024:576" -crf 21 -maxrate 2M -bufsize 6M -preset slow -threads 0 -acodec libfaac -ar 48000 -ab 128k ${file%.*}-1.mp4 >> "${file%}"-encoding.out 2>&1 [00:45] <aixenv> well maybe not a week, but around Christmas it worked [00:46] <aixenv> FFmpeg version SVN-r0.5.6-4:0.5.6-3 [00:46] <aixenv> ah i found it [00:46] <aixenv> ffmpeg -y -i "$file" -vcodec libx264 -r 30000/1001 -deinterlace -s 1024x576 -crf 21 -maxrate 2M -bufsize 6M -vpre slow -threads 0 -acodec libfaac -ar 48000 -ab 128k ${file%.*}-1.mp4 >> "${file%}"-encoding.out 2>&1 [00:47] <aixenv> disregard :) [00:49] <aixenv> bah or not [00:50] <aixenv> that one fails too [00:50] <tiglionabbit> I need to drop the audio channels on my video to to 1 or 2 so it'll work on android. Unfortunately, when I do this, the audio comes out all squeaky beepy and terrible. What can I do about this? [00:51] <tiglionabbit> here is the command: ffmpeg -i pipe:0 -vcodec libx264 -vpre fast -vpre baseline -vpre ipod320 -acodec libfaac -ac 1 -vf scale='320:trunc(ow/a/2)*2' -threads 0 -f mpegts pipe:1 [01:13] <aixenv> figured it out, debian broke my custom ffmpeg, bleh - :P [05:29] <darkstarbyte> Is there an ATI card that competes with cuda? [05:30] <darkstarbyte> (in encoding) [07:58] <overflow_0f8b> darkstarbyte<< try ati x550 pro [07:58] <overflow_0f8b> in PCI-E version, that is a good card [07:59] <overflow_0f8b> though the cheapest integrated will beat that on the current motherboards... [08:33] <thresh> michaelni: hey. do you have plans on removing libpostproc from ffmpeg.git ? [08:51] <michaelni> if i would do that we would loose fate testing it :( [08:54] <thresh> you mean compiling it? [09:05] <funman> !logs [09:30] <TimNich> If I use ffmpeg to capture from a device to a file, how can I cleanly stop the process so the file gets closed properly, or do I need to chose a format that doesn't mind abrupt truncation? [09:31] <Tjoppen> TimNich: just press q? I also think ffmpeg listens to ^C [09:31] <Tjoppen> IIRC mkv is more tolerable than say mov/mp4 [09:35] <TimNich> q sounds possible ^C is listened to but I thought sounded more drastic. mov/mp4 need to do tidying up after finishing the data stream which is why I was wondering what recommendations there were.. [09:37] <Tjoppen> I've had MKVs work even though I ran out of disk space, meaning there was no room for a footer. however, I'm not sure whether MKV uses a footer at all [09:47] <TimNich> Tjoppen: Thanks I will have a play in that area. [10:07] <TimNich> Just looking at the matroska spec, it does have a footer which includes, amongst other things, the cueing data and tags. However it can live without these, at the expense of poor seek performance if the cueing info is missing. [11:57] <kriegerod> what should i do with demuxed RTMP/FLV stream which gives me a lot of packets timestamped with 0? They result in that many outgoing frames&packets with same pts, which offend encoder and muxer. [14:57] <b0ot> Is there a way to take in a rtp or rtsp video stream in with ffmpeg and restream it out to your network in a very low latency format via multicast? [16:03] <kriegerod> isn't that wrong that FLV demuxer does not automatically set streams codecs, thus requiring av_find_stream_info? [16:32] <bgmarete> Hello. I am looking for interlaced video sequences to test the yadif filter. Any down-loadable ones from the web? With varying field parity? Or how can I use FFmpeg to create such? Thanks! [16:45] <b0ot> Is there a way to take in a rtp or rtsp video stream in with ffmpeg and restream it out to your network in a very low latency format via multicast? [17:00] <TimNich> bgmarete: as you left devel Compn: remarked "there are some interlaced mpeg2's on sample repo" [17:26] <mystica555_> TimNich: he left both at the same time ;) [17:33] <TimNich> mystica555_: Hmmm I saw it logged on devel but not ffmpeg.... [18:53] <u42p> googling for "ffmpeg gamma" only gives me ancient things. is it possible to alter gamma when encoding with ffmpeg nowadays? [18:55] <u42p> sweet, seems like i can try mp for that [18:55] <u42p> thanks you awesome manpage! [18:57] <u42p> example for anyone else who might find this at any time in the future: -filter mp=eq2=1.0:2:0.5 (and check the manpage) [19:27] <b0ot> Any idea on how I could take as an input rtp or rtsp video and restream it out my network with as low latency as possible and view it with an end device with as low latency as possible. [19:42] <Pranav> hey Guys, I needed direction in solving the issue in converting a .mov file to .mp4 [19:43] <Pranav> the version of ffmpeg is FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers built on Feb 28 2012 10:43:12 with gcc 4.2.1 (Apple Inc. build 5664) configuration: --enable-libmp3lame --enable-gpl --enable-pthreads --enable-nonfree --enable-avfilter --enable-filters [19:44] <Pranav> the real issue is "decoding prores with ffmpeg" and from online resources I read the proposed solution was to enable "gpl version 2" [19:45] <Pranav> but with the latest release the "enable-version2" does not work with the new ffmpeg build [19:45] <Pranav> is there a workaround? [19:46] <Mavrik> --enable-version3? [19:47] <JEEB> --enable-gpl is what enables gpl2, --enable-version3 is for (l)gpl3 [19:47] <JEEB> also the current ffmpeg code is in a git repository [19:47] <JEEB> not in the svn [19:48] <Pranav> ok thanks [19:48] <Mavrik> (afaik ffmpeg was re-licensed GPLv3) [19:50] <JEEB> ... [19:50] <JEEB> ffmpeg is generally LGPL [19:50] <JEEB> it has some things under the GPL, which is why there's --enable-gpl [19:50] <JEEB> and then there are some libraries etc. that need --enable-version3 [19:51] <burek> ffmpeg itself is even more than lgpl :) but the linked libs are not :) [19:51] <burek> anyway, I finally managed to create static build of ffmpeg with a lot of libs [19:51] <burek> statically built [19:52] <burek> and now that server went offline and I didn't do any backup... :idiot: [19:52] <b0ot> Why do IP cameras have such low latency when viewing the video where as when I stream with vlc there is lots of latency... and is there any way ffmpeg could help me stream extremly low latency video? [19:53] <burek> b0ot, how do you stream your video [19:53] <b0ot> I use vlc, and I specify an rtp ts stream [19:53] <b0ot> and then I view the video with vlc [19:53] <burek> the minimum latency you'll have to live with is the encoding time [19:53] <b0ot> which is about 2 seconds of lag [19:53] <burek> I don't know about rtp [19:53] <b0ot> even if I don't transcode [19:53] <burek> never used it [19:53] <burek> but udp is the fastest [19:54] <b0ot> rtp is a form of udp [19:54] <burek> also, if you use vlc, you'll have to play around with caching params [19:54] <burek> there are a lot of them [19:54] <burek> and you'll need to tweak each of them to get your desired result [19:54] <b0ot> yeah... is there something better to view it with so there is less latency [19:54] <b0ot> maybe through the web? [19:54] <b0ot> different player? [19:54] <b0ot> It is all on a private network [19:54] <b0ot> not through the net [19:54] <burek> try ffplay [20:01] <b0ot> couldn't get it to open it [20:02] <burek> do you have any specific reason to use rtp [20:02] <b0ot> besides it being supposidly better than udp which is mostly considered legacy for programs like vlc [20:02] <b0ot> no [20:03] <burek> well just try udp and see if it works faster [20:03] <burek> I have like 0.5 seconds delay [20:03] <burek> from webcam to displayed video [20:03] <burek> over 2 hops [20:04] <b0ot> what command line did you use [20:04] <burek> ffmpeg? [20:04] <b0ot> could you output your actual line [20:04] <burek> well ffmpeg -i ... -f mpegts udp://target:port [20:04] <burek> not a rocket science :) [20:05] <burek> and play it in vlc like: vlc udp://@:port [20:07] <b0ot> lol it streamed it at like 30x speed [20:09] <b0ot> and how would I do my webcam which is v4l2 [20:11] <burek> i dont understand [20:11] <burek> are you streaming a live event [20:11] <burek> or a file [20:11] <burek> if you are streaming a file, you need to read it not as fast as possible but in "real time", so add the option -re, like this: [20:12] <burek> well ffmpeg -re -i ... -f mpegts udp://target:port [20:13] <b0ot> i have ffmpeg -f video4linux2 -i /dev/video0 -f mpegts udp://127.0.0.1:50010 [20:13] <b0ot> where would I add the re tag [20:14] <burek> if you are capturing a live event (camera) you don't need -re [20:14] <burek> what do you mean by "30x speed" [20:15] <burek> try something more realistic, like ffmpeg -f video4linux2 -i /dev/video0 -vcodec libx264 -crf 20 -f mpegts udp://127.0.0.1:50010 [20:15] <burek> because your cmd didn't use encoding at all, it streamed raw video [20:16] <b0ot> unknown encoder [20:16] <b0ot> libx264 [20:16] <burek> hm [20:16] <burek> did you apt-get install ffmpeg or compiled [20:17] <burek> type ffmpeg -codecs to see all possible (V)ideo codecs [20:17] <burek> and try with i dunno, something like -vcodec mpeg1v or something [20:17] <burek> i never use anything beside h264 [20:17] <burek> it's really supreme [20:18] <b0ot> I still see like .75 second lag [20:18] <b0ot> maybe 0.5 sec [20:18] <b0ot> can I get it lower? [20:20] <burek> the encoding takes that long [20:20] <burek> so I guess no [20:20] <burek> unless you are on fiber optic [20:20] <burek> and you can stream raw video [20:20] <burek> or try lowering the complexity of the encoder [20:21] <b0ot> i haven't done any encoding [20:21] <burek> like, choose lower quality options / faster encodings [20:21] <burek> can you please use pastebin.com, to show your command line and its output? [20:21] <b0ot> still without vcodec [20:21] <burek> i think ffmpeg used some "default" video encoder [20:21] <burek> i really doubt it left it as raw video [20:27] <b0ot> is it 2>&1 ? [20:31] <DynamicFail> burek, (b0ot here) http://paste2.org/p/1919249 [20:31] <b0ot> looks like it used yuv420p [20:42] <burek> DynamicFail, Stream #0.0: Video: mpeg2video, yuv420p, 640x480, q=2-31, 200 kb/s, 90k tbn, 30 tbc [20:42] <burek> ffmpeg uses -vcodec mpeg2video [20:42] <burek> as a default [20:47] <b0ot> lol adding -re [20:47] <b0ot> causes the remote vlc to crash every time i open it [21:07] <burek> why would you add -re :) [21:07] <burek> it's a live stream [21:07] <burek> it cannot physically read the input faster [21:07] <burek> anyway, why do you need even lower lag than 0.5 seconds? [21:08] <burek> you can also try changing q for mpeg2video [21:09] <burek> from 2-31 [21:09] <burek> i don't know exactly which one is faster, but you can play around [21:23] <rainmaker1> Hi, are there any tools that can help in audio/video beeing out of sync? [21:23] <rainmaker1> Basicaly I am looking for a tool that will allow me to compare the original and transcoded file... [21:24] <rainmaker1> timestamps and so on [21:28] <burek> avisynth [21:29] <burek> kdenlive [21:29] <burek> i guess you are looking for a video editor [21:30] <rainmaker1> burek: Ah :) [21:30] <rainmaker1> burek: I saw that you have reported some issues with audio/video beeing out of sync.. [21:31] <rainmaker1> burek: So I was wondering how come that ffmpeg can actually generate out of sync files when input file is perfect? [21:31] <burek> it's magic :D [21:31] <rainmaker1> burek: I belive so :) [21:31] <rainmaker1> s/belive/believe [21:32] <burek> :) [21:32] <rainmaker1> burek: yes yes magic like abracadabra, but is it ffmpeg bug or it has something with outside libs? [21:33] <Halexander9000> Greeting gentlehumans! I'm desperately trying to make sense of things here. I have this .ogv file that uploads and plays without a problem in youtube. Because it's recorded with RecordMyDesktop, with absolutely no editing done. Now, I want to edit said file. Surprize, surprize, OpenShot video editor can't even play the file without video errors, much less export it into a similar or different format without the same video errors. I tried to convert [21:34] <Halexander9000> Does it get cut off? [21:38] <burek> rainmaker1, did you create video with ffmpeg [21:38] <burek> it does [21:39] <burek> if you like essays you can use pastebin.com :) [21:39] <burek> just narrow it down to the problem :) [21:39] <rainmaker1> burek: well it's a live stream that I have recorded to a file without transcoding [21:41] <rainmaker1> burek: now I want to transcode that file to h264 because of the circular buffer overun problem in udp.c you know.. the bug that nobody want to fix :D [21:41] <Halexander9000> burek: Are you talking to me or rainmaker1? [21:42] <rainmaker1> Halexander9000: to you and to me ;) [21:43] <Halexander9000> Greeting gentlehumans! I'm desperately trying to make sense of things here. I have this .ogv file that uploads and plays without a problem in youtube. Because it's recorded with RecordMyDesktop, with absolutely no editing done. Now, I want to edit said file. [21:43] <Halexander9000> Surprise, surprise, OpenShot video editor can't even play the file without video errors, much less export it into a similar or different format without the same video errors. I tried to convert it using this guy's tutorial http://www.youtube.com/watch?v=ChgTqsZM6UM . Same result. [21:43] <Halexander9000> Then I tried this guy http://www.markandey.com/2010/09/how-to-fix-ogv-file-to-upload-on.html . Exactly the same video errors. Those same tetris pieces of scrambled video data covering the text and edges of objects contained within the video. I really need some advice on how to convert it properly. Help please? [21:44] <sacarasc> Halexander9000: If you use ffmpeg instead of recordmydesktop, you won't have to reencode and lose more quality. [21:45] <Halexander9000> Um... So, I need to record that session again? [21:46] <sacarasc> Or, if you don't want to do that and just want to reencode, it would be good to know what your editing program supports. [21:46] <Halexander9000> I don't want that, it's a game session. I want to convert the recording into something managable by openshot or kdenlive. [21:46] <burek> rainmaker1, decode your streams to uncompressed versions (raw audio/video) and combine them (also correcting the sync) and recompress them [21:46] <Halexander9000> sacarasc: I tried to reencode into avi. No matter what I try, I get the same video artifacts. [21:46] <burek> heck dont do that [21:47] <burek> what i wanted to say is just extract both streams without touching them [21:47] <rainmaker1> burek: why should I do that, when my streams are perfect? [21:47] <burek> -vcodec copy and -acodec copy (also -vn and -an) [21:47] <burek> well if you want to fix sync issues? [21:48] <rainmaker1> burek: jel zabranjeno ovde na srpkom da se prica? mislim ono nije lepo i to? [21:48] <Halexander9000> sacarasc: I even tried to open the .ogv file in the respective video editing software. It doesn't even play properly in the preview window, much less render without the same bloody tetris mishmash of hellish chaos. [21:48] <burek> Halexander9000, can you just type ffmpeg -i yourfile.ogv [21:48] <burek> can you please use pastebin.com, to show your command line and its output? [21:48] <Halexander9000> burek: That I can. Stand by. [21:49] <burek> rainmaker1, please speak in english, because the channel is being logged [21:50] <rainmaker1> burek: ok, my input stream is perfectly in sync, then I don't know why should I edit it, it's the ffmpeg transcoded stream that's have sync issues [21:50] <Halexander9000> burek: There you go. http://pastebin.com/0KmrsrDG [21:51] <burek> rainmaker1, what is your overall goal? just to transcode? [21:52] <burek> Halexander9000, did you try to remux that into .mkv ? [21:52] <rainmaker1> burek: yes [21:52] <Halexander9000> burek: I would if I knew how. [21:52] <burek> ffmpeg -i test.ogv -f mkv output.mkv [21:52] <burek> or it is -f matroska [21:52] <burek> type ffmpeg -formats [21:52] <Halexander9000> burek: On it. [21:52] <burek> to make sure [21:52] <burek> and yes [21:52] <burek> wait [21:52] <Halexander9000> Yes? [21:52] <burek> ffmpeg -i test.ogv -vcodec copy -acodec copy -f matroska output.mkv [21:53] <burek> something like that will do [21:53] <Halexander9000> I see. Let me see. [21:53] <burek> rainmaker1, can you do ffmpeg -i yourvideo [21:53] <burek> can you please use pastebin.com, to show your command line and its output? [21:53] <burek> and what format/codec do you want to transcode your video to? [21:54] <Halexander9000> burek: http://pastebin.com/QP2fV9Jv [21:55] <burek> Halexander9000, well it did it [21:55] <burek> try opening the video in video editor [21:57] <rainmaker1> burek: http://pastebin.com/XtSDFX54 [21:57] <Halexander9000> burek: close, but now dice. The audio is pulsing on and off every half-second. It's more noticeable when there's no sound playing. [21:57] <rainmaker1> burek: h264 using libaacplus if possible [21:57] <Halexander9000> burek: I'll see what it does to a biggerfile. [21:58] <burek> Halexander9000, try -f mp4 instead of matroska [21:58] <burek> and set out.mp4 [21:59] <burek> rainmaker1, ffmpeg -i hbo.mpg -vcodec libx264 -crf 20 -acodec libaacplus -ar 32k -ac 2 -f mp4 out.mp4 [22:00] <Halexander9000> burek: http://pastebin.com/c6CLXfvr [22:00] <Halexander9000> burek: outputs an empty file. [22:00] <burek> i guess mp4 cannot hold theora/vorbis [22:01] <Halexander9000> I'll try that other thing with the larger file and see what it does. It will take a while, that I'm sure of. [22:01] <burek> well, remuxing the media doesn't change it's quality [22:02] <burek> streams stay the same, i.e. not reencoded [22:02] <burek> so you can type ffmpeg -formats [22:02] <burek> and try various -f ... [22:02] <burek> to see which ones your video editor support [22:04] <Halexander9000> burek: This is starting to get ridiculous. [22:04] <Halexander9000> Alright, how can I use ffmpeg as a screen recorder? [22:04] <burek> Halexander9000, what OS do you use [22:04] <Halexander9000> Ubuntu latest? [22:04] <burek> install kdenlive [22:04] <burek> and try loading video in that editor [22:05] <Halexander9000> burek: Tried it, trust me, it doesn't even play it properly in the preview window, let along render anything of use. [22:05] <Halexander9000> alone* [22:05] <rainmaker1> burek: what is -crf option (docs are pretty mystic)? and why should I set a number of channel when by default is set to the number of input audio channel anyway? [22:06] <rainmaker1> burek: sorry to bother if you have no time, I'll try to google around [22:07] <burek> crf is x264's option [22:07] <burek> just type x264 --help [22:07] <burek> and remove -ac 2 then [22:07] <burek> if it bothers you :) [22:08] <burek> crf is something like quality measure for x264 [22:08] <rainmaker1> burek: I found and reading :) [22:08] <burek> like constant rate factor [22:08] <burek> constant quality [22:09] <burek> rather then using -cqp which is constant quantization parameter [22:10] <mystica555_> Halexander9000: i had a bug in ffmpeg that caused ogvs to play back horribly too.. i either upgraded or downgraded the version [22:10] <mystica555_> forget which [22:10] <rainmaker1> burek: thanks :) [22:10] <burek> :beer: :) [22:11] <rainmaker1> burek: :) [22:11] <rainmaker1> burek: and.. can I ask now a million dollar question? :) [22:11] <Halexander9000> mystica555_: Well, it has to be a downgrade, cause I have all the latest versions on my computer. [22:12] <Halexander9000> mystica555_: So how do I do that? [22:12] <mystica555_> from where? the distro? ffmpeg git source? [22:13] <burek> :) [22:13] <burek> Halexander9000, did you compile your ffmpeg? [22:13] <Halexander9000> burek: I used the Ubuntu Software Center for most is not all my software needs. [22:13] <burek> then you most probably have an old version, because distros do not update ffmpeg frequently [22:13] <Halexander9000> Sometimes I follow tutorials and add ppa sources. [22:14] <burek> multimedia projects change very fast [22:14] <burek> so your best bet is to compile your ffmpeg if you can and have time/will [22:14] <rainmaker1> burek: ok then, cheers :) [22:14] <burek> rainmaker1, go ahead ask :) [22:14] <Halexander9000> burek: Can you issue the commands and let me follow you blindly? [22:14] <burek> don't ask to ask, just ask :) [22:15] <burek> Halexander9000, just a sec [22:15] <burek> first uninstall your ffmpeg [22:15] <Halexander9000> Hold on. [22:15] <Halexander9000> I'll use USC for that. [22:15] <bartus> hello.. I have a problem with compiling ffmpeg against dependencies installed in non default location. Tried PKG_CONFIG_PATH without results. [22:16] <rainmaker1> burek: :) Ok, didn't understand.. any idea how to troubleshoot circular buffer overun error? I think you maybe saw that already, or I already ask you? [22:16] <burek> Halexander9000, do: apt-get install libtheora-dev libvorbis-dev [22:16] <Halexander9000> burek: Now kdenlive got uninstalled as well. [22:16] <burek> what circ. buf. err? [22:16] <burek> kdenlive requires ffmpeg? [22:16] <Halexander9000> Yup [22:16] <burek> oh [22:17] <burek> ^^ [22:17] <burek> isn't life fun :D [22:17] <rainmaker1> bartus: which dependencies? [22:17] <Halexander9000> burek: It's certainly not boring. [22:17] <burek> rainmaker1, what circ. buf. err? [22:17] <burek> nice.. [22:17] <burek> peace at last :D [22:18] <bartus> rainmaker1, x264 and frei0r [22:19] <burek> bartus, did you read ./configure --help and greped for extra-ldflags [22:19] <rainmaker1> burek: this one http://ffmpeg.org/trac/ffmpeg/ticket/1017 [22:20] <bartus> rainmaker1, now i see ;) [22:21] <rainmaker1> bartus: I believe that burek answered your q :) [22:21] <Halexander9000> Burek, I'm done. Now what? [22:21] <bartus> rainmaker1, sorry ;] may bad ! [22:21] <rainmaker1> bartus: np :) [22:21] <bartus> burek, thanks ! [22:21] <burek> Halexander9000, do you know how to compile things? [22:22] <burek> rainmaker1, just a sec [22:22] <burek> bartus, :beer: [22:22] <Halexander9000> ./config [22:22] <Halexander9000> make [22:22] <Halexander9000> make install [22:22] <Halexander9000> ? [22:22] <burek> yes, do also apt-get install yasm [22:22] <burek> and then go here [22:22] <burek> http://www.videolan.org/developers/x264.html [22:22] <burek> grab the source and compile it [22:22] <Halexander9000> On it. [22:22] <burek> rainmaker1, did you read latest comments [22:23] <rainmaker1> burek: sure [22:23] <burek> did you do "top" while encoding [22:24] <rainmaker1> burek: but, first, it's not true that system is to slow, and second, the problem is fixed so ffmpeg dont crash, just exit, but with the same error [22:24] <rainmaker1> burek: sure [22:24] <burek> I see [22:25] <rainmaker1> burek: and the best of all is that ffmpeg stops after random period .. sometime after 30 min, sometime after 8 hours, sometime after 15 and so on... [22:25] <burek> hm you didnt use any of x264 options [22:25] <burek> so it just defaulted [22:26] <rainmaker1> burek: what? [22:26] <rainmaker1> burek: it's not my ticket [22:26] <burek> try setting some tighter options like -crf 25 (to speed it up a little) [22:26] <burek> or -preset ultrafast [22:26] <burek> oh I see [22:26] <burek> well, try to speed it up [22:26] <burek> just to make sure it's not a speed related issue [22:26] <rainmaker1> burek: it's another user that has the same problem [22:26] <burek> I understand that [22:27] <burek> I'm telling you to rule out that possibility by using these options above [22:27] <rainmaker1> burek: sure, I will try, it bugs me so long that now I don't know what I have used... [22:27] <burek> I know the feeling.. :) [22:28] <rainmaker1> :D [22:29] <burek> also, dont use -re when your input is already in real time [22:29] <burek> like streaming or a webcam [22:29] <burek> its only useful with files, so that ffmpeg doesn't rush with encoding [22:29] <burek> it usually rushes to encode things as fast as possible [22:30] <burek> so you need to tell it to slow down for it to be "watchable" :) that's the purpose of -re [22:30] <rainmaker1> burek: I learned that few days ago when ffmpeg crashed every few minutes :))) [22:30] <burek> :) [22:30] <Halexander9000> Burek: Cannot install [22:30] <burek> cannot install what [22:30] <Halexander9000> Burek: I'll give you the log. [22:31] <burek> rainmaker1, when using -i udp://... it is usually a good idea to prefix that with -f telling ffmpeg what container it is [22:31] <burek> like -f mpegts -i udp... [22:31] <burek> Halexander9000, [22:31] <burek> can you please use pastebin.com, to show your command line and its output? [22:31] <rainmaker1> burek: I did that also in the latest iteration [22:31] <Halexander9000> I was doing just that. [22:31] <burek> ok [22:32] <burek> rainmaker1, also try using -an instead of c:a libfaac [22:32] <burek> to remove audio, just to see if the problems come from the syncing [22:32] <rainmaker1> burek: does it make sense to use maxrate or -b ? [22:32] <Halexander9000> burek: http://pastebin.com/RySt91g3 [22:33] <burek> it does if you want to control your output using bandwidth limitation :) [22:33] <Halexander9000> it wont make install [22:33] <burek> usually -b is what you want rainmaker1 [22:33] <burek> :) [22:33] <burek> its not config [22:33] <burek> oh you got that [22:34] <Halexander9000> Doesn't matter. [22:34] <Halexander9000> I tried sudo make install. So far, I think it worked. [22:34] <Halexander9000> How can I check that? [22:34] <burek> No command 'instal' found, did you mean: [22:34] <burek> Command 'install' [22:34] <rainmaker1> burek: I mean between those two -b or -maxrate ? [22:34] <burek> try sudo make install [22:35] <Halexander9000> burek: I think I figured it out. [22:35] <burek> install: cannot create regular file `/usr/local/bin/x264': Permission denied [22:35] <Halexander9000> burek: Read what I just said. [22:35] <burek> rainmaker1, http://ffmpeg.org/ffmpeg.html [22:35] <Halexander9000> burek: I got it, on my own. [22:35] <burek> Halexander9000, ok [22:36] <Halexander9000> burek: Now how do we move on, or would you like to issue a command to check my work? [22:36] <Halexander9000> Now do we move on* [22:36] <burek> ok now, you have x264 [22:36] <burek> theora, vorbis [22:36] <burek> what else do we need [22:36] <Halexander9000> Kdenlive? [22:37] <burek> we'll get to that later [22:37] <rainmaker1> burek: -maxrate is vbr and -b is cbr or ? [22:37] <burek> maybe we won't even need it [22:37] <burek> Halexander9000, http://ffmpeg.org/download.html [22:37] <Halexander9000> I might need it later when I add audio clips to the file. [22:37] <burek> now if you want to be always at the bleeding edge with ffmpeg [22:37] <burek> you'll naturally want to use git [22:38] <VooDooNOFX_> Where can I read up on cuda/ffmpeg progress/testing/implementation, etc? [22:38] <Halexander9000> Every time it updates? [22:38] <burek> and when you like, you just make uninstall older ffmpeg make distclean and git pull for new version [22:38] <burek> if not [22:38] <Halexander9000> Isn't there a script or something to make that automatically? [22:38] <burek> just download the current snapshot and compile it [22:38] <burek> it is automatically :) [22:38] <burek> rainmaker1, just a sec [22:39] <Halexander9000> I'll download the git then... [22:39] <burek> rainmaker1, can you ask that on #ffmpeg-devel [22:39] <burek> just apt-get install git [22:39] <burek> when it is finished: git clone git://source.ffmpeg.org/ffmpeg.git [22:39] <burek> it will create a directory named ffmpeg in your current dir [22:40] <burek> VooDooNOFX_, cat CHANGES ? :) [22:40] <rainmaker1> burek: sure I hope they don't get angry about it [22:40] <Halexander9000> burek: Guess I should move back to the home directory. [22:40] <burek> rainmaker1, me too ^^ [22:40] <burek> Halexander9000, what ever you find suitable for you [22:40] <burek> that directory will hold the ffmpeg's source code with all the history [22:41] <rainmaker1> burek: then I wont ask, I'll try to find out :) [22:41] <burek> when it is done you'll just type: ./configure --enable-gpl --enable-nonfree --enable-libx264 --enable-libtheora --enable-libvorbis [22:41] <burek> make and make install [22:41] <burek> and that's it [22:41] <burek> rainmaker1, clever idea :D [22:41] <Halexander9000> burek:50%... [22:42] <Halexander9000> Wait, did you just say that my home directory with have all those files dumped into it? [22:42] <Halexander9000> burek: ? [22:43] <rainmaker1> burek: yeah :) [22:44] <burek> no Halexander9000 [22:44] <burek> git clone will create a sub-directory [22:44] <Halexander9000> burek: Hallelujah! [22:44] <Halexander9000> So, I take it I need to move into that subdirectory. [22:44] <burek> :) [22:45] <burek> and type that configure line above [22:45] <Halexander9000> Sure. [22:45] <Halexander9000> burek: I can't even begin to thank you for your time so far. [22:46] <Halexander9000> ERROR: libx264 not found [22:46] <Halexander9000> If you think configure made a mistake, make sure you are using the latest [22:46] <Halexander9000> version from Git. If the latest version fails, report the problem to the [22:46] <Halexander9000> ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. [22:46] <Halexander9000> Include the log file "config.log" produced by configure as this will help [22:46] <Halexander9000> solving the problem. [22:46] <Halexander9000> Whoops... [22:46] <Halexander9000> Yeah, pastebin. On it. [22:47] <Halexander9000> burek: http://pastebin.com/uCL9hNWj [22:47] <rainmaker1> Halexander9000: where is your x264 command located? [22:47] <burek> hmh [22:47] <Halexander9000> rainmaker1: I haven't the slightest idea. [22:47] <rainmaker1> which x264 [22:47] <burek> did you do make install in libx264 [22:48] <Halexander9000> I did a sudo make install, remember? [22:48] <burek> did it throw any error [22:48] <Halexander9000> burek: http://pastebin.com/Jt98FUXz [22:48] <Halexander9000> burek: Is that usefull? [22:48] <Halexander9000> useful* [22:49] <rainmaker1> if you type "which x264" you can see where is it (if it's compiled) [22:49] <burek> Halexander9000, type ldconfig [22:49] <Halexander9000> rainmaker1: /usr/local/bin/x264 [22:49] <burek> and try again with ffmpeg [22:50] <Halexander9000> Same configure command like you told me, same error. [22:51] <burek> let me think [22:51] <Halexander9000> burek: Hold on. I'll try sudo ldconfig. [22:51] <Halexander9000> burek: Don't worry. [22:51] <burek> try this [22:51] <burek> go back into x264 dir [22:51] <burek> make uninstall [22:51] <burek> make distclean [22:52] <burek> ./configure --enable-shared --enable-static [22:52] <Halexander9000> burek: hold on, please. [22:52] <burek> make && sudo make install [22:52] <burek> ok :) [22:52] <burek> and ldconfig after that :D [22:52] <Halexander9000> Sudo ldconfig did nothing. [22:53] <burek> i think you might need sudo make uninstall too [22:53] <burek> -too [22:53] <burek> :) [22:53] <Halexander9000> So, how do I go higher within the file structure? [22:53] <Halexander9000> I tried cd /usr/local/bin/x264 but it says it's not a directory [22:53] <burek> cd .. [22:54] <burek> cd <space> <dot> <dot> [22:54] <Halexander9000> I went as far as it could go. It still won't accept my cd /usr/local/bin/x264 [22:55] <burek> why would you go there [22:55] <burek> where did you download x264 source [22:55] <burek> go there [22:55] <Halexander9000> Oh... [22:55] <burek> sudo make uninstall; make distclean [22:55] <Halexander9000> Ok ok. [22:56] <burek> it might be a good idea to use git repos for x264 too, just to keep up with the latest version if you like [22:57] <Halexander9000> So, I'll leave it at that and go do the same thing with x264 as I did with ffmpeg? [22:57] <burek> just go 1 folder up [22:57] <burek> remove x264 folder [22:57] <burek> and downloaded tar.gz file [22:57] <burek> and do this: [22:57] <burek> git clone git://git.videolan.org/x264.git [22:57] <burek> then cd x264 [22:57] <burek> ./configure --enable-static --enable-shared [22:57] <burek> make; sudo make install [22:58] <Halexander9000> Wait, I need to go two folders up, then up again so git downloads the subfolder into my user directory like ffmpeg. [22:59] <burek> x264 doesnt have to be in the same place as ffmpeg [22:59] <burek> you can put it wherever you like [22:59] <Halexander9000> Why not? I like to keep things tidy. [23:00] <lucas^> I'm confused, I have ffmpeg version 0.8.5.git-9719f25 on one machine, and it accepts -c as an option [23:00] <lucas^> I have ffmpeg 0.8.9 on another machine, and it refuses to recognize -c [23:00] <burek> lucas^, that's normal [23:00] <burek> you probably have used repos on one of those machines [23:00] <burek> to install ffmpeg [23:00] <burek> (that one doesn't recognize -c) [23:01] <burek> because it's a new option [23:01] <lucas^> okay. how can I tell what sorts of options ffmpeg does recognize? [23:01] <burek> ffmpeg --help ? [23:03] <VooDooNOFX_> lucas^, define "options" [23:04] <lucas^> well, I'd like to support both the recent git builds, and the builds most commonly found on various Linux distributions [23:04] <burek> then use "deprecated" options [23:05] <lucas^> apparently the newer builds reject options like -newaudio and such, preferring -map 0:a [23:05] <burek> well, then it's obviously a model of the app that is changed [23:05] <burek> so you can't really influence that [23:05] <burek> maybe you could use grep with --help [23:05] <burek> to detect what is what [23:06] <lucas^> I guess that's what I'm going to have to do [23:06] <burek> but.. if I were you I would create a specific version for a specific version of ffmpeg [23:06] <lucas^> figured there'd be a changelog of what command-line versions were added when [23:07] <lucas^> I intend to use this program on Windows as well as Linux/BSD, and most people on Windows will just fetch the latest binaries from zeranoe.com [23:07] <burek> you are creating a wrapper? [23:07] <lucas^> of sorts. [23:08] <burek> well create diff versions of it :) say v1.1 is for ffmpeg until that date [23:08] <burek> v1.2 is for ffmpeg of the newer date, etc [23:08] <lucas^> unacceptable [23:08] <lucas^> hell I'm keeping it all in one gigantic file just to avoid confusion of 'where do I put these other files' or 'which one do I run' [23:09] <burek> i see [23:09] <burek> well --help | grep :) [23:10] <lucas^> one other question. -video_br, -video_crf apparently were deprecated long ago [23:10] <lucas^> are there any boilerplate recommendations of what bitrate to pass to libx264 / libvpx? [23:11] <burek> it gets even better [23:11] <lucas^> for one-pass encoding [23:11] <burek> as i told you, the model has completely changed as far as i can tell [23:11] <burek> those params not only are not used anymore [23:11] <burek> but new params are introduced, which are actually natural params for the given lib [23:11] <burek> for example -vpreset is no more [23:12] <lucas^> yeah, now it's just -pre [23:12] <burek> it used files with options and stuff [23:12] <burek> no [23:12] <burek> not even that [23:12] <burek> its -preset now [23:12] <burek> and it gets the preset directly from x264 [23:12] <burek> its own presets, which you can list with x264 --help [23:12] <burek> so, the model has changed a lot and i think you will have a really big headache soon :D [23:13] <lucas^> well, ffmpeg 0.7.x doesn't even honor -video_br [23:13] <burek> we will try to make a static build of ffmpeg [23:14] <burek> with only GPL libs linked [23:14] <burek> and make it something like "regular" or official download [23:14] <lucas^> I wouldn't suppose there's a way to get version numbers of libx264, libvorbis, libfaac, etc. from a static build? [23:14] <burek> and you might reference your wrapper to that [23:14] <burek> why would you need that [23:14] <lucas^> encoder tag in video metadata [23:15] <burek> i dont get it [23:15] <lucas^> my 'wrapper' also automatically tags video, and one of the tags is a list of the tools used to encode the video [23:16] <burek> ok [23:16] <burek> well we can version those static builds with the current date [23:17] <Halexander9000> burek: In case you're wondering, the make command for ffmpeg is still processing. In other words, it's working so far. [23:17] <lucas^> well, no big deal if there's no easy way, just curious [23:17] <burek> Halexander9000, ok :) [23:18] <burek> lucas^, Im just saying that this way you don't have to keep all those back compatibilities [23:18] <burek> just change your script/wrapper as ffmpeg changes [23:18] <burek> and on your download page just refer to those latest builds [23:18] <burek> that way you are always safe [23:19] <lucas^> the problem with that is that people don't generally like having to reinstall a new version of ffmpeg over the top of the one their distribution already includes [23:19] <burek> this is not reinstallation, it's a static build [23:19] <burek> like windows exe file [23:20] <burek> download and run [23:21] <lucas^> that could work [23:21] <lucas^> I'll go with that, thanks for the suggestion. will make things much easier [23:21] <burek> i agree :) [23:29] <burek> Halexander9000, are you alive :) [23:29] <Halexander9000> Yes. [23:29] <Halexander9000> It's still making the thing... [23:29] <burek> how does it progress? [23:30] <burek> hm, on my machine it finished in 4 minutes [23:30] <Halexander9000> Do keep in mind that this is a 2008 netbook. A stock ASUS EEE PC 904 HD. [23:30] <burek> ok :) [23:32] <Halexander9000> It's a 1Ghz, 1GB, and Intel onboard video chip computer. [23:34] <burek> ok :) [23:34] <burek> it will take around 10-15 minutes :) [23:34] <Halexander9000> It's getting there... [23:38] <Halexander9000> It's at the s category. [23:42] <Halexander9000> v... [23:42] <burek> :) [23:42] <burek> :beer: :) [23:42] <Halexander9000> :hotcocoa: :D [23:43] <burek> or that :) [23:45] <Christophe_v_E> Hi, I have no experience at all with FFmpeg but I was told it could serve to my purpose. I'm trying to split up .flv files located in a folder into pieces between 6 to 13 seconds. In a seperate action/trigger these would be randomly put together to create a 2 minute movie file (.flv, .avi, .mov) - doesn't matter. I understand it has something to do with using command lines .. however I'm having a hard time reading the manual t [23:45] <Halexander9000> x... [23:46] <Halexander9000> It cuts off after "hard time reading the manual t..." [23:47] <Christophe_v_E> .. however I'm having a hard time reading the manual to understand how it works at all. - I really don't understand how to get started with it. [23:49] <Christophe_v_E> There's no specific language this has to be done with. It's a small school project I decided to start, I'm not getting any grades on it. (Just to say I'm not trying to leech anybody for grades) [23:49] <VooDooNOFX_> Christophe_v_E, you can do this in almost any language you're comfortable with, including bash/windows shell scripting [23:51] <Christophe_v_E> Yes, but I don't understand how I even run everythin [23:51] <Christophe_v_E> g [23:52] <Christophe_v_E> All I have right now is what I downloaded yesterday. I don't know how to set it up let alone make it do anything. I'd love some pointers on that :s [23:53] <Halexander9000> burek: Now I'mma "sudo make install" this think. [23:53] <Halexander9000> Or, thing* [23:54] <burek> :) [23:55] <Halexander9000> burek: I think I'm done. [23:55] <Halexander9000> burek: Now what? [23:55] <burek> Christophe_v_E, just a little bit of patience please [23:56] <burek> Halexander9000, now [23:56] <Christophe_v_E> Certainly, I'm not trying to push :) [23:56] <burek> does it work [23:56] <burek> did you try ffmpeg [23:56] <Halexander9000> burek: Not yet. I'll scroll back an look at the commands again. Hold on. [23:56] <Halexander9000> and* [23:56] <burek> just type ffmpeg [23:56] <burek> does it work [23:57] <burek> Christophe_v_E, you can run an ffmpeg command like this [23:57] <burek> ffmpeg -i input.flv -ss 10 -t 20 -vcodec copy -vcodec copy output.flv [23:57] <burek> that will produce output file output.flv [23:57] <burek> which starts at 10th second of the original [23:58] <burek> and has a duration of 20 seconds (until 30th second) [23:58] <burek> so you basically need a loop to change -ss param and output file name [23:58] <Halexander9000> burek: ffmpeg: error while loading shared libraries: libx264.so.120: cannot open shared object file: No such file or directory [23:58] <Christophe_v_E> I see, and I would put this code in my CMD? I'm running Windows. [23:58] <burek> ldconfig [23:59] <Halexander9000> burek: On it. [23:59] <burek> Christophe_v_E, Start - run - cmd [23:59] <burek> to open command prompt [23:59] <burek> navigate to ffmpeg directory using CD command [23:59] <burek> http://www.cs.princeton.edu/courses/archive/spr05/cos126/cmd-prompt.html [23:59] <burek> if you are new to the command prompt [00:00] --- Wed Feb 29 2012

