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April 2012

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ffmpeg-devel.log.20120424
by burek 25 Apr '12

25 Apr '12

[02:43] <CIA-17> ffmpeg: 03Martin Storsjö 07master * r269cb6751b 10ffmpeg/libavformat/mpegts.c: (log message trimmed) [02:43] <CIA-17> ffmpeg: mpegts: Try seeking back even for nonseekable protocols [02:43] <CIA-17> ffmpeg: The mpegts demuxer reads 5 KB at startup just for discovering [02:43] <CIA-17> ffmpeg: the packet size. Since the default avio buffer size is 32 KB, [02:43] <CIA-17> ffmpeg: the seek back to the start will in most cases be within the [02:43] <CIA-17> ffmpeg: avio buffer, and will in most cases succeed even if the actual [02:43] <CIA-17> ffmpeg: protocol isn't seekable. [02:43] <CIA-17> ffmpeg: 03Diego Biurrun 07master * r9b75ae05cf 10ffmpeg/libswscale/input.c: swscale: K&R formatting cosmetics (part III) [02:43] <CIA-17> ffmpeg: 03Dale Curtis 07master * r8336eb6f85 10ffmpeg/ (libavformat/matroskadec.c tests/ref/seek/lavf_mkv): (log message trimmed) [02:43] <CIA-17> ffmpeg: matroska: Add incremental parsing of clusters. [02:43] <CIA-17> ffmpeg: Reduces the amount of upfront data required for cluster parsing [02:43] <CIA-17> ffmpeg: thus decreasing latency on seek and startup. [02:43] <CIA-17> ffmpeg: The change in the seek-lavf_mkv FATE test is due to incremental [02:43] <CIA-17> ffmpeg: parsing no longer reading as much data as the old parser and [02:43] <CIA-17> ffmpeg: thus not having that additional data to generate index entries [02:43] <CIA-17> ffmpeg: 03Diego Biurrun 07master * ra229d6c285 10ffmpeg/ (7 files in 3 dirs): [02:43] <CIA-17> ffmpeg: Remove libdirac decoder. [02:43] <CIA-17> ffmpeg: libschroedinger is the preferred way to decode Dirac video. [02:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r0ca4414d0f 10ffmpeg/libavformat/au.c: [02:43] <CIA-17> ffmpeg: padding for SIMD-optimized functions. [02:44] <CIA-17> ffmpeg: 03Luca Barbato 07master * r1381081cdb 10ffmpeg/avconv.c: [02:44] <CIA-17> ffmpeg: avconv: fix off by one check in complex_filter [02:44] <CIA-17> ffmpeg: nb_input_files is already an invalid index. [02:44] <CIA-17> ffmpeg: 03Dale Curtis 07master * r7521c4bab2 10ffmpeg/libavformat/matroskadec.c: (log message trimmed) [02:44] <CIA-17> ffmpeg: matroska: Clear prev_pkt between seeks. [02:44] <CIA-17> ffmpeg: The new incremental parser doesn't always clear prev_pkt, [02:44] <CIA-17> ffmpeg: however the packet queue is cleared when seeking. Which leads [02:44] <CIA-17> ffmpeg: to a use-after-free. [02:44] <CIA-17> (2 lines omitted) [11:13] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r9c99bc433e 10ffmpeg/libavcodec/mpc8.c: [11:13] <CIA-17> ffmpeg: mpc8: print error messages on maxband(s) errors [11:13] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:14] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r88ee2aa5ad 10ffmpeg/libavcodec/mpc8.c: [11:14] <CIA-17> ffmpeg: mpc8: make maxband check less picky. [11:14] <CIA-17> ffmpeg: Fixes Ticket1245 [11:14] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [12:20] <mkoch> hi! I just came form #ffmpeg. I'm building a plugin which uses ffmpeg as a lib to process video streams. can I make ffmpeg to use vda on Mac? [12:21] <mkoch> http://ffmpeg.org/doxygen/trunk/vda_8h-source.html <-- I found this [13:04] <mkoch> is that possible that this VDA works in 32bit mode only? [13:06] <Compn> sorry mkoch , i dunno about vda [13:10] <michaelni> mkoch, dilaroga is our vda expert, he isnt here though ATM ... [13:13] <mkoch> thanks, I try to wait for him [14:00] <funman> mkoch: why do you think it is 32 bits only ? [14:01] <funman> http://dilaroga.free.fr/vlc_vda/vlc-2.1.0-git.dmg is built with VDA [15:59] <CIA-17> ffmpeg: 03Nicolas George 07master * r62b745a6d3 10ffmpeg/ffmpeg.c: [15:59] <CIA-17> ffmpeg: ffmpeg: do not report EOF as an error. [15:59] <CIA-17> ffmpeg: Not all processing goes through filters, [15:59] <CIA-17> ffmpeg: so processing can continue even when [15:59] <CIA-17> ffmpeg: all filtergraphs have reached EOF on all sinks. [16:45] <mkoch> funman: it was my fault, I missed to link a lib [16:45] <mkoch> now it seems to work, but I still have no proof that it's using VDA actually [16:45] <mkoch> how can I find it out in runtime? [16:47] <CIA-17> ffmpeg: 03Nicolas George 07master * rfd2cd64fc9 10ffmpeg/libavfilter/avfiltergraph.c: (log message trimmed) [16:47] <CIA-17> ffmpeg: avfiltergraph: fix format selection. [16:47] <CIA-17> ffmpeg: The old code had two bugs: [16:47] <CIA-17> ffmpeg: For audio filters, the format was not set. [16:47] <CIA-17> ffmpeg: For video filters, if several links reference the same format list, [16:47] <CIA-17> ffmpeg: the same format must be selected in the end. This is done by [16:47] <CIA-17> ffmpeg: setting formats->format_count to 1: the other links sharing [17:06] <michaelni> mkoch, put a av_log or abort() or something in the vda code, if you see the effects of that its being executed beyond doubt [17:24] <CIA-17> ffmpeg: 03Carl Eugen Hoyos 07master * rae0591f913 10ffmpeg/libavcodec/shorten.c: Add missing newline in shorten decoder. [19:07] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r44bd2f90a9 10ffmpeg/ffmpeg.c: [19:07] <CIA-17> ffmpeg: ffmpeg: Fix initial dts for streamcopy, if the input packets lack dts. [19:07] <CIA-17> ffmpeg: Fixes Ticket1138 [19:07] <CIA-17> ffmpeg: Based on a patch by: Eelco Lempsink <eml(a)tupil.com> [19:07] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:58] <funman> http://mailman.videolan.org/pipermail/vlc-devel/2012-April/088604.html [20:04] <j-b> this is quite out of topic, if I may say. [20:07] <Daemon404> wtf [20:08] <funman> j-b: well VLC is still the biggest distributor of FFmpeg [20:08] <funman> or perhaps it's Chrome? [20:08] Action: Daemon404 senses some dirty laundry is being aired in public, but noone has any idea whats going on [20:09] <ubitux> nice drama, up to ffmpeg/libav/mplayer* community :) [20:13] <j-b> funman: I believe it is Chrome. [20:13] <j-b> funman: even if they use libav, and strip it down quite a bit. [20:13] <Daemon404> they still use ffmpeg iirc. [20:14] <ubitux> so maybe even BBB@google is using ffmpeg without admitting it [20:14] <ubitux> :° [20:15] <ubitux> i wonder if steam is not one of the biggest distributor too ;) [20:15] <Daemon404> since when does steam distribute it? [20:15] <ubitux> ./Program Files/Steam/bin/avformat-52.dll [20:15] <ubitux> ./Program Files/Steam/bin/avutil-50.dll [20:15] <ubitux> ./Program Files/Steam/bin/avcodec-52.dll [20:16] <ubitux> quite a while :p [20:16] <Daemon404> interesting. [20:17] <JEEB> hah, never noticed that either [20:17] <ubitux> iive mentionned it a few months ago [20:17] <Daemon404> and id imagine more people have chrome than steam. [20:47] <Compn> j-b : offtopic flames are always on topic here ;D [20:48] <Compn> and wow thats a strange looking mail :D [21:15] <Compn> funman : so are you and remi going pistols at dawn ? [21:15] Action: Compn is always in the mood for a duel [21:15] <funman> would be nice [21:15] <Compn> 10 paces ? [21:15] <funman> one last night to write our final maths theorems [21:15] <Compn> blindfolded. [21:16] <funman> I chose water pistols [21:16] <Compn> bah [21:17] <funman> but I'll take a high pressure kärsher [21:18] <Compn> naked, on the cold tundra ? [21:19] Action: Compn should write a movie or something [21:24] <Compn> funman : btw your link in that mail requires a login [21:24] <Compn> :P [21:24] <Compn> the irc logs i think [21:25] <funman> he just removed it [21:25] <funman> (the login) [21:25] <Compn> Resolving dionoea.chewa.net... 91.121.6.101 [21:25] <Compn> Connecting to dionoea.chewa.net|91.121.6.101|:80... connected. [21:26] <Compn> HTTP request sent, awaiting response... 