1 0

ffmpeg-devel.log.20120227
by burek 28 Feb '12

28 Feb '12

[00:49] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r59affed23c 10ffmpeg/ (doc/eval.texi libavutil/eval.c tests/ref/fate/eval): [00:49] <CIA-17> ffmpeg: eval: add root() to solve f(x)=0 [00:49] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:31] <patrykmoura> Hey guys, im tryin to install ffmpeg in my CentOS, ive install all libraries with disabled-shared and prefix=/home/vamolaco/ [03:31] <patrykmoura> My problem is, ffmpeg keeps installing libs at /usr/lib... ive tried to change the --libdir=/home/vamolaco/lib, but ffmpeg again installed libasound.so.2 (link) in /usr/lib and the libasound.so in the same dir (/usr/lib)... i want to install everything under the prefix ive setted... [03:31] <patrykmoura> Can you guys help me, pls? [03:32] <patrykmoura> now im tryin with enable-static... idk if it ll work... [03:32] <patrykmoura> =( [03:42] <Compn> http://www.whatsmypass.com/the-top-500-worst-passwords-of-all-time [04:47] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r493a86e25b 10ffmpeg/tests/ (7 files in 2 dirs): [04:47] <CIA-17> ffmpeg: FATE: remove a bunch of useless -vsync 0 [04:47] <CIA-17> ffmpeg: No changes in the test results. [04:47] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r87d7a92b62 10ffmpeg/ (11 files in 2 dirs): rawdec: set timebase to 1/fps. [04:47] <CIA-17> ffmpeg: 03Paul B Mahol 07master * rcf6914e27f 10ffmpeg/libavcodec/cdxl.c: [04:47] <CIA-17> ffmpeg: cdxl: remove early check for bpp [04:47] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:47] <CIA-17> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [04:47] <CIA-17> ffmpeg: 03Anton Khirnov 07master * rd43040e28b 10ffmpeg/avconv.c: [04:47] <CIA-17> ffmpeg: avconv: refactor vsync code. [04:47] <CIA-17> ffmpeg: No functional changes, just make it more readable. [04:47] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r126daeb550 10ffmpeg/libavcodec/cdxl.c: [04:47] <CIA-17> ffmpeg: cdxl: set pix_fmt PAL8 only if palette is available [04:47] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:47] <CIA-17> ffmpeg: The code in parse_nal_units would always end up setting has_b_frames [04:47] <CIA-17> ffmpeg: to "1", except when stream is explicitly marked as low delay. [04:47] <CIA-17> ffmpeg: Since the parser itself would create 'extradata', simply reopening [04:47] <CIA-17> ffmpeg: the parser would cause this. [04:47] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r13f6917ca9 10ffmpeg/avconv.c: avconv: set discard on input streams automatically. [04:47] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r574dcb5baf 10ffmpeg/: (log message trimmed) [04:47] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/master' [04:47] <CIA-17> ffmpeg: * qatar/master: [04:47] <CIA-17> ffmpeg: lavf: don't guess r_frame_rate from either stream or codec timebase. [04:47] <CIA-17> (12 lines omitted) [12:29] <cbsrobot> Compn: thew - mine is on place 501 [12:43] <Compn> its 123451 isnt it ? [12:52] <iive> please don't put your passwords on publicly logged channel! [12:53] <Compn> iive: http://www.whatsmypass.com/the-top-500-worst-passwords-of-all-time [12:54] <iive> :) [12:54] <cbsrobot> Compn: no it's: 501 [13:02] <merbanan> :/ my pincode is there [13:04] <Compn> if its easy to type, its easy to use for lots of people :\ [13:04] <cbsrobot> merbanan: 666666 ? [13:04] <merbanan> 1111 [13:04] <Compn> six sixes! :P [13:05] <merbanan> I'll switch to 4444 [13:05] <merbanan> it is at least longer down the list [13:09] <Compn> surprised people dont use their names represented by phone keypad [13:10] <Compn> e.g. 2=abc 3=def 4=ghi 5=jkl [13:10] <Compn> or however that is setup [13:10] Action: Compn somehow forgets phone numbers already [13:28] Action: mystica555_ has his name as his google voice last-4 digits ;) [16:43] <gnafu> iive: I don't know what you're talking about. All I saw was this: "<@Compn> its ****** isnt it ?" [16:53] <Compn> do ho ho [16:53] <Compn> hunter2 ! [16:53] <gnafu> Compn: "******* !"? [16:53] <gnafu> ;D [16:53] <gnafu> That's one of my favorites. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * re36504b5a9 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/drawbox): fate: add test for the drawbox video filter. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * r4d27a5b2c2 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/setdar ref/lavfi/setsar): fate: add tests for the setsar and setdar video filters. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * r47bd0bc461 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/testsrc): [17:23] <CIA-17> ffmpeg: fate: add tests for testsrc. [17:23] <CIA-17> ffmpeg: This test uses the lavfi device and [17:23] <CIA-17> ffmpeg: is ready for more similar tests. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * r220de4fa20 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/overlay): [17:23] <CIA-17> ffmpeg: fate: add test for the overlay video filter. [17:23] <CIA-17> ffmpeg: The test also covers somewhat the split, scale, [17:23] <CIA-17> ffmpeg: pad and fifo filters and the graph parser. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * r9380003087 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/fade): fate: add test for the fade video filter. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * rf2a05174eb 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/pad): fate: add test for the pad video filter. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * rb0de06aee6 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/thumbnail): fate: add test for the thumbnail video filter. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * rd8cf9d8a6c 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/transpose): fate: add test for the transpose video filter. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * r01f1468eff 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/select): fate: add test for the select video filter. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * r48f3d43113 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/unsharp): fate: add test for the unsharp video filter. [17:23] <CIA-17> ffmpeg: 03Nicolas George 07master * ra4c22e3cac 10ffmpeg/tests/ (lavfi-regression.sh ref/lavfi/life): fate: add test for the life video source. [17:47] <teratorn> hiya, if I decode a yuv420p frame from a video, and the frame.width == 320, but frame.linesize[0] == 352, something is definitely screwed up, isn't it? [17:47] <ubitux> padding? [17:47] <teratorn> o.O ? [17:48] <teratorn> linesize includes padding between the planes!? [17:48] <teratorn> err [17:49] <nevcairiel> linesize is the stride or pitch, or whatever you call it [17:49] <nevcairiel> it will most of the time include extra padding from the end of a line to the start of the next [17:50] <nevcairiel> this is especially important if your image width is a rather odd size, but the decoder can only effectively work on mod16 [17:51] <ubitux> looks like a mod64 here, no? [17:51] <iive> it is most likely no-emuedge [17:52] <nevcairiel> 352 is only mod32, 320 is mod64. :p [17:52] <nevcairiel> mod32 is quite common for the y plane of yuv420, because that means u/v are mod16 [17:53] <teratorn> hmmm.. OK then, so I should rely on width/heigh when copying plane data...errr this teams I have to copy plane data line-by-line if I must ignore padding on each line? [17:54] <nevcairiel> if whatever you're copying to cannot cope with padding, then yes [17:54] <teratorn> I'm copying the imagine to an SDL YUV overlay buffer [17:54] <teratorn> its linesizes match he width/height without padding [17:54] <teratorn> http://teratorn.org/code/sdlplayer/ [17:54] <teratorn> nevcairiel, ubitux: thx for help alot [17:55] <teratorn> is there any trick or optimized way to get frames without padding? [17:55] <nevcairiel> not really [17:55] <teratorn> OK then - not a practical concern in my app [17:55] <nevcairiel> usually copying line-by-line shouldnt be any slower really [17:55] <nevcairiel> just a few lines of code more [17:56] <teratorn> memcpy is a macro, and inlined so I guess not [17:56] <teratorn> well maybe not a macro [17:56] <teratorn> but inline [17:57] <ubitux> coverage since nicolas' commit: lavd:1.5%13.3% lavf:21.9%38.5% lavu:71.9%73.4% [17:57] <ubitux> commits* [17:58] <ubitux> (lavd increased because of lavfi device) [18:35] <teratorn> would anyone like to help me fix my image copying problem? I don't know what is wrong with my brain today, but I really need to get this done at work - we would be happy to pay $50 USD even if it takes you 5 minutes [18:35] <teratorn> busted code: http://teratorn.org/code/sdlplayer/ [18:36] <av500> money upfront? [18:37] <teratorn> av500: I'm sorry, no [18:37] <teratorn> av500: and I think I have someone helping me now, thanks [18:37] <av500> what is there to fix? [18:37] <av500> you copy the image line by line [18:37] <av500> and you take stride into account [18:38] <teratorn> it's corrupted in the playback... [18:38] <av500> even ffplay should have code for that [18:38] <av500> then you are copying it wrong [18:38] <av500> ffplay uses SDL as well [18:38] <av500> so see it there [18:38] <teratorn> good tip, thanks [18:39] <av500> it was free :) [18:40] <gnafu> Such a nice guy. [18:40] <gnafu> How uncharacteristic of you. [18:40] <gnafu> ;D [18:40] <av500> right [18:40] Action: av500 invoices gnafu [18:40] <gnafu> D: [18:47] <teratorn> well, ffplay.c is just a little bit complicated - so offer still stands in case anyone wants to have a crack at fixing my C code :) [18:51] <av500> overlay->pitches[0] = frame.linesize[0]; [18:51] <av500> why??? [18:51] <av500> you copy from frame to overlay [18:51] <av500> each with its own stride [18:51] <av500> aka linesize [18:52] <av500> or (i = 0; i < rect.h / 2; i++) [18:52] <av500> for (i = 0; i < rect.h / 2; i++) [18:52] <av500> why /2? [18:53] <av500> ah ok [18:53] <av500> missed the other loop [18:54] <av500> so drop overlay->pitches[x] = frame.linesize[x]; [18:54] <teratorn> hm [18:54] <teratorn> oh god [18:54] <teratorn> that is still in there [18:55] <teratorn> av500: paypal address? [18:55] <av500> it works now? [18:55] <teratorn> told you my brain wasn't working today [18:55] <teratorn> yep [18:55] <nevcairiel> shouldnt do any work on mondays [18:56] <av500> s/mon/week/ [18:56] <teratorn> nevcairiel: I fully agree with that sentiment [18:56] <nevcairiel> however, if you dont work on mondays, tuesday becomes the monday [18:56] <nevcairiel> following that logic, we should all just troll around the web for a living [18:57] <gnafu> nevcairiel: It works for him. [22:23] <CIA-17> ffmpeg: 03Carl Eugen Hoyos 07master * r7b4e1466d4 10ffmpeg/ (ffmpeg.c libswresample/swresample.h): Exit ffmpeg if swr_convert() fails. [22:38] Action: beastd wonders if Libav-devel moderation queue gets regularly reviewed [23:10] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rf1808e3048 10ffmpeg/libpostproc/postprocess.c: [23:10] <CIA-17> ffmpeg: libpostproc: add pp=help support to print the help text via av_log() [23:10] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Tue Feb 28 2012

1 0

ffmpeg.log.20120227
by burek 28 Feb '12

28 Feb '12

[00:03] <cbsrobot> brontosaurusrex_: scale=480:480*ih/iw/sar [00:04] <brontosaurusrex_> cbsrobot: uh, nice :) [00:05] <cbsrobot> brontosaurusrex_: and why cropping afterwards ... ? [00:06] <brontosaurusrex_> cbsrobot: just to get mod(2) [00:06] <cbsrobot> you could also use it on the scale - or will you see the difference ? [00:07] <cbsrobot> and isnt it rather: [00:07] <cbsrobot> floor(ih/2)*2 [00:08] <cbsrobot> brontosaurusrex_: try: [00:09] <cbsrobot> -vf "scale=480:floor(480*ih/iw/sar)*2,setsar=1:1" [00:10] <cbsrobot> sorry [00:11] <cbsrobot> -vf "scale=480:floor(240*ih/iw/sar)*2,setsar=1:1" [00:12] <chipotle> hi [00:14] <brontosaurusrex_> cbsrobot: thats great, care to explain where you get 240 ? [00:15] <cbsrobot> brontosaurusrex_: 480/2 [00:15] <cbsrobot> hehe [00:15] <burek> :) :beer: [00:15] <cbsrobot> or 120*2 [00:15] <brontosaurusrex_> yeah, my math sucks badly i guess [00:16] <cbsrobot> just kidding [00:26] <brontosaurusrex_> wa wa we wa wo wo, i guess i have a working wildfireS script, very nice! [00:45] <brontosaurusrex_> cbsrobot: thanks for the math lesson ;) [00:45] <brontosaurusrex_> night all [00:53] <t4nk256> anyone familair with doby e development status? https://roundup.libav.org/issue937 [04:29] <patrykmoura> hey guys [04:29] <patrykmoura> good nite [04:29] <patrykmoura> im tryin to install ffmpeg in my CentOS, ive install all libraries with disabled-shared and prefix=/home/vamolaco/ [04:29] <patrykmoura> My problem is, ffmpeg keeps installing libs at /usr/lib... ive tried to change the --libdir=/home/vamolaco/lib, but ffmpeg again installed libasound.so.2 (link) in /usr/lib and the libasound.so in the same dir (/usr/lib)... i want to install everything under the prefix ive setted... [04:29] <patrykmoura> Can you guys help me, pls? [04:29] <patrykmoura> This is a way for me, to install ffmpeg in my webhost that doesnt offer support for ffmpeg and donŽt allow ssh access =( [07:08] <kurtis> Hey guys, i'm taking a 128k bitrate mp3 file and trying to re-encode it as .ogg using ffmpeg. For some reason, the output file using the same bitrate is 3 times as large. My command: http://pastebin.com/Vj6HVc3d [07:09] <kurtis> Is this normal? [07:10] <mystica555_> kurtis: you didn't specify the codec [07:10] <mystica555_> ogg is a container; on my end, its defaulting to FLAC [07:10] <mystica555_> you can look for yourself in the output [07:11] <mystica555_> Stream #0:0 -> #0:0 (mp3 -> flac) [07:11] <mystica555_> with your command [07:11] <kurtis> mystica555_: ahh okay. I thought I read somewhere it automatically guesses on the codec to use based off of the file name. but i didn't know there were multiple .ogg encodings. Any suggestions? [07:12] <kurtis> Yep, it went to flac [07:13] <kurtis> particularly I guess, is there certain encodings that should be used for the web? [07:13] <mystica555_> again, ogg is a container; it can contain vorbis, h264, subtitles, heck i bet it could even encapsulate mp3 if someone wanted to create an abomination. [07:13] <mystica555_> it was used before mkv came on the market [07:14] <mystica555_> for web, mp3 or ogg/vorbis are about the most standard you can find [07:15] <mystica555_> the ogg container is just not as flexible as mkv, so everyone changed over. before either, there was no proper multiplex format for arbitrary streams [07:15] <kurtis> nice! Yeah I had no idea it was just a container before. I wonder how the HTML5 marketing is going to work if there's no specific encoding. And I think Vorbis is probably right. Thanks! [07:15] <mystica555_> you could do an mpeg program stream with specific things, or a quicktime file / mp4 stream again with specific things [07:16] <mystica555_> not sure on the html5 end; i know with video they have to browser detect because of firefox not doing h264 and others not doing webm [07:18] <kurtis> Yeah, I'm sure that'll come later :) Do I just throw in the '-acoding vorbis' flag before the output file? [07:18] <kurtis> whoops '-acodec vorbis' [07:23] <kurtis> hmm that throws errors [07:23] <mystica555_> yeah [07:23] <mystica555_> that should work [07:23] <mystica555_> or not [07:23] <mystica555_> im an idiot [07:23] <mystica555_> libvorbis [07:23] <mystica555_> the built in one isnt as good [07:23] <mystica555_> ...latency [07:23] Action: mystica555_ smacks his dsl [07:24] <kurtis> haha sweet, thanks :) yeah here's the errors I got, but I'll try libvorbis instead. http://pastebin.com/M34qh8vn [07:24] <mystica555_> yeah libvorbis should work [07:24] <kurtis> hmm, unknown encoding [07:25] <mystica555_> [NULL @ 0x8f88ba0] Codec is experimental but experimental codecs are not enabled, see -strict -2 [07:25] <kurtis> guessing i need libvorbis-dev and probably a recompile of ffmpeg? [07:25] <mystica555_> thatd be the error [07:25] <mystica555_> ffmpeg gave you 'unknown encoding' ? [07:25] <kurtis> Yeah, I saw that error -- but when I try "libvorbis" I get: Unknown encoder 'libvorbis' [07:25] <mystica555_> ohh [07:26] <mystica555_> youd need to be using a ffmpeg compiled with libvorbis support then yes [07:26] <kurtis> sweet, hopefully that won't be too hard :) [07:26] <mystica555_> *an [07:26] <kurtis> Thanks! [07:27] <mystica555_> you're welcome :) [07:28] <kurtis> Is it possible to optimize ffmpeg at all? [07:29] <kurtis> hmm, I installed libvorbis-devel and re-ran ./configure but it says "libvorbis enabled no" [07:31] <mystica555_> hm.. [07:33] <kurtis> I could try using the "experimental" encoder but that sounds kinda scary :) [07:34] <mystica555_> ive even read some stories about how it makes a valid bitstream but is horrible sounding [07:35] <mystica555_> as for the configure.. [07:35] <mystica555_> i don't know why its not finding it. [07:35] <mystica555_> maybe theres a config cache file [07:36] <kurtis> possibly, I found this page: http://xiphmont.livejournal.com/51160.html which uses ./configure switches to disable built-in vorbis and enable libvorbis, hopefully it'll work [07:39] <kurtis> sweet, i think it's going to work! taking forever to configure though [07:51] <mystica555_> fwiw, ive simply done ./configure before and it ends up finding mine.. [07:51] <mystica555_> which os are you on there? [07:52] <mystica555_> o wait no.. [07:52] <mystica555_> why do i always assume, try, then am wrong. [07:52] <mystica555_> i know mplayer auto-finds everything, and was thinking about that [07:52] <mystica555_> whereas ffmpeg needs some --enable bits.. [07:53] <mystica555_> but as i look at my 'no' list, eh, ive got every one of those bits installed and it all syas no.. why they arent enable by default is unknown to me [07:53] <mystica555_> or, default to 'detect' [10:46] <rpvl> Hi! I'm wondering if it's possible to hardcode srt subtitles in an mp4 video with ffmpeg... ? [10:47] <Mavrik> not by itself [10:47] <Mavrik> ffmbc fork should be able to though [10:47] <Mavrik> (never tried it though) [10:48] <rpvl> humm ok. thanks, I'll give that a try [11:31] <Kartagis> rpvl: I use mencoder+ffmpeg. I could give you the code snippets if you like [11:33] <rpvl> Kartagis: that would be nice :) [11:37] <Kartagis> http://pastebin.com/mKDAgKY0 [11:37] <Kartagis> rpvl: ^^ [11:40] <rpvl> Kartagis: tks, I'll just use the subtitles part, the video and audio may passthrough [11:42] <cbsrobot> rpvl: see the libass filter [11:42] <cbsrobot> if you can convert srt to ass you can hardcode subtitles [11:43] <rpvl> cbsrobot: using only ffmpeg? [11:43] <cbsrobot> http://ffmpeg.org/libavfilter.html#ass [11:44] <cbsrobot> you need to compile it with --enable-libass [11:45] <cbsrobot> how many srt files you have ? [11:47] <rpvl> I just have one... my use case is simple.. I have an mp4 movie.. and I want to hardcode the srt subtitle file into it... to watch it on apple tv [11:47] <cbsrobot> I can convert srt to ass if you want [11:48] <rpvl> I found a ruby script that seems to do it: https://github.com/morr/srt2ass/blob/master/srt2ass.rb [11:50] <rpvl> I just have to compile ffmpeg with the -enable-libass [11:55] <cbsrobot> rpvl: theres a little bug in the srt2ass [11:55] <cbsrobot> I let you guess for a bit [11:56] <cbsrobot> I'll be back a bit later [11:56] <cbsrobot> lets say I think there is a little bug .-) [11:57] <rpvl> hehe ok, I'm compiling ffmpeg now... I'll try the script after :) tks [11:58] <Kartagis> cbreak: it doesn't convert? [12:01] <ubitux> ffmpeg -i a.srt a.ass doesn't work? :) [12:01] <ubitux> then you can hardsub the a.ass with -vf if you --enable-libass at compile time [12:02] <rpvl> ubitux: awesome... it works :) [12:02] <Kartagis> hmm [12:02] <rpvl> the conversion to ass I mean [12:02] <rpvl> I'll have to check the syntax to use with -vf [12:04] <Kartagis> ubitux: with .ass, I can directly hardcode subtitle into .mp4 files? [12:05] <cbsrobot> ubitux: thats even faster [12:05] <cbsrobot> ubitux: btw in my expereince most converters ignore the following sentence in the nice ass-spec.doc : [12:05] <cbsrobot> Note that there is a single digit for the hours! [12:06] <cbsrobot> rpvl: ^ this was the little bug btw .... [12:06] <rpvl> eheh ok [12:13] <greg_f> Hello to all!\ [12:19] <rpvl> I tried with > ffmpeg -i movie.mp4 -acodec copy -vcodec copy -vf ass=subs.ass output.mp4 [12:19] <greg_f> I am writing service to broadcast video/audio in hls. I use ffprobe to test ts files like ffprobe -loglevel debug media0.ts and got some errors. Can somebody explain me what they mean? http://dpaste.com/708439/ [12:21] <cbsrobot> rpvl: I think streamcopy wont work with filters [12:21] <Mavrik> rpvl, just copying the video won't change it :) [12:21] <Mavrik> rpvl, you'll have to re-encode [12:32] <sasishiju> On Fedora 16 for x86 machine, I did yum install ffmpeg-devel. Then I built ffmpeg. But I get NULL avcodec_find_decoder(CODEC_ID_H264); [12:33] <sasishiju> I get NULL as return for avcodec_find_decoder(CODEC_ID_H264); [12:33] <sasishiju> Why is it so? I do av_register_all and avcoedc_register_all [12:33] <sasishiju> as initialization code [12:34] <Mavrik> your ffmpeg isn't build with libx264 [12:35] <sasishiju> I built x264 separate [12:35] <sasishiju> I have a separate source base for x264 and ffmpeg.. I build them separate... [12:39] <sasishiju> Mavrik. I built x264 and ffmpeg separate. Thats how we do for both Windows and Mac .. Now I did the same for Linux [12:39] <Mavrik> well, you need to build ffmpeg so it'll link against libx264 [12:40] <Mavrik> with --enable-libx264 as parameter to configure script [12:40] <cbsrobot> sasishiju: or do it the git clone way .... [12:41] <sasishiju> cbsrobot: and Mavrik: I did git clone of the source bases... [12:41] <sasishiju> of both of them and did configure and make for both of them.... [12:42] <cbsrobot> why did you yum install ffmpeg-devel then ? [12:42] <rpvl> Mavrik: damn.. that will take like.. ages... its a 720p movie [12:42] <sasishiju> I didnt use --enable-libx264.. I thought it was for using x264 binaries available in ffmpeg libraries...... [12:46] <sasishiju> cbsrobot: I thought yum install was for meeting the dependencies for the code base to be built..... [12:48] <sasishiju> cbsroot: should I do yum uninstall ffmpeg-devel and try? [12:48] <cbsrobot> sasishiju: yeah [12:52] <Mavrik> rpvl, well, how do you expect to change encoded video frames without reencoding? :) [12:52] <Mavrik> rpvl, you can always do soft-subtitles [12:53] <rpvl> Mavrik: well... I want the movie to run on an Apple TV.. so.. soft subtitles are not an option I guess.. [12:53] <Mavrik> hmm [12:53] <Mavrik> I know iPhone/iPad support soft subtitles in .mp4 files, why wouldn't Apple TV? [12:54] <rpvl> What are the requirements? do you know? I'm new to appletv :) [12:54] <sasishiju> cbsroot:confused... in fact i was wrong.. there was no result to my previous "yum install ffmpeg-devel" [12:55] <sasishiju> cbsroot: So there is no need to do yum erase [12:56] <sasishiju> My doubt is - in windows code, i used avcodec_init(); av_register_all();avcodec_register_all(); [12:56] <sasishiju> In Linux Fedora 16, for the latest git source, avcodec_init doesnt compile [12:56] <Mavrik> rpvl, I used handbrake to put srts in for iPhone.... [12:57] <sasishiju> Anything wrong in my initialization? [13:10] <sasishiju> Mavrik, --enable-libx264 , is it a must even if I build x264 separate? [13:11] <sasishiju> My problem is NULL as return value for avcodec_find_decoder(CODEC_ID_H264); [13:16] <Mavrik> yes. [13:16] <Mavrik> of course it is [13:21] <ubitux> < Kartagis> ubitux: with .ass, I can directly hardcode subtitle into .mp4 files? // if you want to burn the subtitles -vf ass=bla.ass, otherwise ffmpeg -i bla.ass to get it muxed [13:22] <ubitux> < cbsrobot> Note that there is a single digit for the hours! // it looks like ffmpeg is using a single digit: return snprintf(str, strlen, "%d:%02d:%02d.%02d", h, m, s, ts); [13:23] <cbsrobot> ubitux: ffmpeg is but most other converters not [13:24] <cbsrobot> and vf ass will fail on two digit hours [13:24] <cbsrobot> let's say libass will fail .... [13:24] <ubitux> parsing is done by libass [13:25] <ubitux> the ffmpeg filter only deals with the api and blit the bitmaps [13:25] <cbsrobot> ubitux: I know - not sure what to do about it. [13:26] <ubitux> fix the softwares outputing invalid files [13:26] <cbsrobot> Normally I report it to the makers of the converters .... [13:26] <ubitux> and maybe make libass handle it with a warning [13:39] <MindSpark> hi, can someone tell me what the parameters are for android video encoding? I am trying to convert an rtmp stream to rtsp so it would display on an android device [13:40] <MindSpark> but it's only playing audio (black video) [13:41] <Mavrik> depends on device [13:41] <Mavrik> for older you need to keep h.264 in baseline profile up to level 3.0 [13:44] <MindSpark> where do I get these profiles? I compiled ffmpeg from scratch and I cannot find a place where the profiles are located [13:44] <MindSpark> and what do you mean with level 3.0? [13:45] <Mavrik> MindSpark, profiles, levels: http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Profiles [13:45] <JEEBsv> profiles = H.264 feature sets [13:45] <JEEBsv> levels = how much memory the decoder will need to decode the stream [13:45] <JEEBsv> basically ways to have hardware makers do more simple stuff [13:45] <Mavrik> android supported media formats: http://developer.android.com/guide/appendix/media-formats.html [13:46] <JEEBsv> and easy ways of making stuff for various hardware [13:46] <MindSpark> ok, I am going to try to explain what I am trying to do here, because I am not sure if this is even possible [13:46] <Mavrik> encoding h.264 (and profiles/levels) with ffmpeg: http://www.virag.si/2012/01/web-video-encoding-tutorial-with-ffmpeg-0-9/ [13:46] <MindSpark> there's a channel stream in rtmp which I need to convert to rtsp to stream on a phone [13:47] <MindSpark> I am trying to do this on a server using ffserver [13:47] <MindSpark> not on the phone itself [13:47] <MindSpark> and then requesting the stream from that server [13:47] <JEEBsv> have fun, ffserver is not documented and not really known by many [13:48] <MindSpark> JEEBsv, what do you recommend to achieve something like that? [13:49] <MindSpark> I've been trying to get this done for over a week [13:49] <Mavrik> I couldn't get ffserver to work wht h264 yet... the latest documentation still talks about asf files [13:49] <Mavrik> I'm not even sure if ffserver can stream h264 [13:49] <JEEBsv> it should be able to do a lot, but after you do all the required voodoo [13:49] <JEEBsv> MindSpark: I have no idea about reading rtmp, but generally if you want human-usable streaming most people use VLC [13:50] <Mavrik> *sigh* [13:50] <Mavrik> guess I'll have to take a weekend and write something myself :P [13:50] <JEEBsv> VLC on command line should work for most cases [13:50] <MindSpark> Mavrik, I think it can, I actually got it to work using moboplayer on android [13:50] <JEEBsv> and it's much less voodoo-filled than ffserver [13:51] <Mavrik> I've had serious memory leak problems with VLC few versions back [13:51] <JEEBsv> vOv [13:51] <JEEBsv> I've only used VLC usually a few hours in one go [13:52] <Mavrik> anyway, we're licensing Wowza right now, but am still looking for solutions for home/NPO use that are opensource/free :\ [13:52] <MindSpark> I am looking at vlc right now. Is there a difference between RTP and RTSP? sorry newbie here [13:52] <Mavrik> MindSpark, yeah, it is [13:52] <Mavrik> I suggest avoiding RTP, it's not tivial to set it up [13:53] <MindSpark> for streaming, there's http,rtp,udp (?) and mmsh [13:53] <JEEBsv> also, do note that you usually want the newest VLC with the newest libraries, currently 2.0 is newest with relatively OK'ish aged libraries [13:53] <JEEBsv> I think VLC could do rtsp as well [13:53] <Mavrik> yes, VLC can do RTSP [13:53] <Mavrik> documentation is very arcane, not to mention half of it is obsolete now that 2.0 came :( [13:53] <JEEBsv> yeh [13:54] <Mavrik> anyway, I'll stop whining now [13:54] <JEEBsv> but that's still "more usable" than ffserver [13:54] <JEEBsv> lol [13:54] <Mavrik> need to figure out how to grab 720p from these blackmagic devices [13:54] <JEEBsv> I still remember when I looked at ffserver the first time [13:54] <Mavrik> (very appropriate name for a company that makes video hardware) [13:54] <JEEBsv> not gonna look at it another time [13:54] <MindSpark> so what are my other options? [13:54] <MindSpark> would wowza work? [13:56] <cbsrobot> Mavrik: how do you do taht ? [13:58] <Mavrik> cbsrobot, ? [13:58] <Mavrik> MindSpark, yes, but wowza is rather expensive for such use [13:58] <Mavrik> MindSpark, I suggest you look into VLC 2.0 [13:58] <cbsrobot> grab form bm devices [13:59] <MindSpark> Mavrik, I am downloading vlc as we speak :) can it be run on a linux server from cli ? [13:59] <Mavrik> cbsrobot, ffmpeg via directshow for now [13:59] <Mavrik> MindSpark, yeah... I'm just not sure if it can act as a streaming server [13:59] <JEEBsv> it can [14:00] <JEEBsv> you can basically build VLC without the GUI and have it do the usual streaming stuff [14:00] <JEEBsv> and run it without one even if you build it with it [14:01] <Mavrik> ah [14:01] <cbsrobot> Mavrik: https://github.com/lu-zero/decklink-ffmpeg but i heard it will not work with latest ffmpeg [14:01] <Mavrik> cbsrobot, I managed to grab PAL input with latest ffmpeg and intensity shuttle [14:01] <Mavrik> there's something wrong with dshow input plugin when trying to do 720p though, I'll probably have to patch ffmpeg [14:06] <MindSpark> but if I was to use ffserver, would that be possible? [14:07] <MindSpark> I just don't want to regret the week I worked on this [14:07] <JEEBsv> possibly, just that no-one knows how many goats you have to sacrifice for it to work [14:08] <JEEBsv> arcane magic that works with the underwater power of the Cthulhu [14:10] <Mavrik> also, check for planetary alignment, Saturn could have some effects on your succes [14:31] <rainmaker1> Hi, can ffmpeg transcode rtmp output to multicast address? [14:51] <kevin_> GOOD MORNING [14:51] <kevin_> :} [14:51] <kevin_> oh, i never new amsg went across networks. hrm [14:51] <kevin_> knew* [17:27] <teratorn> hiya, if I decode a yuv420p frame from a video, and the frame.width == 320, but frame.linesize[0] == 352, something is definitely screwed up, isn't it? [18:13] <Tostada> Hi.. I tried to upgrade ffmpeg to 0.10 and I get a couple errors. LIBAVFILTER_2 not defined in file libavfilter.so.2 with link time reference [18:14] <Tostada> And this one: WARNING: library configuration mismatch.. Thanks. [21:03] <GTRsdk> How do I convert m4v -> AVI? [21:05] <mystica555_> starting out with something as simple as ffmpeg -i file.m4v file.avi [21:06] <mystica555_> adding perhaps some useful codec and bitrate options in the mix [21:06] <mystica555_> (after the -i before the output name) [21:06] <mystica555_> well after the -i inputfile rather [21:07] <GTRsdk> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [21:09] <mystica555_> could you pastebin the entire command and its output? [21:28] <rainmaker1> vadim_: Hi, no luck this time... after 13 hours ffmpeg exited... [21:28] <rainmaker1> vadim_: did you saw the post on youtube about circulr buffer overun error? [21:29] <vadim_> rainmaker1: hi... nope, give me the link please [21:29] <rainmaker1> vadim_: http://www.google.com/support/forum/p/youtube/thread?tid=35c3e3d1c2ea6f1f&h… [21:30] <rainmaker1> vadim_: the strange thing is that at the end of post he claims that everything works just fine using -maxrate switch [21:31] <rainmaker1> vadim_: however when I tried the same it crashed afetr 13 hours, and now I have used -maxrate with -bufsize ... [21:32] <GTRsdk> mystica555_: sorry I don't have it pastebined yet, but it is a multiple chapter file. Could that be the issue? [21:33] <rainmaker1> vadim_: It really looks like memory leak, I will try to dig into the udp.c [21:34] <mystica555_> it really shouldnt be the issue; it sounds like something is wrong like a codec is being chosen that avi cant contain properly or some other bug that really the text would be helpful to see. otherwise its speculation [21:35] <vadim_> rainmaker1: just a minute... have to read it carefully [21:36] <GTRsdk> mystica555_: I'm running Debian 7 (wheezy/testing) http://pastebin.com/HRxXAw9Z [21:37] <rainmaker1> vadim_: take your time :) [21:39] <mystica555_> GTRsdk: try potentially -c:v copy. one question i have, what is the intended use for this avi output that the m4v wouldnt do? [21:39] <GTRsdk> mystica555_: play on windows media player [21:40] <mystica555_> is there a reason you require windows media player, versus any other more capable piece of softare? also, have you tried ffdshow-tryouts for a really useful codec set for directshow? [21:41] <mystica555_> at best, i can only say mess around with the codec options, or wait for someone who knows a bit more about what the particulars of the error mean.. im still not quite sure why its not working.. [21:41] <GTRsdk> mystica555_: well there's application that requires windows media player to be able to view it in order for the video to be converted to 3D [21:41] <mystica555_> try ffdshow-tryouts [21:41] <mystica555_> that should allow it to open in media player, if it'll open in anything directshow [21:43] <GTRsdk> the movie was overwritted D: [21:43] <GTRsdk> now it is a blank file ;( [21:44] <mystica555_> ouch [21:45] <mystica555_> anything you put on the commandline that appears to be a media file, but without a -i immediately preceeding, is considered to be an output [21:45] <mystica555_> perhaps a previous command tried to output to both files [21:45] <mystica555_> :\ [21:46] <vadim_> rainmaker1: well... as i remember, i have tried -maxrate parameter, not sure how long it worked... if i'm still here, it worked not long enough... [21:48] <mystica555_> GTRsdk: i sort of wish ffmpeg had a -o to specify output in this circumstance.. [21:48] <mystica555_> and a media file without any -i or -o is just an error [21:48] <rainmaker1> vadim_: don't know what to do next except to looking in code.. [21:49] <vadim_> rainmaker1: even if with 1 to 1 stream it work more or less long (let's say up to 13 h)... when i try to get 1 to many streams it crash very fast... [21:50] <vadim_> rainmaker1: we definitly need a programer with experience in *nix programming, and with a fresh look at this code... [21:54] <rainmaker1> vadim_: with two output streams for me it worked abouth 21 hours :) which is again bad.. :) [21:54] <rainmaker1> vadim_: do you know someone? :D [21:54] <rainmaker1> vadim_: and one very strange thing... how come the [21:55] <vadim_> rainmaker1: not yet... :( [21:55] <rainmaker1> vadim_: that nobody except us is complaining about this bug? [21:55] <rainmaker1> vadim_: ffmpeg is widley used vlc too [21:57] <vadim_> rainmaker1: you're wrong... i found at least three open tickets on bug tracker regarding this problem, some of them quite old, more than 4 months... [21:58] <vadim_> rainmaker1: there was some else people asking about this problem right here... but, have no idea if they found a solution... [22:01] <vadim_> rainmaker1: do you use v0.10? [22:02] <rainmaker1> vadim_: me wrong? it's impossible :D [22:02] <vadim_> rainmaker1: pardon?.. [22:02] <rainmaker1> vadim_: no, I use git snapshot a few days old [22:03] <rainmaker1> vadim_: why? [22:10] <vadim_> rainmaker1: i can't use v0.10+ with the segmentation patch... [22:11] <vadim_> rainmaker1: this is another problem for me... [22:11] <rainmaker1> vadim_: why you dont use latest git version? [22:15] <vadim_> rainmaker1: i use it... but only to test the udp module... [22:16] <rainmaker1> vadim_: did you happend to get out of sync videos? I mean audio and video are out of sync? [22:17] <vadim_> rainmaker1: yes... it happens to me after about 3-4 hours of transcoding [22:19] <vadim_> rainmaker1: but at this moment, this is not so important [22:19] <vadim_> rainmaker1: the only good thing on this situation, is my english practice )) [22:21] <rainmaker1> vadim_: :D:D:D [22:22] <rainmaker1> vadim_: well the bad thing is that now it takes me couple of hours to repeat the problem so it's gonna to be very slow debug :( [22:23] <b0ot> Does anyone know of any thing in linux that would be good for streaming low latency video over a lossy link with linux? [22:23] <rainmaker1> vadim_: do you know a better way to crash it faster? [22:23] <vadim_> rainmaker1: i'm in the same situation [22:23] <b0ot> I noticed when you stream video with vlc you get a lag of around 2 seconds. [22:24] <rainmaker1> b0ot: which is not bad, but if you want you can use zerolatency option in x264 [22:25] <rainmaker1> b0ot: but keep in mind that 2sec is maybe related to buffering at the other end.. [22:27] <b0ot> rainmaker1, Why does it work with things like ip cameras where you have like zero lag. [22:27] <b0ot> and how do I use the zero latency option [22:28] <vadim_> rainmaker1: i think, i know... but will get you the cli in about 30 minutes... i'm recompiling ffmpeg [22:29] <rainmaker1> vadim_: that would be great [22:29] <rainmaker1> b0ot: -tune zerolatency [22:30] <rainmaker1> b0ot: again, it depends if you have buffering on the other end... [22:31] <b0ot> Sorry I'm a bit confused, I was told that the people here would be able to help. Is this using some program to do the streaming like vlc? [22:31] <b0ot> or is the "program" x264 [22:31] <b0ot> And what type of viewer would be best to result in the lowest latency for viewing [22:33] <rainmaker1> b0ot: the program is ffmpeg it uses x264 lib in order to transcode incoming stream to h264 but, it could be used for various other tasks [22:34] <rainmaker1> b0ot: don't know about viewer, and I am not the expert about anything related to ffmpeg or x264 [22:34] <b0ot> So I could take an rtp video stream transcode it with ffmpeg [22:34] <b0ot> and stream it out multicast to my network? [22:36] <tclarke> can ffmpeg encode redcode files? [22:36] <tclarke> if not, does anyone know of a library or tool that can? [22:37] <rainmaker1> b0ot: I guess, but I didn't do it, maybe someone else can help [22:37] <VooDooNOFX_> tclarke, as in the red file format used in RED cameras? [22:38] <tclarke> yes...I have large vvideo (8kx8k) I'm trying to encode and I need a format widely supported by NLVE tools [22:38] <tclarke> red seems a good choice if I can find an encoder [22:38] <tclarke> I'm open to other ideas as well [22:38] <VooDooNOFX_> why not just use the red command line tools from RedCine(x)? [22:40] <tclarke> I'll look into that...depends on the licensing but that could work [22:40] <tclarke> thanks [22:40] <b0ot> Does anyone have any information on how to take an rtsp video stream and use ffmpeg to restream that out (low latency as possible) mulitcast across a network. [22:41] <VooDooNOFX_> tclarke, i'm pretty sure that ffmpeg can't really read red's raw sensor data. I'm no expert on it though. If you find it can, please let me know! [22:42] <VooDooNOFX_> btw, 8k red images? that's a pretty high end camera dude [22:42] <tclarke> it's not camera data...it's synthetic [22:42] <VooDooNOFX_> sounds like a major release post production thing [22:42] <tclarke> http://ffmpeg.org/general.html lists r3d as a supported decode format [22:43] <finsternis> I have a .avi file that does not work right (audio pauses frequently and there are visual artifacts) in vlc 2.0 and older versions of mplayer (do not have newer versions) but it plays well with a very recent ffplay. mplayer omits errors like "ac-tex damaged at". [22:43] <finsternis> I tried ffmpeg -i in.avi -acodec copy -vcodec copy out.avi to fix it, but that caused the .avi to show this problem throughout the file in vlc, but the in.avi only had it starting 10 minutes into it [22:43] <tclarke> VooDooNOFX_: thanks for the pointers...should get me moving a little further [22:44] <finsternis> any suggestions how to further debug and fix this? Goal is to make it work in vlc like it does in ffplay. [22:54] <finsternis> ffmpeg -i in.avi -acodec copy -vcodec copy out.mp4 seems to work [22:57] <finsternis> and turning the mp4 into an avi seems to fix it [23:01] <Takyoji> Any way to aptly cut off from the end of a video, without having to re-encode the whoooole thing? (H.264 in MP4 container) [23:03] <vadim_> rainmaker1: are you still here? [23:04] <rainmaker1> vadim_: yes [23:06] <vadim_> rainmaker1: do you see my pinvitation to private? [00:00] --- Tue Feb 28 2012

1 0

ffmpeg-devel.log.20120226
by burek 27 Feb '12

27 Feb '12

[05:21] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r87840eeb71 10ffmpeg/libavcodec/utils.c: avcodec: disallow reget_buffer() if pix_fmt changed. [05:21] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r830f70442a 10ffmpeg/libavcodec/fraps.c: [05:21] <CIA-17> ffmpeg: fraps: release reference buffer on pix_fmt change. [05:21] <CIA-17> ffmpeg: Prevents crash when trying to copy from a non-existing plane in e.g. [05:21] <CIA-17> ffmpeg: a RGB32 reference image to a YUV420P target image [05:21] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [05:21] <CIA-17> ffmpeg: CC: libav-stable(a)libav.org [05:21] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r33cd32b389 10ffmpeg/libavcodec/kgv1dec.c: [05:21] <CIA-17> ffmpeg: kgv1: use avctx->get/release_buffer(). [05:21] <CIA-17> ffmpeg: Also fixes crashes on corrupt bitstreams. [05:21] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [05:21] <CIA-17> ffmpeg: CC: libav-stable(a)libav.org [05:21] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r19adb0bc2d 10ffmpeg/libavcodec/vorbisdec.c: [05:21] <CIA-17> ffmpeg: vorbisdec: add a flush() function [05:21] <CIA-17> ffmpeg: clear MDCT overlap buffer and reset previous window mode when seeking [05:21] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r2bb628f870 10ffmpeg/libswscale/utils.c: swscale: fix another integer overflow at large dimensions/rescales. [05:21] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8e03912133 10ffmpeg/libavcodec/mpegaudiodec.c: [05:21] <CIA-17> ffmpeg: mpegaudiodec: dont memcpy() more than needed. [05:21] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [05:21] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r904817cb9f 10ffmpeg/libavcodec/cdxl.c: [05:21] <CIA-17> ffmpeg: cdxl: fix ham6/8 on big endian [05:22] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [05:22] <CIA-17> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [05:22] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r159a2436b0 10ffmpeg/tests/ (6 files in 3 dirs): [05:22] <CIA-17> ffmpeg: fate: add tests for cdxl video [05:22] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [10:12] <belegdol> hello, did anything change in ffmpeg 0.10 wrt png support? [10:13] <belegdol> I am trying to build mplayer and it complains about missing ff_png symbols [13:14] <Compn> yeah png changed, but should be fixed in mplayer [13:14] <Compn> if he comes back [13:23] <CIA-17> ffmpeg: 03Derek Buitenhuis 07release/0.10 * r9f82cbf7c1 10ffmpeg/libavcodec/wavpack.c: (log message trimmed) [13:23] <CIA-17> ffmpeg: wavpack: Don't shift minclip/maxclip [13:23] <CIA-17> ffmpeg: Since we are clipping before we shift the values to [13:23] <CIA-17> ffmpeg: 16 or 32 bits, we should not shift the min/max clip [13:23] <CIA-17> ffmpeg: values to compensate. [13:23] <CIA-17> ffmpeg: Fixes 8 and 24 bit lossy decoding. [13:23] <CIA-17> ffmpeg: Fixes ticket #871. [13:24] <CIA-17> ffmpeg: 03Michael Niedermayer 07release/0.10 * rdcde8e1c90 10ffmpeg/libavcodec/wavpack.c: [13:24] <CIA-17> ffmpeg: Revert "Improve decoding quality for lossy wavpack." [13:24] <CIA-17> ffmpeg: This has been implemented more correctly. [13:24] <CIA-17> ffmpeg: This reverts commit a915618a29f3f4197832151a4ed03ccdd585f9cf. [13:24] <CIA-17> ffmpeg: (cherry picked from commit 32e74395a8e88dee1c149aeb36e7a21df431c181) [19:40] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rd7804de351 10ffmpeg/libavformat/cdxl.c: [19:40] <CIA-17> ffmpeg: cdxl demux: do not create packets with uninitialized data at EOF. [19:40] <CIA-17> ffmpeg: Fixes a failure in cdxl-pal8 test for the valgrind FATE instance. [19:40] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [20:42] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r02670762d2 10ffmpeg/ (4 files in 3 dirs): [20:42] <CIA-17> ffmpeg: libavfilter/libmpcodecs: add vf_pp from current mplayer svn [20:42] <CIA-17> ffmpeg: This will allow us to do regression tests on libpostproc. [20:42] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:29] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r923092697a 10ffmpeg/ (doc/eval.texi libavutil/eval.c tests/ref/fate/eval): [22:29] <CIA-17> ffmpeg: eval: Allow specifying the variable id. [22:29] <CIA-17> ffmpeg: Reviewed-by: Nicolas George <nicolas.george(a)normalesup.org> [22:29] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:29] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8c5ee45d13 10ffmpeg/libavutil/eval.c: [22:29] <CIA-17> ffmpeg: eval: Check that single parameter functions dont get a second param. [22:29] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Mon Feb 27 2012

1 0

ffmpeg.log.20120226
by burek 27 Feb '12

27 Feb '12

[00:00] <lucas^> is there a list of those guidelines floating around somewhere? [00:00] <JEEB> anyways, unless you're setting it yourself if you're getting high profile with an ipod-centric preset something's doing something wrong [00:01] <lucas^> going to rerun my script and check to see what the command-line arguments actually are [00:01] <JEEB> lucas^, there's really no guidelines, if it's 3GS level or newer you can just hit it like usual with level 4|4.1 high profile H.264 + AAC, otherwise encode level 3 baseline with 1000kbps vbv-maxrate/bufsize and hope it won't choke [00:01] <lucas^> basically I'm writing a Python program to transcode, remux, tag, sub, clip commercials, etc. etc. from MythTV / WTV recordings automatically [00:02] <lucas^> saw a bunch of iPod options in ffmpeg and figured, what the hell, why not support it [00:02] <lucas^> while I'm here, libvpx crashes immediately in the latest SVN versions of ffmpeg on Windows - is that a known problem? [00:03] <lucas^> don't think it happens in linux/bsd [00:04] <JEEB> someone has mentioned some binaries doing that IIRC, but I don't remember anything else [00:05] <lucas^> could be just a bad build then [00:05] <lucas^> one other question, if you don't mind [00:06] <lucas^> ffmpeg -i concat:foo.mpg|bar.mpg <-- this basically just directly joins the files without doing any other preprocessing? [00:06] <Mavrik> yeah [00:07] <lucas^> it works pretty well actually, but there seem to be some frames dropped, as programs like ccextractor will extract closed-captions which slowly drift out of sync with more clips concatenated [00:07] <lucas^> I can't think of a good way to calculate just how many frames are lost [00:07] <lucas^> the MythTV folks seem to do some additional processing, i.e., the last 10 seconds of foo.mpg and the first 10 seconds of bar.mpg or whatever get converted to i-frames [00:08] <lucas^> is there a way to accomplish this sort of thing with ffmpeg alone? [00:14] <lucas^> ha. found the problem, apparently I had some old preset files lying around [00:25] <lucas^> -intra looks like it does exactly this, awesome [11:06] <TheFuzzball> Hi, I'm trying to implement HTTP 206 partial content for a streaming service and I need to transcode some stuff, is it possible to feed ffmpeg part of a file to convert, or does it have options to transcode only a byte range? [11:07] <Kartagis> hi [11:07] <Kartagis> Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height <--- how to fix this? [11:08] <Kartagis> I've been trying to extract audio tp mp3 from webm [11:08] <Kartagis> s/tp/to/ [11:12] <rav0> what's yoru full command? [11:14] <Kartagis> ffmpeg -i mhld9Pc3HnU.webm -f matroska -ar 128 -ab 44100 Dropbox/Public/Grup\ Seksendort-Simdi\ Hayat.mp3 [11:17] <Kartagis> rav0: ^^ [11:17] <rav0> -ar 128 -ab 44100 [11:18] <rav0> at least that needs to be soemthing like -ar 44100 -ab 128k [11:18] <rav0> swapped, and the k added [11:19] <rav0> er also you are saving a matroska file [11:19] <rav0> to open as matroska, put the -f argument before the -i [11:19] <rav0> simplest example would be ffmpeg -i foo.webm bar.mp3 [11:20] <Kartagis> hrm [11:20] <Kartagis> ffmpeg -f matroska -i foo.webm bar.mp3 ? [11:21] <rav0> yes [11:21] <Kartagis> okay then [11:43] <qxt> I am doing a lot of 2 pass encoding. Was wondering if there is a way to get pass 1 to use threading. Pass 2 automatically kicks threading into gear so no need for -threads 0 there. [11:44] <qxt> A typical cmd would like like this [11:44] <Kartagis> rav0: Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 30.00 (30/1 [11:44] <qxt> ffmpeg -i movieIn.avi -pass 1 -vcodec libx264 -preset veryslow -b:v 1226.72134943k -f matroska -sn -an /dev/null && ffmpeg -i movie.avi -pass 2 -vcodec libx264 -preset veryslow -b:v 1226.72134943k -acodec libfaac -b:a 128k -ac 2 -sn -y movieOut.mkv [11:49] <rav0> sorry, add -vn as well (no video) [11:50] <Mavrik> qxt, as far as I know there are no encoders that do pass 1 analysis in multiple threads [11:51] <rav0> x264's first pass is a regular encode with some features turned off, and a stats file written. i imagine it's threaded like any pass [11:51] <theholyduck> x264 can cant it? [11:51] <theholyduck> yeah its a simple encode [11:52] <theholyduck> x264 usually doesnt need 2 pass though [11:52] <theholyduck> crf makes more sense in almost all cases [11:53] <qxt> Mavrik is this a technical issue. Remember when MT was a fork. As you know MT programming can be a real pain to program. [11:54] <qxt> theholyduck I know its OK but I have to make files that fit onto a medium of lets say 450.0 MB at work. [11:55] <theholyduck> qxt, then 2 pass it is [11:55] <Mavrik> rav0, well, non of the libx264 versions I've used were capable of using different passes for analysis [11:55] <theholyduck> Mavrik, theres more to 2 pass than just pure analysis though [11:56] <Mavrik> what's your point? [11:56] <theholyduck> Mavrik, just because the 2-3 lookahead processes arent threaded individually [11:56] <theholyduck> it still uses your multicore pretty well [11:57] <Mavrik> obviously all my monitoring tools on the transcoding farm are wrong then. [11:57] <theholyduck> it doesnt use it as well as - [11:57] <theholyduck> *--preset veryslow" [11:57] <theholyduck> Mavrik, you're using it through ffmpeg? [11:58] <theholyduck> all my experiences are with x264 itself [11:58] <Mavrik> through ffmpeg and standard x264 (not the newest this one though) [11:58] <Mavrik> it used only a single of 16 cores [11:58] <theholyduck> Mavrik, hmm [11:59] <theholyduck> honestly, i'm not too sure abut how it works in ffmpeg [11:59] <Mavrik> if x264 fixed that now it's probably an ffmpeg issue [11:59] <Mavrik> for us it's impractical to use x264 currently so single-threaded pass 1 is a small price to pay [11:59] <theholyduck> Mavrik, well, previously, with ffmpeg [12:00] <theholyduck> you had to specify threads manually [12:00] <Mavrik> which we do :P [12:00] <theholyduck> to 16? [12:00] <Mavrik> no [12:00] <theholyduck> or to 0? [12:00] <Mavrik> it depends on codec [12:00] <theholyduck> cores*1.5 for xx264? [12:00] <Mavrik> for libx264 the spec was 0 initialy [12:00] <qxt> I guess you could moved the entire video into RAM and not have to worry about issues like that when I think about it. Going to really look into the code when I get some time. [12:01] <Mavrik> but now we're setting it to 4 manually since otherwise the performance is crap [12:01] <qxt> But you will need huge ram [12:01] <theholyduck> Mavrik, hm, 0 usually works the best [12:01] <theholyduck> unless you run multiple encodes at the same time [12:01] <Mavrik> theholyduck, it didn't in my case [12:01] <qxt> and it will save your ssd if you have one [12:01] <Mavrik> it just used alot of time waiting for locks in kernel space [12:01] <theholyduck> Mavrik, hmm [12:01] <theholyduck> weird... [12:01] <Mavrik> so running two transcodes at 4 threads instead of one at 0 threads resulted in way better performance [12:01] <theholyduck> Mavrik, could be io i guess [12:02] <theholyduck> depends on how fast each of your core is [12:02] <Mavrik> that doesn't explain the performance of multiple concurrent transcodes :) [12:02] <Mavrik> plus, the RAID array is pretty fast [12:02] <Mavrik> it's a 3.0 GHz 8-core Hyper-threaded Xeon IIRC [12:03] <theholyduck> Mavrik, hmm indeed. [12:03] <theholyduck> how do you run ffmpeg on it? [12:03] <Mavrik> theholyduck, yeah, it's pretty wierd [12:03] <cbsrobot> Mavrik: can you test something for me on this machine ? [12:03] <Mavrik> what do you mean "how"? [12:03] <qxt> Is there a better way to grab the video_duration then doing this [12:03] <theholyduck> commandline or simelar [12:03] <qxt> video_duration=$(ffprobe -i blah 2>&1 | awk '/Duration/ {gsub(/,/, "", $2); split($2, a, /:/); print 3600*a[1] + 60*a[2] + a[3]}') [12:03] <Mavrik> cbsrobot, it's a production machine, so no... in couple of months when we retire it to staging :P [12:04] <theholyduck> qxt, i think mediainfo has some decent commandline output [12:04] <cbsrobot> with test I mean do you have ffplay installed ? [12:04] <theholyduck> outside of using ffmpeg. i dunno [12:04] <theholyduck> *inside of using ffmpeg [12:04] <cbsrobot> does ffplay recognize the hyperthreading ? [12:05] <qxt> Yeah I am grabbing it out of the mediainfo and then converting it into seconds [12:05] <theholyduck> qxt, mediainfo can atleast output plaintext. xml and stuff [12:05] <theholyduck> qxt, mediainfo as in the app [12:05] <Mavrik> theholyduck, ffmpeg -i <input> -codec:v libx264 -aspect <aspect> -vf scale=<size> -preset slow -sws_flags lanczos -g 75 -threads 4 -acodec libfaac -ab 128k -ac 2 -ar 44100 output.mp4 [12:05] <theholyduck> Mavrik, possibly the scale [12:05] <Mavrik> could be yea [12:05] <theholyduck> ffmpeg scalng is single threaded remember [12:06] <theholyduck> might be libfaac aswell. i cant remember how fast it is [12:06] <qxt> cbsrobot yeah it does if you are using a newer one. I'm on ffmpeg version N-33599-gb008ac1 built on Feb 25 2012 04:53:09 with gcc 4.6.2 [12:06] <Mavrik> libfaac is single-threaded anyway [12:06] <theholyduck> Mavrik, yeah, but provided its fast enough, its fine [12:06] <cbsrobot> qxt what machine ? how many cpus ? [12:06] <theholyduck> lanczos resize is slow and scalingi s single threaded [12:06] <Mavrik> for us it's ok since we have to transcode each file to about 4 or 5 output videos [12:06] <Mavrik> so we just run it in parallel [12:07] <theholyduck> Mavrik, well yaeh, if you hae a single threaded slowdown, its better to do it that way [12:07] <theholyduck> i do the same when i do avisynth encoding [12:07] <theholyduck> or atleast i did [12:07] <cbsrobot> I have a xeon 2x6Core HT and ffplay uses only 12 cores .... [12:07] <theholyduck> i hear the new mt avisynth stuff is pretty decent [12:07] <Mavrik> wonder if they'll let me enroll in GSoC for ffmpeg this year, I could fix some of those issues [12:07] <theholyduck> Mavrik, thread the scaling stuff? [12:08] <qxt> cbsrobot I am on a i7 2700 clocked up to 5Ghz using Debian wheezy [12:08] <theholyduck> ffmpeg has always had a bit stupid interactions with x24 [12:08] <Mavrik> yeah, and fix some dshow input trouble I have with BM devices [12:08] <theholyduck> *x264 [12:08] <cbsrobot> qxt how many cores ? [12:08] <qxt> cbsrobot 4 cores with ht [12:08] <theholyduck> Mavrik, <BugMaster> all passes are multithread. so yes [12:08] <theholyduck> <BugMaster> but at first pass there is slicetype decision which sometimes (--b-adapt 2 and a lot of B-frames) can become bottleneck [12:08] <theholyduck> on x264 first pass threading [12:09] <Mavrik> ah [12:09] <cbsrobot> qxt: thanks - I will try again with the latest version [12:09] <Mavrik> gotta open the issue with ffmpeg people then [12:09] <qxt> cbsrobot if you are not sure about your how many cores you have or ht just do this [12:09] <qxt> cbsrobot grep -c ^processor /proc/cpuinfo 2>/dev/null [12:10] <theholyduck> Mavrik, too bad theres no way of using neroaac with ffmpeg [12:10] <cbsrobot> qxt i'm on osx - maybe it's a osx bug [12:10] <theholyduck> libfaac is pretty bad in comparison [12:10] <lucas^> I use it in my program [12:10] <theholyduck> Mavrik, i'm glad i'm mostly doing bd rips and stuff [12:10] <Mavrik> theholyduck, yeah, that's annoying as hell [12:10] <theholyduck> gives me time to encode video and audio seperatly. [12:10] <qxt> cbsrobot hmm no proc in OSX. All the same BSD (don't kill me =P ) is a nice OS [12:11] <Mavrik> theholyduck, we're using libaacplus for the low-quality outputs [12:11] <Mavrik> makes alot of difference [12:11] <lucas^> ffmpeg -y -i foo.mpg -vn -acodec pcm_s16le -f wav foo.wav [12:11] <lucas^> neroAacEnc -q 0.3 -if foo.wav -of foo.aac [12:11] <Mavrik> but sadly it's limited up to 64k of bitrate [12:11] <theholyduck> Mavrik, atleast you guys arent dealing with avisynth [12:11] <theholyduck> as much as i love, it, its a pain aswell [12:11] <Mavrik> well... don't say too soon [12:11] <theholyduck> single threaded SLOW filtering pass before encoding [12:11] <theholyduck> slows everything down to a crawl [12:12] <Mavrik> right now I have to get a solution that can grab images from Intensity Shuttles, transcode it and pass two streams up to Wowza [12:12] <Mavrik> preferrably on laptop hardware :P [12:12] <Mavrik> lucas^, that's nowhere near practical [12:12] <Mavrik> lucas^, for automated transcoding [12:12] <lucas^> well, uh, I just transcoded like 100+ TV episodes that way [12:12] <theholyduck> Mavrik, well its doable with a fair bit of bash scripting, but not reliable no [12:13] <lucas^> you could also pipe it, but I see no reason [12:14] <theholyduck> lucas^, i've seen an approach like that bork if theres enough variation between your inputs [12:14] <lucas^> variation? what do you mean [12:14] <theholyduck> ofcourse, it shouldnt, because ffmpeg shouldnt drop frames. etc.etc [12:15] <theholyduck> lucas^, in some cases with some inputs, if you split the audio off that way, it wont fit correctly with the video again [12:15] <theholyduck> because of delays or what not specified in the container [12:15] <theholyduck> that wasnt preserved [12:15] <theholyduck> it wont happen on sane normal content, but who knows what Mavrik 's inputs are [12:15] <lucas^> the raw MythTV / WTV recordings I have do all sorts of crazy things [12:16] <lucas^> constant framerate changes, corrupted frames, etc. [12:16] <theholyduck> lucas^, they usually dont have special metadata and what not. [12:16] <lucas^> does VBI closed-caption data count? [12:16] <theholyduck> lucas^, :) [12:16] <theholyduck> 1 thing i know that occationally does flummux this approach [12:17] <theholyduck> is if you take a video file, uses mkvmerge to split it or merge it [12:17] <theholyduck> mkvmerge is a great tool for that, lets you losslessly split and merge almost anything [12:17] <theholyduck> through mkv trickery [12:17] <lucas^> I'm using ffmpeg to split / join to cut commercials, and Project-X to demux [12:17] <theholyduck> but sometimes if you submit a video cut like that into a process like yours [12:17] <theholyduck> you can end up with unsynced joins [12:18] <theholyduck> lucas^, mkvmerge is faster, and 100% lossless and frame accurate [12:18] <theholyduck> its impressive really [12:18] <theholyduck> it even cuts audio and subtitles [12:18] <theholyduck> correctly [12:18] <lucas^> mkvmerge is for Matroska files [12:18] <lucas^> my raw recordings are MPEG-2 [12:18] <theholyduck> lucas^, it takes any files as input [12:18] <theholyduck> outputs .mkv yeah [12:18] <lucas^> I spent god knows how long trying to get accurate cuts [12:18] <theholyduck> lucas^, take a look at mkvmerge then? [12:18] <theholyduck> it accepts mpeg2 input [12:18] <theholyduck> i think it cuts it aswell [12:19] <theholyduck> doesnt cut on frame but on a timestamp [12:19] <lucas^> one problem is that if I use mkvmerge to cut, it's going to strip the VBI data [12:19] <theholyduck> hh:mm:ss.nnnnnnnnnn [12:19] <lucas^> ffmpeg -c copy -map 0:v -map 0:a doesn't [12:19] <theholyduck> lucas^, possibly [12:20] <lucas^> then I'd need to run ccextractor on each separate segment within the file [12:20] <lucas^> and I know from experience that just doesn't work [12:20] <lucas^> I'm already using mkvmerge to mux Matroska files, I'll have to check into it, maybe I could get a bit more accuracy out of it [12:21] <lucas^> but for right now I have it about as close as I'm going to get it [12:21] <lucas^> any idea if there are any MKV players that actually display metadata correctly? [12:21] <theholyduck> doesnt mplayer and mpc-hc? [12:21] <theholyduck> atleast chapter and stuff [12:21] <theholyduck> works [12:21] <lucas^> mpc-hc doesn't show anything [12:21] <lucas^> doesn't have chapter support either, subs work fine [12:21] <theholyduck> what sort of metadata? [12:22] <lucas^> vlc supports subs and chapters, but shows all of three ID3-style metadata tags [12:22] <theholyduck> i got mpc-hc showing chapters, subtitle track names and audio track names [12:22] <lucas^> dumps the rest into an 'other info' tab, multiple ACTOR tags fail [12:22] <lucas^> mplayer doesn't really show much of anything [12:22] <lucas^> actually I think smplayer doesn't even play the damn file [12:22] <theholyduck> lucas^, really? [12:23] <lucas^> meanwhile, I mux to MPEG-4 and tag with AtomicParsley [12:23] <theholyduck> it should, unless your mplayer is old and terrible. [12:23] <lucas^> everything works beautifully on iTunes [12:23] <theholyduck> lucas^, well, i dunno how many of them support idv3 stuff though [12:23] <lucas^> haven't figured out the tag / subtitle format for WMP, VLC and MPC-HC work too [12:23] <lucas^> I'm using the official tags listed on matroska's website [12:24] <theholyduck> lucas^, most of them suport the info created/written by mkvmerge [12:24] <lucas^> some of them like 'artist', 'record', etc. are more popular than others in terms of actually getting implemented in players [12:24] <lucas^> basically the equivalent tags in ID3v1 for music [12:25] <Mavrik> theholyduck, yeah, my inputs are "ANY horror a crappy half-pro camera can create and is then raped by an amteur with Cinelerra/Premiere/etc" [12:25] <theholyduck> Mavrik, figured [12:25] <Mavrik> which is slightly better than my previous description "ANY horror the broadcasters are capable of stuffing into MPEG2-TS" :( [12:25] <Mavrik> I still wonder why I'm not an alcoholic after working in this industry [12:26] <theholyduck> Mavrik, i would never consider doing videoo professionally [12:26] <theholyduck> doing it as a hobby is fun [12:26] <theholyduck> dealing with it in real life seems like suffering [12:27] <theholyduck> sure, i have to deal with industry issues [12:28] <theholyduck> especially the foundness some studios have for authirng their nice 1080p content as 1080i60 telecine [12:28] <theholyduck> well not necesarily 1080p originally [12:28] <theholyduck> but some sort of progressive content [12:28] <theholyduck> so i have to ivtc the crap [12:32] <lucas^> what sort of video filters should I use on ATSC content? [12:32] <lucas^> I noticed that enabling a deinterlace filter in a video player helped quite a bit [12:33] <theholyduck> lucas^, depends on the content [12:34] <theholyduck> if its interlaced as in actually interlaced, then a deinterlacing filter is all you can do really [12:34] <theholyduck> the slower the better [12:34] <theholyduck> if its interlaced as in telecined [12:34] <theholyduck> then you can ivtc it [12:34] <theholyduck> if its some horrible mixture. you really really need avisynth [12:34] <JEEB> you'd need something else but ffmpeg in any case if it's telecined [12:35] <JEEB> since ffmpeg lacks an inverse telecine filter [12:35] <lucas^> when watching lip movements in cartoons, I notice several horizontal lines of alternating colors [12:35] <theholyduck> lucas^, well in cartoons, its usually a ivtc that sneeded [12:36] <lucas^> I also noticed a cropdetect filter, that might be useful [12:37] <theholyduck> not that i rember much of atsc and the net on this train is too bad to look up [12:37] <theholyduck> but i'm asuming like with other telecined content, there is a pattern you can see [12:38] <theholyduck> JEEB, you're better at this than me [12:39] <theholyduck> seems like it should be the same as ntsc in that respect [12:39] <theholyduck> open the video in mplayer. find a place with continous motion [12:39] <theholyduck> and hit the . key [12:39] <theholyduck> to step through the frames [12:40] <theholyduck> if its telecined, you should see 3 good frames and 2 interlaced frames. [12:40] <theholyduck> if its actually interlaced. all frames should be interlaced [12:40] <theholyduck> if some horrible broken mixture [12:40] <theholyduck> then who knows [12:41] <lucas^> I'll have to try different filters and read up on it I guess [12:42] <lucas^> thanks for your help man [12:42] Action: JEEB usually makes people check field-by-field, but not sure if that can be easily done in anything else but Avisynth [12:42] <JEEB> separatefields().selecteven() -> make a person check if he gets 1-2-3-4-4 or 1-2-3-4-5, then have him remove selecteven() and check if he gets the same picture twice (or two separate pictures) [12:43] <theholyduck> JEEB, mplayer framestep works just as fine [12:43] <JEEB> yes, but it doesn't let you see field-by-field [12:43] <JEEB> only two fields stuck together :3 [12:43] <theholyduck> the pattern is still there [12:44] <lucas^> wouldn't suppose there's a way to figure out the versions of various libraries in static ffmpeg builds? [12:44] <lucas^> e.g., libx264, libvp8, libfaac, libvorbis [12:48] <Mavrik> um, what exactly is telecine? [12:51] <theholyduck> a way of converting progressive content into interlaced content [12:51] <Mavrik> oh I see [12:51] <theholyduck> normally its done on movies or cartoons/anime [12:51] <theholyduck> to turn say 24p progressive [12:51] <theholyduck> into 60i [12:52] <theholyduck> using a 3:2 pattern [12:52] <Mavrik> I see, just reading the article. Luckly never had to deal with that particular problem -_- [12:54] <theholyduck> Mavrik, theres a good article if you're intrested [12:54] <Mavrik> always :) [12:54] Action: theholyduck goes to find [12:54] <theholyduck> http://mod16.org/hurfdurf/?p=12 Mavrik [12:57] <Mavrik> ooo, thanks [13:04] <theholyduck> Mavrik, pretty good eh? [13:04] <Mavrik> yeah, it's pretty neat [13:05] <theholyduck> fluff writes some pretty decent stuff on video at times [13:05] <Mavrik> now I just have to find some resources that deal specifically with PAL since I mostly handle that content :) [13:05] <theholyduck> Mavrik, pal ivtc is easy enough [13:05] <theholyduck> usually, they just interlace it plainly or speed it up [13:06] <theholyduck> *and speed it up [13:06] <theholyduck> or they introduce a duplicate field [13:06] <theholyduck> every 12 frames? [13:06] <Mavrik> yeah, that's probably why I've never had to deal with it [13:07] <theholyduck> http://www.mplayerhq.hu/DOCS/HTML/en/menc-feat-dvd-mpeg4.html Mavrik [13:07] <Mavrik> usually it's just sped up from [13:07] <JEEB> 99%+ of all PAL stuff is sped up, yeah [13:07] <theholyduck> some info on pal pulldowns [13:07] <JEEB> then you have the few things that are telecined [13:07] <Mavrik> most cameras I have lying around here record @ 25 fps (or 50i) anyway [13:07] <JEEB> yeah [13:07] <theholyduck> JEEB, i have SEEN one file with the PAL 2:2:2:2:2:2:2:2:2:2:2:3 pulldown [13:08] <theholyduck> but i know its rare [13:09] <Mavrik> hmm, here one source claims some people just push in 2 dups after every 48 frames [13:09] <JEEB> also, PAL is a mixed bag in the case you get 30i content with actually either 30 or 60 pictures in a second there's really no pretty way of putting that into PAL [13:09] <theholyduck> Mavrik, fields [13:09] <theholyduck> probably [13:09] <Mavrik> sorry, fields yeah [13:09] <theholyduck> Mavrik, thats basicly what i mean. [13:10] <theholyduck> "Every 12th frame is shown for the duration of three fields, instead of just two. This avoids the 4% speedup issue, but makes the process much more difficult to reverse. It is usually seen in musical productions where adjusting the speed by 4% would seriously damage the musical score." [13:10] <theholyduck> i.e every 24th field [13:10] <theholyduck> more or less [13:14] <theholyduck> Mavrik, the wikipedia article is pretty good aswell [13:14] <theholyduck> on telecine that is [15:43] <pif> hi, can I losslessy cut a mp4 file with ffmpeg? [15:52] <sacarasc> Probably, but I'm not sure how accurate it would be. [15:53] <sacarasc> Unless ffmpeg has non-keyframe seeking now. [16:29] <adv_> i have a video where a person is talking outdoors and some birds are chirping and there's some outdoor noise. is there a way to postprocess the video and make the human voice louder/more apparent ? or reduce somehow the external noise? [16:34] <Kartagis> your best bet is play with it with equalizer [16:35] <adv_> Kartagis, can you please elaborate? [16:36] <cbreak> adv_: I've just seen someone in an other channel ask the same question [16:36] <adv_> cbreak, it's me [16:37] <Kartagis> adv_: open it with a video editor, and use equalizer to achieve what you want [16:37] <cbreak> an equalizer is a combination of frequency filters [16:38] <adv_> Kartagis, can you suggest one? [16:39] <Kartagis> avidemux maybe [16:39] <Kartagis> what cbreak said [16:40] <adv_> avidemux and there is an equilizer "filter" in it? [17:32] <pwlnw> hi! does -f rawvideo works ? [17:32] <pwlnw> i try 2-pass h.264 and get Unable to find a suitable output format for 'rawvideo' [17:36] <pwlnw> ok -f work.but -profile not. [17:39] <pwlnw> what's wrong here http://pastebin.com/J8jqMmMz ? [17:40] <pwlnw> profile still Main [17:41] <pwlnw> this options work with libav, but libav broke another option - passlogfile [17:41] <pwlnw> i need both or them [17:48] <cbsrobot> pwlnw: try: -vprofile high -preset slow [17:50] <cbsrobot> or the whole comand: [17:50] <cbsrobot> ffmpeg -i defy.flv -v verbose -pass 1 -passlogfile 6412 -an -b:v 256k -vcodec libx264 -vprofile high -preset slow -level 30 -f rawvideo -y /dev/null [17:51] <pwlnw> cbsrobot there are now more presets file in git! [17:51] <pwlnw> wtf? [17:52] <pwlnw> i do make install and have two libx264-ipod3* [17:52] <cbsrobot> pwlnw: ffmpeg passes presets directly to x264: http://mewiki.project357.com/wiki/X264_Settings#preset [17:53] <pwlnw> cbsrobot your variant still use Main profile [17:53] <cbsrobot> skip the -level setting [17:54] <pwlnw> cbsrobot still haven't. will build 0.10 ffmpeg so called "stable" [17:56] <pwlnw> same shit with 0.10 [17:56] <pwlnw> where is preset files of x264 ? [17:58] <cbsrobot> pwlnw: my guess would be that for pass 1 it does not matter [17:58] <burek> pwlnw, what actually doesn't work for you? [17:58] <cbsrobot> check pass 2 and see if it works [17:58] <burek> pastebin is ok [17:59] <pwlnw> oow. profile is not work on first pass [17:59] <cbsrobot> pwlnw: but it's just a guess - not 100% sure [17:59] <pwlnw> but I don't need anymore -pre:v ? where is 'slow.preset' files? [18:00] <burek> there are no preset files anymore [18:00] <burek> -preset -profile -tune are being directly passed to libx264 [18:00] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [18:00] <JEEB> ^ list of libx264's internal presets [18:19] <stf> does this "[matroska @ 0x202cc00] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 3782 >= 3723" looks anyone of you familiar and solvable? [19:31] <level09> cant get my ffserver to work [19:32] <level09> I'm trying to stream a small mp4 file from my server [19:33] <level09> Here is my config file http://paste.ubuntu.com/858151/ [20:41] <brontosaurusrex_> how would i scale/pad to certain resolution (in pixels) with par=1 [20:41] <brontosaurusrex_> ? [20:42] <brontosaurusrex_> and the same for scale/crop? [20:43] <brontosaurusrex_> (the input par must be identified/respected) [20:45] <Mavrik> hmm [20:45] <Mavrik> brontosaurusrex_, passing "-aspect" with full resolution to scale filter always made PAR 1 video for me [20:47] <brontosaurusrex_> Mavrik: like -aspect 480:320 ? [20:47] <Mavrik> like -aspect 1.666 or whatever your video aspect is :P [20:48] <brontosaurusrex_> ok, then how to auto pad or auto crop? [20:51] <relaxed> what is the source's par? [20:53] <Mavrik> brontosaurusrex_, hmm, good question [20:53] <brontosaurusrex_> relaxed: anything, i am scripting this [20:54] <brontosaurusrex_> for starters i will try scale 480:-1 and see how many errors i get ... [21:07] <brontosaurusrex_> lol, well, seems like 480:-1 is all thats needed actually, the phone will play files that are actually a bit larger than its screen size as well. [21:08] <brontosaurusrex_> also the aspect seems to be preserved with -1 [21:10] <Mavrik> yes, but also PAR [21:12] <brontosaurusrex_> nope, par is now 1 as it seems [21:14] <brontosaurusrex_> hrm [21:14] <brontosaurusrex_> actually not sure [21:15] <brontosaurusrex_> you are right, par is also the same as input [21:15] <MindSpark> hi, anyone here with an android phone and willing to help with transcoding rtmp to rtsp using ffserver? [21:15] <MindSpark> been trying to get this done for over a week now without success [21:16] <MindSpark> this rtmp://aljazeeraflashlivefs.fplive.net/aljazeeraflashlive-live/aljazeera_ar… needs to be converted to something viewable by android, any hints? [21:17] <brontosaurusrex_> Mavrik: so, any clues how to make output par to 1? [21:17] <Mavrik> I did it by manually setting aspect ratio of output video [21:18] <brontosaurusrex_> Mavrik: ok [21:19] <brontosaurusrex_> MindSpark: uhmm, android phones are quite different i believe, different screen size, and so on, Iam just trying to make a workable preset for wildfire S right now [21:39] <brontosaurusrex_> Mavrik: how about -vf scale=iw*sar:ih, scale=480:-1 ? [21:39] <brontosaurusrex_> or -vf scale=iw*sar:ih,scale=480:-1,setsar=1:1 even? [21:41] <MindSpark> brontosaurusrex_, is the videosize crucial for the video to display? [21:41] <MindSpark> I am getting no video whatsoever [21:41] <MindSpark> only audio [21:43] <brontosaurusrex_> MindSpark: what is your command line right now? [21:44] <brontosaurusrex_> would consequential resizes somehow merge, or are this separated operations? [21:45] <MindSpark> brontosaurusrex_, I am using ffserver configuration file [21:46] <MindSpark> brontosaurusrex_, http://pastebin.com/pWDwjBZt [21:48] <brontosaurusrex_> MindSpark: uf, aim just using a preset and profile: -preset veryfast -profile baseline [21:48] <brontosaurusrex_> where preset very fast denotes -refs=1 [21:48] <MindSpark> brontosaurusrex_, I recompiled ffmpeg on my ubuntu machine and I don't have any of those presets/profiles. Any idea where I can get them? [21:48] <brontosaurusrex_> and since Iam the one that requested internal x264 presets, i'd rather stick with them ;) [21:49] <brontosaurusrex_> MindSpark: they are built in [21:49] <MindSpark> brontosaurusrex_, so I need nothing in the /.ffmpeg dir? [21:50] <brontosaurusrex_> nope [21:50] <MindSpark> can't find "profile" in the ffserver config though [21:51] <brontosaurusrex_> Mavrik: -vf scale=iw*sar:ih,scale=480:-1,setsar=1:1 <- this works btw, just not sure how dumb it is [21:51] <MindSpark> ffserver.mpeg.conf:19: AVPreset error: veryfast [21:52] <MindSpark> hmm, did you mean MindSpark ? [21:52] <brontosaurusrex_> MindSpark: i'd try to put that into videocoder options [21:52] <Mavrik> brontosaurusrex_, that's double resizing though [21:53] <MindSpark> brontosaurusrex_, what do I do? [21:53] <brontosaurusrex_> Mavrik: yes [21:53] <MindSpark> I am getting this error anyway [21:53] <brontosaurusrex_> Mavrik: but in 99% of the cases my video is allready in par=1, so &. [22:03] <MindSpark> they don't seem to be built in [22:03] <MindSpark> I keep getting errors [23:51] <brontosaurusrex_> anything smarter than : -vf scale=iw*sar:ih,scale=480:-1,crop=iw:"floor((ih/2)*2)",setsar=1:1 ? [23:55] <burek> brontosaurusrex_, it depends [23:56] <brontosaurusrex_> burek: basically i7d like scale1 and 2 in one step [23:56] <brontosaurusrex_> *i'd [23:59] <brontosaurusrex_> or a question: would 2nd scaling step be absolutely ignored if there is no scaling to be done? [23:59] <brontosaurusrex_> i can't see any speed penalty, but not sure how to test properly [00:00] --- Mon Feb 27 2012

1 0

ffmpeg-devel.log.20120225
by burek 26 Feb '12

26 Feb '12

[00:05] <pasteeater> why is it so hard to get "This program is not developed anymore" reworded in ubuntu's libav's "ffmpeg"? [00:05] <pasteeater> i need more bikes. [00:07] <iive> pasteeater: hum? [00:08] <ubitux> iive: i guess he's refering to libav packaging an old ffmpeg binary [00:08] <pasteeater> https://bugs.launchpad.net/ubuntu/+source/libav/+bug/939863 [00:11] <ubitux> seriously& :( [00:12] <nevcairiel> One could claim malicious intend and assume that the assumptions users make when they read that is wanted [00:13] <iive> nevcairiel: given that libav people have a history of such dubious claims, I would not mark such assumption as unlikely. [00:15] <iive> also siretart's argument to reject the patch is example of logical fallacy. [00:19] <ubitux> i admit that's not really fair competition [00:19] <ubitux> i guess all shot are allowed against the evil ;) [00:20] <durandal_1707> what patch? [00:20] <iive> durandal_1707: in the bug thread. [00:21] <Compn> pasteeater : well, siretart is in the channel, you could talk to him here [00:21] <Compn> he is ubuntu packager [00:21] <Compn> i think [00:21] <iive> debian. [00:21] <Compn> his freenode host says ubuntu :P [00:21] <iive> ok, both. [00:22] <nevcairiel> isnt ubuntu just debian with a bit of marketing on top? :p [00:22] <iive> yes and some corporation to back it up. [00:24] <durandal_1707> time when such bug is reported is interesting [00:25] <Compn> pasteeater : you should just argue that they should make ffmpeg an alias to avconv and get it over with [00:25] <Compn> so we stop getting dumb libav bugreports [00:26] <nevcairiel> i think they removed it from git [00:26] <pasteeater> i think you're right [00:26] <ubitux> when updating libav the issue won't be valable anymore [00:26] <Compn> i didnt see an ffmpeg.c file, no [00:27] <ubitux> yes, they droped it [00:27] <ubitux> as well as ffmpeg removed avconv [00:27] <nevcairiel> of course updating will take at least one new ubuntu release, maybe even 2 [00:27] <ubitux> yes sure& [00:27] <durandal_1707> that is one year? [00:29] <Compn> its like if iceweasel popped up a message saying 'firefox is not developed anymore, this is provided blah blah api blah' [00:29] <Compn> ;) [00:33] <ubitux> except that "iceweasel" is not a confusing name ;) [00:33] <Compn> ehe [00:36] <iive> Yep, and then refuse to change the message, because the new version of firefox have different options :) [00:36] <iive> thus it is no longer "this program". [00:40] <iive> Compn: I assume that iceweasel would/could pop that message when you try to run 'firefox' :) [00:44] <ubitux> they could just ship ffmpeg, with avconv symlink to ffmpeg if they care so much about their tools [00:45] <ubitux> and it's also interesting for them because it will make the users think libav has a lot of features ;) [00:45] <ubitux> anyway, i think i'll stop trolling, my flame barometer is in the red [00:51] <Compn> one more troll [00:51] <Compn> ffmpeg includes almost all avconv changes anyways [00:52] <Compn> well that is fact, but ehhhhhh :P [00:52] <buzz_> i added my agreement to the ubuntu bug. see how much reinhard will continue to resist. im sure he will, and then it will be even more obvious that it is for the wrong reasons [00:53] <Compn> buzz_ : why not talk to reinhard here? --------> siretart [00:53] <Compn> :P [00:53] <Compn> siretart : darn those pesky users, always wanting to use ffmpeg and not switch to avconv! [00:53] <buzz_> im happy to if he replies here [00:54] <buzz_> i stopped using the debian packages once they were slyly switched to libav [01:09] <CIA-17> ffmpeg: 03Derek Buitenhuis 07master * rb93c91579d 10ffmpeg/tests/ (30 files in 3 dirs): [01:09] <CIA-17> ffmpeg: fate: Overhaul WavPack coverage [01:09] <CIA-17> ffmpeg: WavPack has a comprehensive test suite, and a bunch [01:09] <CIA-17> ffmpeg: of corner cases. [01:09] <CIA-17> ffmpeg: Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [01:09] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <durandal_1707> now show begins... [03:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r1e44dda825 10ffmpeg/libavformat/flvdec.c: [03:03] <CIA-17> ffmpeg: flvdec: fix "libavformat/flvdec.c:195:29: warning: variable size1 set but not used" [03:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r4dbdd3ec6f 10ffmpeg/libavformat/mp3enc.c: [03:03] <CIA-17> ffmpeg: mp3enc: fix "warning: best_bitrate_idx may be used uninitialized in this function" [03:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r3d3b603f7f 10ffmpeg/libavcodec/a64multienc.c: [03:03] <CIA-17> ffmpeg: a64multienc: fix "warning: buf may be used uninitialized in this function" [03:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r7b9d8703f3 10ffmpeg/libavcodec/adpcm.c: [03:03] <CIA-17> ffmpeg: adpcmdec: fix "warning: array subscript is above array bounds" [03:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r418403662b 10ffmpeg/libavformat/rmdec.c: [03:03] <CIA-17> ffmpeg: rmdec: fix "warning: format %d expects argument of type int, but argument 7 has type int64_t" [04:26] <Compn> contact the largest users of ffmpeg, like youtube, facebook, megavideo, blip.tv and others and ask them for access to samples that do not decode correctly. [04:26] <Compn> >megavideo [04:26] Action: Compn sad again :( [04:28] <CIA-17> ffmpeg: 03Christophe GISQUET 07master * r34454c761f 10ffmpeg/libavcodec/ (5 files in 2 dirs): (log message trimmed) [04:28] <CIA-17> ffmpeg: SBR DSP x86: implement SSE sbr_sum_square_sse [04:28] <CIA-17> ffmpeg: The 32bits targets have been compiled with -mfpmath=sse for proper reference. [04:28] <CIA-17> ffmpeg: sbr_sum_square C /32bits: 82c (unrolled)/102c [04:28] <CIA-17> ffmpeg: C /64bits: 69c (unrolled)/82c [04:28] <CIA-17> ffmpeg: SSE/32bits: 42c [04:28] <CIA-17> ffmpeg: SSE/64bits: 31c [04:28] <CIA-17> ffmpeg: 03Alex Converse 07master * re32548d133 10ffmpeg/libavcodec/tiff.c: [04:28] <CIA-17> ffmpeg: tiff: Make the TIFF_LONG and TIFF_SHORT types unsigned. [04:28] <CIA-17> ffmpeg: TIFF v6.0 (unimplemented) adds signed equivalents. [04:28] <CIA-17> ffmpeg: 03Christophe GISQUET 07master * r2e74a5abc2 10ffmpeg/libavcodec/ (arm/sbrdsp_init_arm.c sbrdsp.c sbrdsp.h): [04:28] <CIA-17> ffmpeg: SBR DSP: use intptr_t for the ixh parameter. [04:28] <CIA-17> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [04:28] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r14c98973f5 10ffmpeg/libavformat/apetag.c: [04:28] <CIA-17> ffmpeg: apetag: do not leak memory if avio_read() fails [04:28] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [04:28] <CIA-17> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [04:28] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r6d11057006 10ffmpeg/libavformat/apetag.c: [04:28] <CIA-17> ffmpeg: apetag: propagate errors. [04:28] <CIA-17> ffmpeg: Fixes crashes if reading the tag value fails. [04:28] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [04:29] <michaelni> Compn, megavideo ? [04:29] <michaelni> you mean FBI ;) [04:31] <Compn> FBI needs to get some more videos up [04:31] <Compn> those lazy people [06:07] <overflow_0f8b> yeah FBI was always lazy sharing porn movies and such ;/ [06:08] <mystica555_> lol [06:12] Action: michaelni imagines anonymous putting porn of FBI agents on fbi.gov [06:13] <overflow_0f8b> well i know of a site that is running in cooperation with FBI and contains "suspicious" content [06:14] <overflow_0f8b> looks like some things can be an exception if they use it as a jailbait [06:14] <ohsix> they get license to do it for that :] [06:21] <overflow_0f8b> well if they can headshot terrorists, blow up helicopters, take any private property at will [06:22] <overflow_0f8b> everything they do is legal, since they are the law [09:33] <pwlnw> hi! what about this issue ? http://pastebin.com/mMCpLWcS [09:34] <pwlnw> i need disable generation of nop instructions in x264, but can't [09:36] <pwlnw> oops. i have linked ffmpeg to wrong library /usr/lib/i386-linux-gnu/libx264.so.120 [10:45] <siretart> Compn: ? [12:14] <pierow> Hello, I was wondering if anyone could give me some help with AVCodecContext values for video and audio encoding? Is there a reference where I could find good default values for encoding different audio and video codecs? [12:14] <pierow> Even just a point to somewhere in ffmpegs code would be very helpful :) [13:23] <Compn> siretart : nothing, people were just submitting bugs to ubuntu and i was telling them that they could contact you on irc [13:42] Action: Compn should stop talking about ffmtech because its upsetting [13:47] <Compn> ok one more mail... [13:49] <wbs> Compn: regarding the amr fix, I've explicitly said I don't want to be paid by ffmtech for that issue (in that particular mail you quoted) [13:49] <Compn> wbs : i know :D [13:49] <Compn> just locating mails for my proposals is hard , and that one was easy to find [13:50] <Compn> i didnt repropose it, just mentioned it for historical purposes [13:50] <Compn> in that, it was an ffmpeg proposal that didnt get accepted [13:52] <siretart> Compn: sorry, I fail to see the problem with people filing bugs in launchpad for bugs in ubuntu. As long as the bugreport is kept at a civil, technical level, that is. [13:53] <Compn> you're absolutely correct [22:08] <NovaSt0rM> hello [22:08] <NovaSt0rM> i've googled the hell out of this but i still can't find an answer: when using a custom streaming IO buffer, and returning EOF when there's no more data, av_read_frame() returns a partial frame, but not an error. I can clearly see that EOF is reached, but more data will be forthcoming. Is there any way to force av_read_frame to re-try the entire read once more data is available? [22:08] <NovaSt0rM> Currently, it's returning another broken frame (the 2nd half of the first frame) on the next call. [22:11] <NovaSt0rM> perhaps there's a way to append the frames together to be fed into avcodec_decode_audio3? [22:11] <NovaSt0rM> or just some way to force av_read_frame to restart after EOF [22:29] <rainmaker1> Hi, is this a right place to ask to request new feature/debug for ffmpeg? I mean paid one. [22:40] <iive> rainmaker1: here or/and the maillist. [22:42] <rainmaker1> iive: Well then, can I speak with you on private please? [22:44] <iive> no. I'm not active in ffmpeg atm, so i won't be of any help to you. [22:45] <rainmaker1> iive: ok, then I'll wait for someone to contact me, thank you. [22:46] <iive> rainmaker1: it would help if you give some hint about the task, as people usually have areas of expertise. [22:46] <iive> aka is it about (de)muxing files, codec, custom program or streaming? [22:46] <beastd> rainmaker1: also sending an offer on the ffmpeg-devel ML should get you attention from more developers [22:47] <iive> yes, all active developers follow the mail list. [22:49] <rainmaker1> well, I don't know how to make an offer, as I don't know how much something can cost. For example, this is for fixing a bug (I belive) [22:49] <rainmaker1> but, ok, I will try to figure out what to say at ML [22:51] <beastd> rainmaker1: you don't have to put exact values in the public mail. just say you have this or that problem with ffmpeg and would pay for getting it fixed/or getting the feature implemented. [22:52] <rainmaker1> thats fine. thank you for your help to iive and you. [00:00] --- Sun Feb 26 2012

1 0

ffmpeg.log.20120225
by burek 26 Feb '12

26 Feb '12

[00:02] <vadim_> rainmaker1: 32 bits... this system was already installed on a test machine and i decided not to reinstall [00:04] <rainmaker1> vadim_: still running, I will leave it like this in screen will see if it will crash until tommorow [00:04] <vadim_> ok... let me know somehow )) [00:05] <rainmaker1> vadim_: I'll be here tommorow [00:05] <vadim_> rainmaker1: ok [00:17] <Christophe_v_E> I've managed to activate ffmpeg - how do I run a command to it through php? [00:21] <teratorn> Christophe_v_E: the same way you run any command in php, I guess [00:24] <Christophe_v_E> so if I were to call an execute in php I could just for example put this in there and it would work? ffmpeg â¬i video.flv -acodec libfaac -ss 00:00:00 -t 00:03:00 -ar -ab -f flv -s videopart.flv [00:30] <teratorn> Christophe_v_E: I don't use php, so I wouldn't know [00:32] <Christophe_v_E> And if I wouldn't want to do it in php? Say I have a folder with flv's - with what could I split all the flv's in the folder into random pieces and put 20 random pieces together? Any help is appreciated :) [00:32] <nextime> Christophe_v_E : probably you will have issue in managing the execution time [00:32] <Christophe_v_E> In php I've set the execution time to unlimited .. I don't know if that'll account for everything, but I suppose so - when using xampp. [00:33] <nextime> ouch [00:33] <nextime> what a bad thing to do :) [00:33] <Christophe_v_E> it's to run locally for a school program [00:34] <nextime> Christophe_v_E : i have to admit that i'm a sort of anti-php taleban, so, do what you are doing and don't take care of my opinions on that [00:34] <nextime> :) [00:34] <Christophe_v_E> I'm willing to pay somebody to make something that will split the flv's up to random pieces. Then - with a seperate trigger put them together again as one video and save it, every new generation would have a different name of course. [00:35] <Christophe_v_E> Depending on the price of course - if it's going to cost me too much for what you would make in half an hour I'd rather learn to do it myself for an entire day :) [00:56] <rickk> hi [00:56] <rickk> I have compiled avcodec with libx264 library. but the fuction avcodec_find_encoder cannot find CODEC_ID_H264. [00:56] <rickk> ./ffmpeg -codecs|grep x264 [00:56] <rickk> EV libx264 libx264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 [00:56] <rickk> EV libx264rgb libx264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 RGB [01:07] <Christophe_v_E> Can somebody please explain me how I run a command. Thank you. [01:12] <Christophe_v_E> Is there any guide at all on how to use this? It seems like the documentation dives right into commands while I don't even know what to run them in to get something done. I'd really like advise. [01:15] <pasteeater> Christophe_v_E: your example will re-encode. you can also "copy and paste" with relative accuracy. [01:15] <pasteeater> ffmpeg -i input.flv -ss 00:01:29.00 -t 15 -c copy output.flv [01:15] <Christophe_v_E> I have no idea what I have to do with my example - where do I put it to get something done, pasteeater ? [01:16] <Mavrik> why are you asking us PHP questions? [01:16] <Christophe_v_E> doesn't have to be php at all - [01:16] <Mavrik> for the sake of all your users and server security, go read the damn PHP documentation so you'll understand what you're doing [01:16] <Christophe_v_E> .. that is .. rather friendly. I'm going to run it locally - for a small school thing. [01:18] <pasteeater> are all flv in your directory going to have the same ss and t? [01:18] <Christophe_v_E> they're all variable [01:18] <Christophe_v_E> the flv's are different lengths, as should be the splitted parts, as should be the outputs [01:19] <Christophe_v_E> But I'd really like to know where I would put a command .. literally. I haven't come accross anything telling me where I should go with my command .. - thanks for your interest :) [01:30] <undercash> is it possible to stream a playlist with ffmpeg? [01:50] <pasteeater> undercash: if it's a simple text file you can do it with a bash script. [01:51] <pasteeater> maybe it would be a good SOC project. [01:52] <undercash> using this cmd [01:52] <undercash> ffmpeg -re -i "doc.avi" -vcodec libx264 -preset fast -crf 26 -acodec libfaac -ab 128 -ar 44100 -f flv rtmp://live.justin.tv/app/xxx [01:52] <undercash> shouldn't this be a xml file like vlc [01:53] <pasteeater> you probably want 128k instead of 128 (or omit -ab as 128k is default, IIRC). [01:53] <undercash> oh yea [01:53] <undercash> but how is supposed to be the text file? any exemple? [01:56] <pasteeater> i meant use can have a text file of a list of your inputs and then "for file in ${inputfile} do", etc. [01:56] <undercash> i d like just to use the terminal to stream.. [01:56] <pasteeater> maybe i'm not understanding what you're asking, but now i have to go back to work. [01:57] <undercash> ok passteeater thx [01:58] <qxt> From shell how can I get the length of a video. [02:00] <qxt> Writing a script to do 2 pass encoding with a predetermined output size. [02:03] <daimonion> Hi! [02:03] <daimonion> Is anybody here? [02:04] <qxt> Sort of . We are just acting like we are cool and ignoring ppl. SUP [02:04] <sacarasc> qxt: Use the programme mediainfo. It's awesome at this kind of thing. [02:05] <daimonion> Well, I have a problem compiling ffmpeg. [02:05] <qxt> sacarasc I just remembered ffprobe [02:05] <sacarasc> I prefer mediainfo for things like this. [02:05] <daimonion> I think it is due to changes in vo_aacenc [02:05] <qxt> daimonion what OS [02:06] <daimonion> Arch Linux. [02:06] <daimonion> qxt: Any other information needed? [02:07] <sacarasc> daimonion: The output from make would be nice. :p [02:08] <daimonion> http://pastie.org/3451503 [02:08] <qxt> daimonion I got a few scripts I wrote that will install upgrade remove ect for debian systems. Not 100% sure it will work with Arch (a great distro btw) [02:08] <pasteeater> daimonion: what's wrong with ffmpeg-git from AUR? [02:09] <daimonion> I don't know, I do not use it. :) [02:09] <pasteeater> it works. adding libvo_aacenc support is easy. [02:09] <daimonion> I use my own PKGBUILD, which compiles from here: http://www.ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 [02:09] <pasteeater> vo-aacenc PKGBUILD is also in AUR [02:10] <daimonion> I know, I have the latest vo-aacenc and vo-amrwbenc installed. [02:10] <pasteeater> all you have to do is add vo-aacenc to depends and '--enable-libvo-aacenc --enable-version3' to the ffmpeg configure if you use the AUR PKGBUILDs [02:10] <daimonion> Which btw broke ffmpeg: ffmpeg: error while loading shared libraries: libvo-amrwbenc.so.0: cannot open shared object file: No such file or directory [02:11] <daimonion> That's why I want to recompile, but I get error. [02:11] <pasteeater> vo-* probably is built as shared by default [02:11] <pasteeater> (the ones from AUR do not) [02:12] <qxt> daimonion Ill put a automagic script up for ArchLinux tomorrow over at my pastebin http://pastebin.com/u/Onryo [02:12] <qxt> Just because I like Arch =) [02:12] <pasteeater> i don't understand what's wrong with using the simple AUR packages [02:13] <daimonion> Here are my build flags: http://pastie.org/3451538 [02:13] <daimonion> OK, thanks qxt. [02:14] <pasteeater> you don't need --enable-runtime-cpudetect unless you're a distro, cross-compiling, or going to be using the binary on other machines with different cpu [02:14] <pasteeater> --enable-avfilter is no longer an option, IIRC [02:14] <qxt> daimonion The Ubuntu guys baggered me a while back for this http://pastebin.com/uHZx6wJR but that is the last thing I am doing for Ubuntu. Use Debian myself [02:15] <pasteeater> onryo: ffmpeg -i IronMan.mkv 2>&1 | awk '/Duration/{print $2}' | sed 's/,//g' [02:15] <pasteeater> probably a million other ways to get the same. not sure if it will work for all inputs. [02:15] <onryo> pasteeater yup you got a darn good memory. I asked you about speeds =) [02:16] <onryo> pasteeater was the same clip. You had a 920 sandy bridge IIRC? [02:16] <onryo> Well it was fast [02:17] <daimonion> Sorry, my connection broke. [02:18] <pasteeater> onryo: i7 860. [02:18] <pasteeater> upgraded from an old P4 room heater. [02:18] <daimonion> pasteeater: As you can see, ffmpeg-git from AUR uses --enable-nonfree build flag, which I don't want. Also, it doesn't use vo_aacenc. [02:19] <pasteeater> pasteeater | all you have to do is add vo-aacenc to depends and '--enable-libvo-aacenc --enable-version3' [02:19] <rickk> can anybody help me with av_log ? [02:19] <pasteeater> nonfree is for faac. just remove faac from depends and nonfree libfaac from configure [02:19] <daimonion> pasteeater: This is whyt I did in my own PKGBUILD script. [02:20] <pasteeater> that's one of the reasons to have AUR. so you can modify the build scripts. [02:20] <rickk> void avcodec_register(AVCodec *codec) [02:20] <rickk> { [02:20] <rickk> AVCodec **p; [02:20] <rickk> avcodec_init(); [02:20] <rickk> p = &first_avcodec; [02:20] <rickk> while (*p != NULL) { [02:20] <rickk> av_log(*p,1,"Name : %s CodecID 0x%x",(*p)->name,(*p)->id); [02:20] <rickk> p = &(*p)->next; [02:20] <rickk> } [02:20] <pasteeater> rickk: you should use pastebin.com for a multi line paste [02:20] <rickk> sorry [02:21] <pasteeater> daimonion: you take a working AUR PKGBUILD and modify it to suit your needs. it's usually easier and faster than doing it from scratch. [02:21] <rickk> http://pastebin.com/EjTQqTNG [02:22] <rickk> i cannot display av_log [02:22] <rickk> i like to list all the codecs to be registered [02:23] <daimonion> I don't understand what part you don't get: I did exactly that, the problem isn't in my PKGBUILD script, the problem is in the ffmpeg itself. ffmpeg wont compile since vo_aacenc and vo_amrwbenc upgrade. [02:23] <pasteeater> i'll try [02:26] <pasteeater> Unknown option "--enabke-libvo-amrwbenc" [02:26] <pasteeater> durrrrr [02:30] <pasteeater> daimonion: worksforme [02:30] <pasteeater> don't build as shared [02:41] <onryo> OK my script seems be be working. Calculates the b:v and b:a if need for predetermined "movie" sizes. My question is if there is anything wrong with the output format from this script or if it could be cleaner. [02:41] <onryo> ffmpeg -i movieIn.avi -pass 1 -vcodec libx264 -preset veryslow -b:v 1226.72134943k -f matroska -sn -an /dev/null && ffmpeg -i movie.avi -pass 2 -vcodec libx264 -preset veryslow -b:v 1226.72134943k -acodec libfaac -ab 128k -ac 2 -sn -y movieOut.mkv [02:41] <onryo> Yeah the floating points are there in the video etc [02:42] <pasteeater> looks fine to me. check to see that the output is actually yuv420p which is probably what you want. if the input is dv it might pick yuv422 instead. [02:43] <cali> is there any work on the Adobe F4F crap decoding? [02:45] <rickk> can anyone help me... [02:46] <rickk> i like to list all the codecs to be registered [02:46] <rickk> http://pastebin.com/EjTQqTNG [02:46] <onryo> pasteeater good idea. Going to put in a few checks. But seems to look good. I guess Ill drop this into qt to make it easy for others to use too. calculating 2 pass predetermined sizes is a drag. [02:46] <rickk> I don't know y av_log doesn't show.. [02:46] <rickk> i set the log level as 0 though [02:52] <pasteeater> onryo: you could just add '-pix_fmt yuv420p' [02:53] <onryo> pasteeater Was thinking about that. Would have have to use that for both passes? [02:54] <pasteeater> yes [02:54] <onryo> thx [02:55] <pasteeater> if you use -b:v, then you can also use -b:a for consistency instead of -ab </nit>. [02:56] <onryo> -b:a 128k ... For some reason I think my computer can read my mind. Sloppy of me. [02:59] <onryo> Does ffmpeg round off the floating points in the bit rate? [02:59] <onryo> example my script calculated this for video -b:v 1226.72134943k [03:10] <onryo> cout << "The size of a float is:\t\t" << sizeof(float) << " bytes.\n"; [03:10] <onryo> sry [03:29] <onryo> pasteeater Is a planar YCbCr colorspace (pix fmt yuv420p) the best quality/compression for x264. Not going to say I am an expert in that. Wrote some C program a few years back for converting RGB 24bpp -> YUV420P 12bpp and got to admit I did not have a clue about what I was doing =) [03:31] <onryo> Anybody who might have some advice is welcome with feedback. [03:33] <pasteeater> i think a better determinant is what players you want to support. all can play yuv420. [03:34] <pasteeater> techically, x264 does support RGB, but i don't think it's encouraged and most players only support YUV. [03:34] <pasteeater> AFAIK [03:40] <onryo> pasteeater Well its not a bad way to roll that is for sure. Not full of BS like wine tasters and say "I get a hint of spring peaches" regarding colors I can't see anyway. The numbers add up. pix_fmt yuv420 planer it is. [03:44] <onryo> Judging by my spelling its about time for me to go to bed. 4am again...I'll never learn. gn [03:44] <pasteeater> full bouquet. earthy. diesel. leather. chalk. [11:12] <guy> simple ffmpeg produces not what expected when simply trying to get file info [11:12] <guy> https://gist.github.com/f65fb9b045c0096cb7a4 [11:12] <guy> am I doing something wrong? [11:13] <guy> it is complaining that second file paramter is needed to start encoding. but I am simply trying to get audio info [11:16] <mystica555_> -i is to designate input file, not give information. i personally recommend the 'mediainfo' package for what you are neeidng [11:16] <mystica555_> *needing [11:16] <JEEB> or ffprobe [11:17] <JEEB> which comes with ffmpeg [11:17] <JEEB> also you're getting audio information, no? [11:17] <mystica555_> JEEB: good point [11:20] <mystica555_> tbh i havent even messed with ffmpeg outside of the actual ffmpeg/ffplay commands [12:12] <pierow> Hello, I was wondering if anyone could give me some help with AVCodecContext values for video and audio encoding? Is there a reference where I could find good default values for encoding different audio and video codecs? [12:12] <pierow> Even just a point to somewhere in ffmpegs code would be very helpful :) [12:13] <pierow> hmmm should I maybe be asking this in #ffmpeg-devel? [12:30] <gpannwitz> http://pastebin.com/yYY9yVRw I am trying to encode videos for burning to DVD, but the produced files are way too big. I tried setting -b to something small, but it seems to be ignored. [12:49] <sacarasc> gpannwitz: First, you're not using ffmpeg but libav... Anyway, remove -sameq. [15:10] Action: insider is away: auto-away [16:01] <superdump> ahoy [16:09] <pwlnw> does ffmpeg or libav have option for calculate proportional output video size ? i need convert multiple videos and don't want calculate [16:22] <Mavrik> pwlnw, yes [16:22] <Mavrik> use the "scale" filter [16:22] <Mavrik> -vf scale=-1:720 [17:11] <pwlnw> Mavrik so i need calculate scale ? i want just use like "not more than 320x200". and ffmpeg automatically calculate sizes [17:37] <pwlnw> Mavrik seems not work. i have libav : avconv -i defy.flv -y -v verbose -vf scale=-1:320 -codec:a libfaac -ac 2 -ar 48000 -vcodec libx264 -pre:v slow -profile:v high -level 30 out.x.flv [17:38] <pwlnw> Mavrik it doesn't support scale calculation ? [17:42] <pwlnw> -vf scale=320:-1 - works ok. but -vf scale=-1:320 not [17:58] <iive> don't use libav. [18:00] <relaxed_> pwlnw: quote the filter [18:02] <pwlnw> relaxed_ don't understand how [18:02] <pwlnw> -vf 'scale=-1:320' ? [18:03] <relaxed_> yes, does that work? [18:04] <pwlnw> [libx264 @ 0x14a64e0] width not divisible by 2 (569x320) [18:04] <pwlnw> better but not work [18:06] <relaxed_> -vf 'scale=ceil(-1):320' [18:07] <pwlnw> same [18:09] <relaxed_> well, -vf 'scale=570:320' [18:10] <pwlnw> but i can't calculate size any time [18:11] <relaxed_> most of the time it works [18:13] <cbreak> mplayer's -vf scale supports rounding to full 2 or 16 multiples [18:14] <cbreak> maybe ffmpeg does too [18:15] <relaxed_> ffmpeg only has -1, where mplayer has -2+ [18:21] <spalo> can we join two videos using ffmpeg? [18:21] <relaxed> spalo: look at concat in the man page [18:23] <spalo> thanks [18:27] <pwlnw> i need send exactly coding x264 params which i see on another video. what i need to do ? i can read it from programe MediaInfo. but can't send for example 'qpmin' [18:27] <pwlnw> or 'ref=4' [18:27] <cbreak> pwlnw: strings -n10 moviefile [18:28] <cbreak> that will give you the full details [18:28] <cbreak> (maybe better than mediainfo) [18:28] <pwlnw> i already get it. but can't find how to send from ffmpeg to x264 lib [18:29] <pwlnw> not all params available from ffmpeg? [18:29] <pwlnw> and mediainfo.exe can copy to text [18:30] <pwlnw> many of params i find [18:32] <pwlnw> left is qpmin=0 qpmax=69 . [18:33] <pwlnw> --qpmin 10 --qpmax 50 : Unrecognized option '-qpmin' [18:35] <gpannwitz> sacarasc: thanks so much :D [18:35] <pwlnw> should be put options into preset file ? [18:39] <pwlnw> x264 says : Option qpmin not found. [18:40] <pwlnw> but x264 --fullhelp | grep qpmin says --qpmin <integer> Set min QP [0] [18:44] <pwlnw> http://mailman.videolan.org/pipermail/x264-devel/2010-November/007995.html. ok [22:03] <NovaSt0rM> heyo [22:06] <NovaSt0rM> i've googled the hell out of this but i still can't find an answer: when using a custom streaming IO buffer, and returning EOF when there's no more data, av_read_frame() returns a partial frame, but not an error. I can clearly see that EOF is reached, but more data will be forthcoming. Is there any way to force av_read_frame to re-try the entire read once more data is available? Currently, [22:06] <NovaSt0rM> it's returning another broken frame (the 2nd half of the first frame) on the next call. [22:08] <NovaSt0rM> aaand i just noticed the topic. [22:08] <Mavrik> ^^ [22:13] <NovaSt0rM> lol "Questions about using FFmpeg or developing with the libav* libraries should be asked in #ffmpeg" [22:13] <NovaSt0rM> their topic sends me here :P [22:14] <NovaSt0rM> wonder if using av_grow_packet to append the next packet to the previous broken frame would work cleanly [22:15] <NovaSt0rM> seems kinda sketchy though [22:16] <vadim_> NovaSt0rM: i think your questions should be addressed to developers in #ffmpeg-devel [22:17] <Mavrik> NovaSt0rM, problem is we mostly don't know ins and outs of latest ffmpeg APIs :\ [22:20] <rainmaker1> vadim_: Hi, just to tell you that ffmpeg crashed after 30 min... [22:22] <NovaSt0rM> the API base is quite a beast .. [22:22] <vadim_> rainmaker1: bad news... thank you... [22:23] <rainmaker1> vadim_: I have two ideas right now, first have you tried to get source other then network (ie. file) [22:23] <rainmaker1> vadim_: second, haveyou tried vlan? [22:23] <rainmaker1> vadim_: sorry VLC :D [22:23] <vadim_> rainmaker1: yeap... it worked very well [22:24] <vadim_> relaxed: i eman the file source [22:24] <vadim_> i mean* [22:25] <rainmaker1> vadim_: aaa the file source, that's interesting, I thought to make a setup (if feasible) to capture traffic using tcpdump then pipe it to ffmpeg [22:29] <vadim_> rainmaker1: vlc - the anwser is yes and no )) yes - i used it a few monthes ago on windows as a all in one solution for transcoding -> streaming, same udp source... it crashed irregularely, but i was unable to find out why... [22:30] <vadim_> rainmaker1: and no - haven't tryed it under linux [22:33] <vadim_> rainmaker1: on bug tracker there is a ticket regarding our truble, even i'm very unfamiliar with *nix programing, i think the idea described there makes sense... https://ffmpeg.org/trac/ffmpeg/ticket/609 [22:34] <rainmaker1> vadim_: I will try to specify pkt_size to see if it helps. I am running your example as it crashes more often then my :D [22:35] <vadim_> rainmaker1: lol... ok... but i did such tests, this does not mean you should not do them ))) guess what was the result?.. [22:35] <rainmaker1> vadim_: yeah, everything worked just fine? :D [22:36] <vadim_> rainmaker1: would i be here than? )) [22:38] <rainmaker1> vadim_: Just kidding, but I think you just love to pull out your hair for the bug that just don't exist :) [22:41] <vadim_> rainmaker1: really? )) [22:45] <rainmaker1> vadim_: :))) Now to be serious a little :) Can you tell me if michaelni reporduced this bug? [22:45] <vadim_> rainmaker1: yes [22:45] <rainmaker1> vadim_: and? [22:46] <vadim_> rainmaker1: and... made a patch, wich didn't help much.... [22:47] <rainmaker1> vadim_: and he knows about it, or that fixed the bug at his test server? [22:48] <vadim_> rainmaker1: probably he did not wait for 30+ minutes to see the result )) [22:51] <rainmaker1> vadim_: pitty... [22:51] <vadim_> rainmaker1: in fact, he resolved the problem ))) it does not "hang" any more... it just exists... [22:54] <rainmaker1> vadim_: you see, it's fixed :) [22:56] <clockwize> hey, is there a way to make ffmpeg require no input and not output the progress of the conversion? if the file exists, i just want it to overwrite, etc (running ffmpeg from another piece of software) [22:57] <rainmaker1> clockwize: -y ? [22:58] <vadim_> clockwize: about output, -loglevel quiet [22:58] <clockwize> ok, thanks guys [23:02] <rainmaker1> vadim_: new record :) frame=45190 fps= 25 q=28.0 size= 219582kB time=00:30:05.64 bitrate= 996.2kbits/s dup=16 drop=0 [23:03] <vadim_> rainmaker1: what about this record?.. [23:03] <rainmaker1> vadim_: well it's beyound 30 min :) [23:04] <vadim_> rainmaker1: aaa..... i see )) [23:05] <rainmaker1> vadim_: about your example command, did you actually use sparse or dense mode for multicast? [23:07] <vadim_> rainmaker1: if you'd tell me the difference, may be i will answer your question... [23:08] <rainmaker1> vadim_: maybe I shoud first ask if your source is some webcam or mpeg2 ts ? [23:08] <vadim_> rainmaker1: mpeg2 ts [23:11] <rainmaker1> vadim_: ok, then when you first start ffmpeg (with your example) it must send igmp join message for a multicast group. The difference is, that in sparse mode you MUST send join bla blah where in dense mode you don't have [23:11] <rainmaker1> vadim_: except in case your source is in the same vlan [23:12] <rainmaker1> vadim_: then it's pretty much the same [23:14] <rainmaker1> vadim_: right now I am using other program to send regular igmp join message as in my network I use sparse mode [23:17] <vadim_> rainmaker1: does it mean that in sparse mode ffmpeg would not read from multicast if no other program sends igmp join message? [23:21] <rainmaker1> vadim_: it means that ffmpeg must send igmp report message at regular intervals [23:23] <rainmaker1> vadim_: it's late again, I'll put ffmpeg in screen and see if it will crash till tommorow [23:24] <rainmaker1> vadim_: I will let you know here.. [23:24] <vadim_> rainmaker1: ok )) have to go to bed too ) [23:43] <lucas^> is there a list somewhere or description of which iPod Touch versions can actually play stuff encoded by -f ipod -vpre ipod320/640? [23:45] <Mavrik> pretty much any that supports Level 3.0 and baseline it seems [23:45] <Mavrik> should be all of the, [23:48] <lucas^> what's the difference between 320/640? I'm unfamiliar with the settings listed in them [23:51] <Mavrik> is there a reason you don't check yourself? [23:53] <lucas^> the reason being, I have no idea what the significance of 'level 13' is over 'level 30' [23:53] <lucas^> also, for whatever reason, ffmpeg shows the output video as high profile, which I'm assuming isn't baseline [23:54] <Mavrik> nop [23:54] <Mavrik> lucas^, "level" is a limit of macroblocks/s in a video [23:54] <JEEB> levels are basically ways of telling the decoder how much RAM it will need to decode the stream [23:54] <Mavrik> which limits your resolution + bitrate [23:54] <Mavrik> lucas^, basically you take spec from the ipod you want to support and check which max level it supports [23:55] <JEEB> they should all be level 3.0 (30) and baseline IIRC [23:55] <JEEB> only the very first ones might've had problems with level 3 [23:55] <JEEB> resolution at 320xSOMETHING [23:55] <Mavrik> yeah [23:55] <lucas^> 240p I think [23:55] <JEEB> 320x180/320x240 I guess? [23:55] <Mavrik> JEEB, that's probably the reason the ipod320 profile limits level to 1.3, ipod640 to 3.0 [23:56] <lucas^> yeah I think I remember playing 640x480 h.264 encoded videos on this thing a year or two ago [23:56] <JEEB> is the model newer or older than the iphone 3gs? [23:56] <lucas^> a macroblock is basically an arbitrary chunk of video data that needs to be decoded within a certain timeframe for realtime playback? [23:57] <Mavrik> lucas^, basically ANY iPod Touch will play ipod640 encoded videos [23:57] <lucas^> I think it's newer, could be wrong though [23:57] <Mavrik> (since even 1st gen Touches support Baseline 3.0) [23:57] <lucas^> I think the big problem is that, for whatever reason, these videos are being encoded with -profile high [23:58] <JEEB> if it's newer you could feed it pretty much anything that's within level 4/4.1 and high profile [23:58] <lucas^> the iPad can (supposedly) support high-profile level 30 video as well? [23:58] <Mavrik> For High profile you need the iPhone 4 gen Touches afaik [23:58] <JEEB> high/level4|4.1 as well for ipad [23:58] <JEEB> Mavrik, nope [23:58] <JEEB> 3GS-> [23:59] <Mavrik> ah, awesome [23:59] <Mavrik> thought 3GS only supported Main [23:59] <JEEB> nah [23:59] <lucas^> the recently encoded media isn't showing up, assumed that meant iTunes or whatever deemed it unplayable [23:59] <JEEB> yes, iTunes doesn't like [23:59] <JEEB> iTunes = will only accept what apple officially wants the device to "supporT" [00:00] --- Sun Feb 26 2012

1 0

ffmpeg-devel.log.20120224
by burek 25 Feb '12

25 Feb '12

[00:01] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r1b93244381 10ffmpeg/libavcodec/v408enc.c: [00:01] <CIA-17> ffmpeg: v408enc: switch to encode2() [00:01] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [00:01] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:01] <CIA-17> ffmpeg: 03Paul B Mahol 07master * r1f801f3a70 10ffmpeg/libavcodec/yuv4enc.c: [00:01] <CIA-17> ffmpeg: yuv4enc: switch to encode2() [00:01] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [00:01] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:01] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r946de27d25 10ffmpeg/libavutil/ (Makefile random_seed.c): [00:01] <CIA-17> ffmpeg: random_seed: add selftest [00:01] <CIA-17> ffmpeg: Reviewed-by: Nicolas George <nicolas.george(a)normalesup.org> [00:01] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:01] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r7cfd626bda 10ffmpeg/tests/fate/libavutil.mak: [00:01] <CIA-17> ffmpeg: fate: add random_seed test [00:01] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:01] <CIA-17> ffmpeg: 03Paul B Mahol 07master * rd1ede561a6 10ffmpeg/libavcodec/v308enc.c: [00:01] <CIA-17> ffmpeg: v308enc: switch to encode2() [00:02] <CIA-17> ffmpeg: Signed-off-by: Paul B Mahol <onemda(a)gmail.com> [00:02] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:20] <burek> btw, is there any replacement for -newaudio, because if I understand it's deprecated now, right? [00:20] <burek> how can people add new audio tracks in the output? [00:22] <ubitux> from what source? [00:22] <ubitux> just use -map [00:22] <ubitux> i think you already asked for this :p [00:24] <burek> did I ? :shy: [00:24] <burek> ok, I'll search the logs ^^ [00:24] <ubitux> well, you'll get the same answer i guess [00:24] <ubitux> just use -map [00:25] <burek> but if the audio input doesn't have 5 channels, but 1, can I still resolve it with -map? [00:25] <burek> i mean audio input with 1 track, to create audio output with 5 tracks [00:25] <ubitux> map it multiple times [00:25] <burek> I see :) makes sense :) [00:25] <burek> thanks :) [00:40] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r6eb12ffe0c 10ffmpeg/tests/ref/fate/random_seed: [00:40] <CIA-17> ffmpeg: fate: add forgotten random_seed ref [00:40] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:52] <vadim__> michaelni: thx for your help! but i'm still not happy :( take a look at this, please... http://pastebin.com/QmYdRMDq after "circular_buffer: overrun" the stream becomes undecodable, the reading terminates and the program exists... [00:53] <durandal_1707> vadim__: how fast is you CPU? [01:07] <vadim__> michaelni: it's a 2.4GHz intel xeon 4 cores, not sure with the model number [01:09] <durandal_1707> vadim__: durandal_1707 != michaelni [01:10] <vadim__> durandal_1707: michaelni: sorry, my mistake )) have to go to sleep olready ))) [01:54] <Compn> durandal_1707 : are you in croatia ? [01:54] <Compn> i'm looking to collect some coins from other countries... :) [01:55] <durandal_1707> Compn: no coins here [01:56] <Compn> any interesting paper money ? [01:56] <durandal_1707> Compn: you want money, how much? [01:56] <Compn> i only collect low denominations [01:57] <Compn> preferably coins or interesting paper/plastic money [01:58] <Compn> and of course i can reimburse with usd or paypal [02:01] <durandal_1707> Compn: whay currency? [02:01] <durandal_1707> s/whay/what [02:02] <Compn> i dont know what currency croatia has [02:02] <Compn> i have euros, so i dont need those [02:03] <durandal_1707> are you interested in older currencies? [02:03] <Compn> yeah [02:03] <Compn> older is cool too [02:03] <Compn> i think i have an iraq coin with saddam on it ;) [04:35] <CIA-17> ffmpeg: 03Anton Khirnov 07master * rf7fa73ac91 10ffmpeg/libavcodec/libtheoraenc.c: libtheoraenc: switch to encode2(). [04:35] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r3798205a77 10ffmpeg/ (6 files in 2 dirs): [04:35] <CIA-17> ffmpeg: mov: set channel layout for AC-3 streams based on the 'dac3' atom info [04:35] <CIA-17> ffmpeg: fixes Bug 225 [04:35] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r148fc99506 10ffmpeg/libavcodec/lclenc.c: lclenc: switch to encode2(). [04:35] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r6d9c27dc85 10ffmpeg/libavcodec/jpeglsenc.c: jpeglsenc: switch to encode2(). [04:35] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r278d88689b 10ffmpeg/libavcodec/ffv1.c: ffv1enc: switch to encode2(). [04:35] <CIA-17> ffmpeg: 03Anton Khirnov 07master * rdf53a47317 10ffmpeg/libavcodec/libschroedingerenc.c: libschroedingerenc: switch to encode2(). [04:35] <CIA-17> ffmpeg: 03Martin Storsjö 07master * rc2ff63e3ac 10ffmpeg/libavformat/ (movenc.c rtpenc.c rtpenc.h rtsp.c): [04:35] <CIA-17> ffmpeg: rtpenc: Move the trailing comma into FF_RTP_FLAG_OPTS [04:35] <CIA-17> ffmpeg: This simplifies adding more flags to the macro. [04:35] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [04:35] <CIA-17> ffmpeg: 03Martin Storsjö 07master * rada4e362b9 10ffmpeg/libavformat/rtpenc.c: [04:35] <CIA-17> ffmpeg: rtpenc: Add an error message [04:35] <CIA-17> ffmpeg: Also return a proper error code. [04:35] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [04:35] <CIA-17> ffmpeg: 03Martin Storsjö 07master * rc4584f3c1f 10ffmpeg/libavformat/ (7 files): (log message trimmed) [04:35] <CIA-17> ffmpeg: rtpenc: Allow packetizing H263 according to the old RFC 2190 [04:35] <CIA-17> ffmpeg: According to newer RFCs, this packetization scheme should only [04:35] <CIA-17> ffmpeg: be used for interfacing with legacy systems. [04:35] <CIA-17> ffmpeg: Implementing this packetization mode properly requires parsing [04:35] <CIA-17> ffmpeg: the full H263 bitstream to find macroblock boundaries (and knowing [04:35] <CIA-17> ffmpeg: their macroblock and gob numbers and motion vector predictors). [04:35] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r3c0ed7d1a8 10ffmpeg/libavcodec/asv1.c: asv1enc: switch to encode2(). [04:35] <CIA-17> ffmpeg: 03Ronald S. Bultje 07master * r2b83e8b700 10ffmpeg/libavcodec/truemotion2.c: [04:35] <CIA-17> ffmpeg: truemotion2: error out if the huffman tree has no nodes. [04:35] <CIA-17> ffmpeg: This prevents crashers and errors further down when reading nodes in the [04:35] <CIA-17> ffmpeg: empty tree. [04:35] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [04:35] <CIA-17> ffmpeg: CC: libav-stable(a)libav.org [04:35] <CIA-17> ffmpeg: rtpenc: Move max_packet_size to a context variable [04:35] <CIA-17> ffmpeg: This is in preparation for exposing this via an avoption. [04:35] <CIA-17> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [04:35] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r0ecf54f9dc 10ffmpeg/libavcodec/flashsvenc.c: flashsvenc: switch to encode2(). [04:36] <CIA-17> ffmpeg: 03Anton Khirnov 07master * ra5f848c86d 10ffmpeg/libavcodec/libxavs.c: [04:36] <CIA-17> ffmpeg: libxavs: split extradata writing out of encode_nals(). [04:36] <CIA-17> ffmpeg: This is done in preparation for the following patch implementing [04:36] <CIA-17> ffmpeg: encode2(). [04:36] <CIA-17> ffmpeg: This commit is analogous to 05d699222dd5af4f5775f9890aa825ede05a144f for [04:36] <CIA-17> ffmpeg: libx264. [06:22] <overflow_0f8b> how do you usually round a float value correctly? [06:23] <overflow_0f8b> lrintf rounds incorrectly [06:23] <overflow_0f8b> lroundf is slow as hell [06:25] <mystica555_> I am curious, how much code is shared between ffmpeg's input routines and ffplay itself [06:26] <mystica555_> specifically in the fact that ffplay doesn't understand -vcodec mjpeg when trying to play v4l2 video [06:26] <mystica555_> yet ffmpeg itself does, and properly selects either mjpeg or uyvy [06:55] <michaelni> mystica555_, -codec:v mjpeg might work [06:57] <mystica555_> lets try [06:57] <mystica555_> /dev/video0: could not open codecs [11:19] <thresh> I believe something is wrong with the git repo. [11:19] <thresh> I see the commit that removed libpostproc, and yet it's still there in current HEAD [11:22] <av500> my git says the folder is not in git [11:22] <av500> it has some stray files inside [11:24] <thresh> I've did a completely new checkout to an empty dir [11:24] <ubitux> the commit is in the merge, it doesn't mean it's been applied as-it [11:24] <av500> thresh: ah, in ffmpeg [11:24] <av500> no idea [11:24] <thresh> av500: fail :P [11:25] <thresh> ubitux: well I expected git log not to lie [11:25] <thresh> also, merge commits mentions the removal. [11:26] <ubitux> check the files affected by the conflicts [11:26] <av500> I dont see the merge commit removing it [11:26] <ubitux> yes indeed [11:27] <av500> conflict was resolved by not removing it [11:27] <thresh> ah, the merge commit log says it was removed [11:27] <av500> maybe it has dual nature [11:27] <thresh> duh [11:28] <av500> it's wave and particle at the same time [11:29] <thresh> shroedinger's postproc [11:29] <av500> thresh: when you dont look at the folder, then it's not there [11:42] <kriegerod> ubitux: hi, do you have anything to say regarding the filter i've posted? [11:43] <ubitux> hi kriegerod :) [11:43] <ubitux> well, yes i have a few things in mind, but i need to be sure of something [11:43] <ubitux> it seems to me that lavfi device isn't trying to poll before request [11:43] <ubitux> (in opposition to common usage you have in ffmpeg for instance) [11:44] <ubitux> i'm not sure why there is such difference [11:44] <ubitux> and i need saste to be available to discuss this :) [11:45] <kriegerod> i don't understand what you said ) [11:48] <ubitux> kriegerod: in ffmpeg.c, the usage is basically while (av_buffersink_poll_frame(outgraph)) { av_buffersink_get_buffer_ref(outgraph, &ref, 0); ... } [11:48] <ubitux> so it's polling from the output to check if there is data [11:48] <ubitux> and then request a buffer [11:49] <ubitux> but when using lavfi device (ffmpeg -f lavfi -i amovie=file.wav,yourfilter), there is no such call to poll_frame [11:49] <ubitux> it's directly calling get_buffer_ref() [11:49] <ubitux> so basically in your filter, the poll_frame() callback is ignored [11:50] <ubitux> and thus, i don't know how that is supposed to work, and also what's the point of poll_frame if it is not mandatory [11:51] <ubitux> kriegerod: how did you test your filter? with the api example? [11:55] <kriegerod> yes, and also with my own app [11:58] <ubitux> ok, so you called av_buffersink_poll_frame() [11:59] <ubitux> but atm, ffmpeg has no -af, it's using the lavfi device [11:59] <ubitux> so the common usage for audio filters is to use ffmpeg -f lavfi -i amovie=file.wav,yourfilter [12:00] <ubitux> and the lavfi device (libavdevice/lavfi.c) which emulates an input, doesn't call that poll_frame, and thus lead to issues with your filters [12:00] <ubitux> so i think this is something to be fixed in lavfi device [12:00] <ubitux> so you are not at fault, but i'd like to be sure of this [12:04] <ubitux> i'm wondering if the audio filters couldn't just return a samplesref with 0 samples, or just not return anything, or something [12:04] <ubitux> but well anyway, please be patient a little [12:06] <kriegerod> i got the idea [12:06] <kriegerod> in request_frame, currently it returns error when no data actually in fifo [12:06] <kriegerod> there i can add the same code from poll_frame that calls avfilter_request_frame [12:07] <kriegerod> i even thought to add it, but it seemed to me being code duplication [12:07] <kriegerod> but now i understand that is majes sense [12:08] <ubitux> in my point of view, the filter should be as simple as possible, and the polling/request system looks complicated enough as is, so i'm wondering if that's really a good thing to do [12:09] <ubitux> you'll get feedback, just need to figure out a few things [12:17] <kriegerod> i just thought that normal lavfi usage is pushing frame, then polling, if poll says there's at last 1 frame, then request [12:18] <kriegerod> because if poll is not required to be used, why it exists at all? [12:36] <ubitux> kriegerod: that's exactly what i'd like to know :) [12:36] <kriegerod> so you are planning to eliminate polling? [12:37] <ubitux> i'm planning to get an answer first :) [12:37] <ubitux> this push/poll/request system might be needed for video for some reason [12:38] <ubitux> also, i think the avfilter functions should deal with the "i haz data" on their own instead of needing the filters to implement complex callback systems [12:40] <kriegerod> is the name of the designer of avfilter known? :) [12:41] <ubitux> Vitor Sessak and Stefano worked a lot on it [12:41] <ubitux> Michael too btw [12:47] <Compn> ubitux : you should fix documentation for others after you figure push/poll out too :) [12:48] <ubitux> the current lavfi api is meant to be changed anyway [12:48] <ubitux> i'll fix the doxy if needed sure, and the doc/examples [12:49] <ubitux> the priority to me at this point is to get -af in ffmpeg [12:49] <ubitux> and this will require a lot of hack or changes [12:49] <ubitux> also, i need to work on -sf :) [12:54] <Compn> the docs mention that external ass files work with -ass, does that mean embedded mkv ass do not ? [12:54] <ubitux> yes unfortunately [12:54] <ubitux> -vf libass=file.ass [12:54] <ubitux> that's a workaround [13:01] <Compn> well getting that working with -sf would be nice , probably :) [13:01] <Compn> or however it needs to be done , without pissing of the mkv gods [13:01] <Compn> and the anime pirates [13:19] <ubitux> Compn: why do you think they would get pissed? [13:19] <ubitux> i mean, all they want is basically how to burn karaoke in the op/ed, given an ass file and mux the others; afaict, it's possible ATM [13:20] <ubitux> i don't know if the mkv muxer does it work well though [13:32] Action: Daemon404 pokes michaelni [13:37] <overflow_0f8b> hi [13:37] <overflow_0f8b> how can i give a hint to gcc for evaluating my constants if it can be evaluated ? [13:38] <overflow_0f8b> ( specifying the evaluation procedure for an inline asm function ) [13:39] <Daemon404> wait [13:39] <Daemon404> are you adding inline asm to ffmpeg? [13:42] <overflow_0f8b> i'm creating inline asm SSE optimized functions, and i'd like them to have an alternative function that gcc can evaluate at compile time. [13:42] <overflow_0f8b> or maybe optimized... [13:44] <overflow_0f8b> i don't think gcc will evaluate asm instructions [13:44] <ubitux> it's common to evalute yourself and hardcode results, with the code to generate the results under comments or #if 0 [13:44] <overflow_0f8b> ubitux<< yeah but that way you lose the possibility of gcc -o3 optimizing anything [13:45] <overflow_0f8b> for example, i am calling functions with the results of other functions [13:45] <nevcairiel> If you write asm, you hand-optimize, what would you ever want gcc to optimize it even further [13:46] <overflow_0f8b> nevcairiel<< muxrecti( muxp2i( roundfz( (po).x - rcent.x ), roundfz( (po).y - rcent.y ) ), \ [13:46] <overflow_0f8b> muxp2i( roundfz( (po).x + rcent.x ), roundfz( (po).y + rcent.y ) ) \ [13:46] <nevcairiel> that doesnt looke like asm [13:46] <overflow_0f8b> those are macros... [13:47] <overflow_0f8b> macros can contain inline asm [13:47] <nevcairiel> sounds like awful code [13:48] <overflow_0f8b> if it is not optimized then yes ;/ [13:48] <nevcairiel> macros are not function calls though, gcc should always expand them [13:48] <overflow_0f8b> yes [13:48] <Daemon404> overflow_0f8b, just note that you should not be adding new inline asm to ffmpeg [13:49] <Daemon404> but rather, as external asm files. [13:49] <nevcairiel> imho, writing the asm is more important then learning how that evil yasm macros work [13:49] <Daemon404> yes [13:49] <Daemon404> but adding more gcc-style inline asm to ffmpeg is Not Nice [13:50] <overflow_0f8b> :) [13:50] <overflow_0f8b> btw i only use macro when i have to. [15:51] <michaelni> Daemon404, pong [16:01] <Daemon404> michaelni, hey [16:01] <Daemon404> did you see my email? [16:01] <michaelni> yes, just replied [16:01] <Daemon404> o [16:01] <Daemon404> thanks [16:02] <Daemon404> hmm i figured as such [16:02] <Daemon404> michaelni, also, did you see my comment on the wavpack stuff you reverted? [16:18] Action: Daemon404 has some patches he wants to send to ffmpeg-devel but they requrie things not in ffmpeg [16:19] <ubitux> where are they? :P [16:21] <Daemon404> ubitux, sitting in libav, or on my hdd [16:21] <Daemon404> the one thing they need, is somehting micheal reverted [16:21] <Daemon404> er, michael [16:21] <ubitux> ok :p [16:23] <av500> Daemon404: send a patch I guess [16:31] <michaelni> Daemon404, what comment ? [16:37] <Daemon404> michaelni, my final fix to wavpack [16:37] <Daemon404> (fixes an error in kostya's last commit) [16:38] <Daemon404> as well as a comprehensive fate suite for wavpack (with reference hashes generated by teh reference decoder) [16:43] <michaelni> Daemon404, if there are some commits missing/wrong/or whatever, please post patches or a git branch to merge [16:44] <Daemon404> sec [16:45] <burek> Sutra u subotu, 25. februar, u Novom Sadu, protest protiv cenzure interneta: http://www.021.rs/Novi-Sad/Vesti/Protesti-protiv-cenzure-interneta-i-u-Novo… [16:45] <Daemon404> michaelni, oh you merged it now [16:45] <Daemon404> cool [16:45] <Daemon404> i can send my fate stuff then :) [16:45] <av500> burek: ides? [16:45] <burek> oops [16:46] <burek> amsg goes on all networks? :X [16:46] <burek> fck.. [16:46] <burek> i apologize, i didnt know amsg goes to all networks Im connected to :( [16:46] <burek> av500, of course :) [17:53] Action: Daemon404 goes to shower while fate-rsync runs [18:16] <kriegerod> is anybody going to transform libfaac.c to encode2? [18:17] <Daemon404> elenril is doing audio next [18:17] <Daemon404> iirc [18:17] <Daemon404> since video is all done [18:24] <durandal_1707> Daemon404: not me, i'm busy atm with other stuff [18:24] <Daemon404> huh? [18:24] <Daemon404> ruggles seems to be anyway. [19:57] <j-b> michaelni: what is the security contact of ffmpeg? [20:12] <Compn> its not ffmpeg-security [20:12] <Compn> @ffmpeg.org [20:12] Action: Compn cant remember where security stuff goes [20:48] <pasteeater> Compn: is that security through obscurity? [20:54] <Compn> pasteeater : its no one tells me anything around here :P [21:12] <michaelni> j-b, hmm, i guess just mail it to me + reimar, we need to setup some ML for that, the mplayer security list probably could be used too probably ... [21:13] <j-b> ok# [23:01] <rainmaker1> Hi, I need a little help regarding -re switch. I have a live mpeg2 stream (mpeg2video) which I want to transcode to rtmp (flv) and every time when I use -re switch ffmpeg crashes with "circular buffer overflow" error after ~10 sec, without -re after some time audio and video get out of sync. I can provide pastebin if anyone is willing to take a look.. [23:01] <rainmaker1> Oh sorry wrong channel :( [23:01] <rainmaker1> please ignore [00:00] --- Sat Feb 25 2012

1 0

ffmpeg.log.20120224
by burek 25 Feb '12

25 Feb '12

[00:00] <emile_s> Hi, I'm trying to transcode a MP4 file with multiple audio tracks. I want to keep/copy the video track (-vcodec copy) but re-encode the audio tracks from AAC to MP2 (-acodec mp2). My problem is that I'm not sure how to keep all of the audio tracks. My command-line isn't probably smart enough to do that. [00:00] <emile_s> Any ideas? (http://pastebin.com/HaAiF4pG) [00:15] <burek> emile_s, [00:16] <burek> you could use -new_audio [00:16] <burek> but i think it's deprecated now [00:17] <emile_s> burek: I'm trying that out right now. It seems to be working. [00:18] <burek> yes, just use -newaudio and after that -acodec .. [00:18] <burek> for each audio channel [00:18] <Wasmosy> im trying to encode to x264 ... http://pastebin.com/F00mvqQc [00:19] <nextime> ok [00:19] <nextime> after looking in the code on the git browser [00:19] <Wasmosy> any ideas? [00:19] <nextime> ffmpeg doesn't have any trace of the old patches to add tidsp support [00:19] <burek> Wasmosy, can you use mediainfo on that file to see what is it [00:20] <burek> it seems it's not h264 at all [00:20] <Wasmosy> burek: mediainfo? [00:21] <nextime> so, as i don't have spare time nor enough audio/video knoweledge to write the support by myself, it seems that the only way for me is to change to what i really don't like: gstreamer. [00:21] Action: nextime now is very sad. [00:21] <burek> Wasmosy, or ffprobe [00:21] <Wasmosy> ok ... [00:21] <burek> nextime, or use the old git version, which had that support? [00:26] <burek> emile_s, if you are using the latest ffmpeg, it might complaint that -newaudio is deprecated [00:26] <burek> so you'll have to use -map [00:26] <chipotle> hi, what media players on linux use ffmepg library? [00:26] <burek> chipotle, http://ffmpeg.org/projects.html [00:27] <chipotle> i want to take advantage of hd audio (e.g. dts-hd and trueHD) as my receiver supports it, and it looks like ffmpeg is the only way to do it on linux? [00:27] <chipotle> am i correct? or am i missing something? [00:27] <chipotle> ah, MythTV... yes, i've heard that does indeed do hd audio! [00:27] <chipotle> burek: thank you! do you have a recommendation for linux? it's a long list [00:27] <burek> ffplay :) [00:28] <burek> just grab the latest git source code [00:28] <burek> compile it to support dts-hd and thats it [00:28] <burek> on newer machines it takes 5-6 minutes [00:29] <chipotle> burek: cool. and it works via hdmi and/or optical? or just hdmi? [00:30] <chipotle> and this will give me a passthrough to my receiver, or does ffmpeg decode it and send it decoded right to my receiver? [00:30] <chipotle> sorry for my ignorance :/ [00:32] <nextime> burek : the old git version has a number of other things that aren't ok for me [00:33] <chipotle> burek: still here? [01:04] <pasteeater> nextime: i wonder if the author has an updated patchset or a repository that can be reviewed. [01:17] <jamiel> burek: How would you get the second branch to pass empty frames? another select that has the opposite conditions that goes to overlay? [01:18] <jamiel> re: o_0oo's problem [01:24] <nextime> pasteeater : no, it seems that he has continued gst-dsp but never done the work of the separate lib [03:10] <lalaland1125> I seem to be having are error giving me "[mjpeg @ 0xdb9090] overread 8" [03:11] <lalaland1125> This happens when the CPU maxes out and the code lags. [03:11] <lalaland1125> Any ideas on how to fix it? [03:12] <lalaland1125> (I am reading over the network) [04:04] <MistaD> How can encode to 7.1 ch AAC? it seems vo-aacenc only supports 2 channels and libfaac stops at 5.1, any ideas? [08:59] <lucas^> ffmpeg N-37208-g01fcbdf (0.10 I think) crashes immediately when I try encoding anything with libvpx [08:59] <lucas^> is this supposed to happen? [09:52] <juli1> hello [09:53] <juli1> Need help plz : what is the function to split the video and audio from ts file [09:56] <Mavrik> split? [09:57] <lucas^> ffmpeg -i your_file.ts -vn -acodec copy -f pcm_s16le audio.wav [09:57] <lucas^> ffmpeg -i your_file.ts -vcodec copy -an -f mpegts video.ts [09:59] <lucas^> ffmpeg -i your_file.ts -vn -acodec pcm_s16le -f wav audio.wav # my mistake [10:01] <Mavrik> (if the .ts file has PCM audio) [10:01] <Mavrik> ah, you've fixed it, ok :) [10:02] <lucas^> any idea why ffmpeg crashes with libvpx? [10:02] <lucas^> I'm seeing one guy on the internet claim it doesn't happen with 32-bit ffmpeg [10:03] <Mavrik> works fine on 64-bit builds here [10:03] <Mavrik> do you have the stable version? [10:04] <lucas^> I'm using ffmpeg N-37208-g01fcbdf, which was released the same day as 0.10 [10:05] <Mavrik> platform? [10:05] <lucas^> Windows [10:06] <Mavrik> ah, don't use those, so can't help :\ [10:11] <juli1> lucas^: i don't want use cmd prompt [10:12] <juli1> in fact i have a ts file and i want create a c file wich use ffmepg to convert it [10:12] <juli1> but i don't know where ffmpeg split video and audio [10:13] <juli1> to convert [10:16] <juli1> and what is the cmd to convert ts file to avi file plz? [10:16] <juli1> "ffmpeg -i input_file.ts -vcodec libxvid -b 2000k -acodec libmp3lame -ac 2 -ar 44100 -ab 128k output_file.avi" don't work [10:17] <juli1> he doesn"t know libxvid [10:17] <lucas^> make sure your ffmpeg binary is compiled with --enable-libxvid [10:19] <juli1> oh no :p [10:19] <juli1> how do? [10:19] <juli1> configure --enable-libxvid? [10:20] <lucas^> yeah, when you build it, and you'll need libxvid installed along with headers and such on your system [10:20] <lucas^> here I'll give you my build string [10:21] <lucas^> http://codepad.org/QPr0xsgn [10:36] <juli1> lucas: you have made : ./configure --enable... (all your build string)? [10:39] <lucas^> that's the Windows prebuilt binary [10:39] <lucas^> but yeah [10:40] <lucas^> basically you'll have to have a library built for most of those options [10:40] <lucas^> e.g. if you want --enable-libx264 you need x264 first [10:40] <jamiel> Morning all, we are trying to use filter graphs to split a stream, then add a color filter on a part of the stream which matches our select. However, when we run our filter we only get the part of the stream matching the select as though it's the main stream. The -vf line we are using is here: http://pastebin.com/bznGMUCF [10:41] <jamiel> Can anyone point out what is being misunderstood here about how split work? [10:43] <jamiel> With this line I'm just getting the 10 seconds of that select() playing .. rather than the whole stream and having the 10 seconds sent to overlay (so that we can add another filter onto that select and manipulate just that portion of the stream) [10:49] <juli1> lucas: do you have devlopp ffmpeg? [10:49] <lucas^> no I'm just a user [10:51] <juli1> and do you have already use function from ffmpeg in an other program? [10:51] <juli1> i need to do this [10:51] <juli1> and i'm looking for help [10:52] <juli1> i think it's not difficult [10:54] <lucas^> I have a Python program that just invokes ffmpeg with CreateProcess() or system() [10:54] <lucas^> there are ways to link directly to ffmpeg as a library but I've never done it [10:58] <jamiel> We'll offer a bounty to help us with this problem this morning. The goal is to get a transparent color over a rtmp stream for 20 seconds every 20 seconds. This is how far we are trying to grab 20 seconds of the stream to add the filter, but this ends up only showing the 20 seconds rather than showing the whole stream and selecting those 20 seconds for filtering (adding the color filter). http://pastebin.com/bznGMUCF [10:59] <jamiel> any help appreciated, thanks guys [11:02] <juli1> lucas: how set the path of libxvid with configure plz? [11:11] <gpeipman> hi! [11:11] <gpeipman> i found interesting problem when writing ffmpeg output to named pipes on windows [11:11] <gpeipman> using same mp4 as source i get correct avi when converting directly on command line [11:12] <juli1> how to specify the path of a library when i make ./configure --enable-"library" plz? [11:12] <gpeipman> when i tri to send output to named pipe then named pipe receives a little bit different bytes and the result is invalid avi [11:13] <gpeipman> in both cases bytes count is exactly the same [11:13] <gpeipman> can anybody help me find out what's going on? [11:14] <lucas^> ./configure --help should give some idea of it [11:15] <juli1> lucas^: when i do --help i can see --enabvle but not how give the path of each library [11:15] <lucas^> you should probably just install libxvid and such into /usr/local/lib, /usr/local/include... [11:56] <o_0oo> does anyone know if there is a video equivalent of the astreamsync filter? [12:10] <juli1> does someone know the id of TS file? [12:10] <juli1> CODEC_ID_TSCC? [12:11] <juli1> ou CODEC_ID_TSCC [12:16] <juli1> ou CODEC_ID_MPEG2TS? [13:56] <pom> The preset formerly known as fastfirstpass does not seem to exist any longer. What is the new name? [13:57] <JEEBsv> already there in libx264 if you specify first pass in multipass encoding [13:57] <JEEBsv> also you should generally use libx264's internal presets and not the ffpreset files :) [13:58] <pom> What do you mean by already there exactly? You mean I don't have to supply it? [13:58] <JEEBsv> yes, libx264 uses faster certain settings during first passes automagically unless you specifically tell it not to with a specific option, or if you use the placebo preset [14:01] <pom> Alrighty! I currently run this command, and it fails on a new installations: http://pastie.org/3445248 [14:01] <pom> So I should try just removing -vpre clause? [14:01] <pom> Have I got it right? [14:02] <JEEBsv> -preset:v your_desired_preset [14:02] <JEEBsv> http://mewiki.project357.com/wiki/X264_Settings#preset [14:02] <JEEBsv> listing [14:02] <pom> But... hmm, didn't you say it was automagic? [14:03] <JEEBsv> yes, I just don't know if the default preset 'medium' is what you want speed/compression wise :P [14:03] <JEEBsv> because generally you want to use the slowest preset you can take :) [14:03] <pom> Okay. Thanks a lot. [14:03] <JEEBsv> also, crf and first pass set? [14:04] <JEEBsv> I really wouldn't recommend it tbh [14:04] <JEEBsv> it's meant for cases where you will always have a second pass [14:05] <pom> JEEBsv: Maybe, tbh right back this is code I wrote two years ago and for now I'd just like to get it going on this machine again. But I'm absolutely gonna update myself on this in the near future. [14:06] <JEEBsv> are you even doing a second pass? [14:06] <pom> Always. [14:06] <JEEBsv> then just set the bitrate? [14:06] <pom> Set the bitrate on what? [14:07] <JEEBsv> on what you will be using on the second pass [14:07] <pom> But that depends on the resolution of the input. [14:07] <JEEBsv> how does that matter [14:07] <JEEBsv> I mean [14:07] <JEEBsv> you will know the resolution and will be setting the bitrate, no? [14:09] <JEEBsv> the only case where crf as the first pass rate control mode makes sense is where you run a normal (without the speed-ups) crf first pass, and then check if it goes over a certain bitrate limit. If yes, you run a second pass with the bitrate you want. [14:09] <Mattias> When streaming, ffmpeg / x264 is smart enough to not use enough bandwidth if only a small part of the screen is updated. but if the whole screen changes. how can I limit bandwidth to say just 100kb/s or something? [14:09] <pom> You know what, I can't really remember exactly the reason here. But the thinking went something along the path, that if the source was really high quality, 1080p for example, we wanted to grant a much higher bitrate. [14:09] <Mattias> I tried -b:v 30 and that didn't help much [14:09] <JEEBsv> you would still know the damn resolution before hand [14:10] <JEEBsv> Mattias: you can limit the average bitrate within a buffer [14:10] <pom> JEEBsv: Yes, haha, sorry, I don't remember exactly. Can I get back to you when I've got more time set off to optimize this aspect? :) [14:10] <JEEBsv> Mattias: vbv-bufsize and vbv-maxrate [14:10] <Mattias> thanks [14:11] <JEEBsv> no idea how those settings are written in le ffmpeg, but that should give you a good thing to google for [14:11] <pom> It was my first pass at doing something like this, and I guess from the result of it, I could use a second. [14:11] <JEEBsv> what [14:11] <pom> The code that uses that cmdln [14:12] <JEEBsv> excuse me but I can't really connect the dots of sense from your lines right now [14:12] <Mattias> JEEBsv: for buffer sizes, I'm not good with them and not sure how to set them good. maxrate is easy enough [14:12] <Mattias> any recommendations? [14:13] <JEEBsv> the amount you buffer on the playback side [14:13] <Mattias> in kb? [14:13] <JEEBsv> I have no idea how ffmpeg sets its stuff [14:14] <Mattias> looking at the x264 docs but it doesn't mention, just integer... [14:14] <JEEBsv> x264 wants kb [14:14] <Mattias> ok, thanks [14:14] <JEEBsv> ffmpge is by default bytes tho [14:14] <JEEBsv> so NFI [14:14] <JEEBsv> the idea of vbv-maxrate/bufsize is to make it so that if you buffer once the amount of bufsize, and the average transfer speed never falls below maxrate, you shouldn't need another buffering session. [14:15] <Mattias> found a site that translates from x264 to ffmpeg :) [14:19] <Mattias> hm, somehow it still caps my upload no matter how low or how high I get.. [14:20] <Mattias> clueless on what to lower -.- [14:21] <Mattias> woho, increased it by 1000 and it worked [14:23] <Mattias> JEEBsv: thanks, now it is capping :) finally [15:10] <bulll> FF ?DCC SEND ff???f?????????????? 0 0 0 [15:11] <sacarasc> Yay! It's 7 years ago again! \o/ [15:11] <JEEBsv> lol [15:12] <mystica555_> wat >.> [15:14] <bulll> ??ýDCC SEND "ff???f?ð¹ð°ð·ð³ð¶ð³ðºðŒð·ð®ðŒð¬ðº" 0 0 0 [15:16] <sacarasc> iive: ^ [15:20] <iive> too late... [15:26] <pwlnw> hi! how to ovveride cpu detection ? i have VIA Nehemiah C3 and get "Illegal instruction" message. before i get message"[libx264 @ 0x812e080] using cpu capabilities: MMX2 [15:30] <kidoz> pwlnw, rebuild ffmpeg [15:31] <kidoz> --disable-mmx2 [15:32] <pwlnw> only rebuild? [15:32] <pwlnw> can't disable it runtime? [15:35] <kidoz> Only rebuild [15:45] <pwlnw> there are no option --no-mmx2 in gcc...no-mmx? [15:47] <jamiel> does the "movie" filter source support rtmp? ie. -vf "movie='rtmp:// ... [15:48] <rainmaker1> pwlnw: 15:31 < kidoz> --disable-mmx2 [15:49] <pwlnw> rainmaker1 but gcc haven't option for disable mmx2. how it works? [15:50] <sacarasc> pwlnw: It's a ./configure option. [15:51] <rainmaker1> pwlnw: you can use ./configure --help to see a list of [15:52] <rainmaker1> pwlnw: available configure options.. [16:12] <kriegerod> can libx264 encoding work correctly with encoding AVCodecContext.time_base being 1/90000? Seems to me that no. [16:12] <Mavrik_> hmm, isn't that the default for most H.264 encoders? [16:13] <kriegerod> that's why i ask being so surprised [16:13] <pwlnw> i can't find does ffmpeg use gcc -mtune=c3-2 or not ? [16:21] <kriegerod> i've broken sth in my app. When i have encoding AVCodecContext.time_base 1/90k, x264 produces frames with QP=51.00 with few bytes in each, which results in gray screen [16:28] <pwlnw> ./configure -cpu=c3-2 produce me gcc -mtune=generic -march=i586. is it ok ? [16:45] <burek> Sutra u subotu, 25. februar, u Novom Sadu, protest protiv cenzure interneta: http://www.021.rs/Novi-Sad/Vesti/Protesti-protiv-cenzure-interneta-i-u-Novo… [16:46] <burek> i apologize, i didnt know amsg goes to all networks Im connected to :( [16:48] <kidoz> :D [17:02] <hogganjosh> hey all [17:02] <hogganjosh> What happened to https://roundup.ffmpeg.org/roundup/ffmpeg/ [17:02] <hogganjosh> I can't login anymore [17:08] <wrnestor> Can I generate a 7.1 (8 channel) aac file? I tried ffmpeg -i file1.aif -i file2[.....]file8.aif -strict experimental out.aac [17:08] <wrnestor> Got a mono [18:45] <Tostada> Hi. I'm trying to upgrade to 0.10 but get this error: root@Dell:/usr/local/ffmpeg# ./configure --enable-shared [18:45] <Tostada> yasm not found, use --disable-yasm for a crippled build [18:47] <JEEB> sounds like you either don't have yasm, or the version of yasm you have installed is too old [18:47] <Tostada> JEEB Okay, thanks. : ) [19:02] <Tostada> I'm getting a lot of warnings after running: make. Is that normal. ? [19:41] <bakers> I got a little spycam that takes mjpeg video. When I try and convert to mp4 I get: [mpeg4 @ 0x2057e00] timebase 16687/500000 not supported by MPEG 4 standard, the maximum admitted value for the timebase denominator is 65535 [19:41] <bakers> What does that mean, and how can I fix it? [19:45] <Christophe_v_E> Hey, I was wondering how I would use ffmpeg to convert a .flv file once it's downloaded - through php. [19:50] <sacarasc> bakers: It means that the FPS is weird and you should probably change it to something that can go into an MP4. [19:51] <bakers> sacarasc: Ok... how would I do that [19:52] <bakers> would it help if I posted a test clip [19:52] <sacarasc> ffmpeg -i input -r X other options output.mp4 [19:53] <bakers> sacarasc: http://www.fpaste.org/Gq5a/ [19:53] <bakers> Too many bits per frame? [19:59] <hugogee> greets all. [20:18] <hugogee> when i transcode some files with ffmpeg i get output that echoes the initial recording. e.g. one, one, one.,.. between 4-6 times. Has anyone run into this problem? [20:48] <Christophe_v_E> Hi, I'd like to split a .flv in multiple pieces and then put different pieces from different videos together. [20:48] <Christophe_v_E> How would I approach this? I have no experience with ffmpeg whatsoever and a very basic php knowledge. [21:30] <bakers> Can someone help me convert this clip to mp4? http://www.perturb.org/tmp/spycam-test.avi [21:31] <bakers> I'm having a hard time because the audio samples are 8000khz [21:31] <bakers> [aac @ 0xd732a0] Too many bits per frame requested [21:33] <bakers> Here is my command and output: http://www.fpaste.org/89iN/ [21:36] <bakers> Do I need to upsample the audio? [21:44] <SPucki> bakers: Not that i am a ffmpeg pro, but maybe you should set the output audio codec? AAC with 8khz Sampling looks odd to me. [21:48] <SPucki> bakers: changing the audio rate to 12000 would help (-ar 12000). [21:48] <bakers> SPucki: Aha! That was it! [21:49] <SPucki> bakers: Glad it helped! Now just tell me: How can i use ffserver without reencoding my mjpeg video? [21:51] <bakers> I've never used ffserver sorry [21:53] <SPucki> OK, anybody here, who can help me with a ffserver question? I have some /dev/video0 webcam which delivers mjpeg data. I just don't know how to tell ffserver not to reencode the stream! [21:55] <SPucki> "VideoCodec copy" is not valid. [22:59] Action: nextime after 1 and half day passed to right with gstreamer cause of hw limitations, i prefer to move the audio/video decoding/encoding part on another hw and come back to ffmpeg [22:59] <nextime> s/right/fight [22:59] <Mavrik> what problems did you have with gstreamer? [23:00] <nextime> Mavrik : it exists [23:02] <rainmaker1> Hi, I need a little help regarding -re switch. I have a live mpeg2 stream (mpeg2video) which I want to transcode to rtmp (flv) and every time when I use -re switch ffmpeg crashes with "circular buffer overflow" error after ~10 sec, without -re after some time audio and video get out of sync. I can provide pastebin if anyone is willing to take a look.. [23:02] <nextime> Mavrik : seriously, i don't like it, too many pieces, powerfull but at cost to be very complex sometime. From my point of view is like qmail vs postfix: the first is a bunch of little pieces and pipes that i don't like, the second is an elegant and functional software [23:03] <nextime> Mavrik : and... the only motivation to use it (for me) was the lack of an hw accelleration support in ffmpeg [23:03] <nextime> but also in gst i can't do what i need [23:03] <Mavrik> I see, never tried gstreamer yet so that's why I'm wondering :) [23:03] <nextime> so, as the only way to solve my issue is to move the audiovideo part on a separate hw with less limitations [23:03] <nextime> i can now use ffmpeg [23:05] <vadim_> rainmaker1: does it hangs or exits?.. [23:06] <nextime> now the only question for me is: can i dump audio and video output on two separate pipes from an rtsp stream? [23:07] <rainmaker1> vadim_: I would call it :hangs" [23:07] <vadim_> rainmaker1: what version of ffmpeg do you use? [23:07] <Mavrik> nextime, hmm, maybe if you define several inputs, but I'm not sure if more than one pipe is defined [23:08] <nextime> Mavrik : multiple inputs? [23:08] <Mavrik> *outputs [23:08] <Mavrik> ffs. [23:08] <Mavrik> way too late in the evenin [23:08] <rainmaker1> vadim_: today :) which was patched yesterday by michealni IIRC you were on ffmpeg-devel channel :) [23:08] Action: nextime it's 11:08pm here [23:09] <rainmaker1> vadim_: I belive your problem is solved yesterday? [23:10] <Christophe_v_E> Hi, I really can't figure out how to do anything with FFmpeg .. even after going through the manual.. I basically have my folder of ffmpeg and I want it to split .flv files in pieces of 6 to 12 seconds. I understand I need to run a command line, I have no idea how I run it and I don't know if I have to start something before it will work .. like xampp or something. Any help is REALLY appreciated .. thanks! [23:11] <vadim_> rainmaker1: for me the patch from yesterday leads to "exit" after a few seconds after "circular_buffer: OVERRUN" error... I believe the problem is not solved, or it leads to another one... [23:12] <vadim_> rainmaker1: our questions should be addressed to developers... but i have no idea how to convince them... [23:14] <Christophe_v_E> *running windows [23:14] <rainmaker1> vadim_: can you pastebin your ffmpeg command? [23:15] <vadim_> rainmaker1: sure... [23:16] <nextime> Mavrik : it work, thanks [23:16] <nextime> i didn't know that ffmpeg support multiple outputs [23:18] <vadim_> rainmaker1: do you want the cli only, or and the output? [23:19] <rainmaker1> vadim_: cli and output would be fine [23:25] <vadim_> rainmaker1: http://pastebin.com/a3Gjhsuh [23:25] Action: nextime now has to find a way to resync audio and video [23:26] <nextime> as the video will pass in opencv and of course will be slower than audio [23:28] <rainmaker1> vadim_: let me try your command.. [23:28] <vadim_> rainmaker1: you're welcome )) [23:32] <rainmaker1> vadim_: :) when I can expect that ffmpeg exits? [23:33] <vadim_> rainmaker1: about 30 seconds after overrun error [23:33] <rainmaker1> vadim_: and when I can expect overrun error? [23:34] <vadim_> rainmaker1: there is no fix time... [23:35] <rainmaker1> vadim_: in minutes/hours or days? :D [23:36] <vadim_> rainmaker1: lol... for me it may happen in about 1 minute... some time may work longer... [23:36] <vadim_> rainmaker1: does it work ok for you?.. [23:37] <rainmaker1> vadim_: yes, it still works.. but only five minutes.. I'll wait a little longer to see if anything happens.. [23:37] <rainmaker1> vadim_: the bad thing is that the same error happened to me after 21 hours :) [23:38] <vadim_> rainmaker1: michaelni, belives it is related to the system speed / configuration... [23:39] <rainmaker1> vadim_: whats your cpu? [23:40] <vadim_> rainmaker1: 2,4 GHz, Xeon, 4 cores... not sure about the model [23:42] <rainmaker1> vadim_: that's faster then my [23:43] <rainmaker1> vadim_: OS? [23:43] <vadim_> relaxed_: ubuntu 10.04. yours? [23:47] <vadim_> rainmaker1: ubuntu 10.04. yours?.. [23:51] <rainmaker1> vadim_: kubuntu blah blah 11.10 I guess [23:52] <vadim_> 64 or 32 bits? [23:57] <rainmaker1> vadim_: 32 I have some strange problems with 64 [23:57] <rainmaker1> vadim_: and you? [00:00] --- Sat Feb 25 2012

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