401 Authorization Required [21:26] <Compn> Authorization failed. [21:26] <funman> ah it was here for a few seconds [21:27] <funman> http://dionoea.chewa.net/irc/ yep it's all deleted , must have been my browser's cache acting [21:28] <Compn> that remi's server ? [21:28] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r2b336df3cb 10ffmpeg/libavutil/parseutils.c: [21:28] <CIA-17> ffmpeg: parseutils: reliably detect out-of-range alpha. [21:28] <CIA-17> ffmpeg: This should fix the FATE test on ARM (not tested), [21:28] <CIA-17> ffmpeg: but it should also detect alpha values like 2^128 [21:28] <CIA-17> ffmpeg: reliably as invalid which would be another out-of-range [21:28] <CIA-17> ffmpeg: case with implementation-dependant behaviour. [21:28] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [21:34] <funman> yeah it's shared between a bunch of ex ECP students [21:40] <Compn> whoa [21:40] <Compn> http://mailman.videolan.org/pipermail/vlc-devel/2012-April/ is all 404 [21:41] <Compn> mailman.videolan.org that is [21:41] <j-b> yep [21:41] <Compn> j-b : you guys having fun over there ? [21:41] <j-b> Compn: not really. [21:42] <iive> looks like this is contagious. Mplayer infected FFmpeg, and now it spreads [21:42] <Compn> wait how can you blame mplayer for this ? :P [21:42] <Compn> oh wait, nm i remember [21:42] <funman> no it's quite far from being fun [21:43] <iive> it is never fun when it is in your house. [21:43] <j-b> Compn: I dislike people insulting each other. [21:45] <Compn> j-b : well i wish it would all work out for everyone. [21:45] <j-b> Compn: it won't. [21:45] <j-b> Compn: cause at the end of this, VLC dies. [21:45] <Compn> iive : i think we should blame google instead :P [21:46] <Compn> or fsf :P [21:46] <iive> why? [21:46] <j-b> Compn: cause I will quit the VideoLAN project, and believe me the project will die slowly. [21:46] <ubitux> huh? [21:46] <Compn> j-b : yeah, i know you work pretty hard at it :( [21:46] <ubitux> why would you do that? [21:47] <j-b> ubitux: just to annoy you and your cat :) [21:47] <iive> ubitux, you have a cat? [21:47] <ubitux> :D [21:47] <ubitux> no i don't iive :) [21:47] <Compn> videolan has really owned the video player market for far too long [21:47] <j-b> Compn: that is totally untrue. [21:47] <Compn> its too easy to use :) [21:47] <ubitux> i have a logitech mouse though [21:48] <Compn> ubitux : how big is your /usr/bin/cat ? [21:48] <Compn> 1/11/1999 04:00a 22,016 cat.exe [21:48] <ubitux> HUGEZ [21:49] <Compn> cat (GNU textutils) 2.0 [21:51] <Compn> Daemon404 : you think more people have chrome vs steam ? [21:51] <Compn> he concurrent users peak was 5 million on January 2, 2012. (steam) [21:54] <Compn> Google Chrome Browser Has 200 Million Users [21:54] <Compn> whoa [21:54] <Compn> guess so ;) [21:54] <j-b> We are at around 100M [21:55] Action: gnafu uses VLC, doesn't use Chrome or Steam. [22:33] <VooDooNOFX_> To followup my earlier conversation, r3d appears to now have a version 2 of their file format with new cameras. [22:46] <Compn> VooDooNOFX_ : well samples and specs and patches welcome :) [23:49] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r3158160334 10ffmpeg/libavcodec/ffv1.c: [23:49] <CIA-17> ffmpeg: ffv1: move the decoder side ac->vlc switch to decode_slice() [23:49] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:49] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * re37419a806 10ffmpeg/libavcodec/ffv1.c: [23:49] <CIA-17> ffmpeg: ffv1: move ac->vlc switch to encode_slice() [23:49] <CIA-17> ffmpeg: This will allow us to use vlc coding in 1.3 too [23:49] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:49] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r44f789c092 10ffmpeg/libavcodec/ffv1.c: [23:49] <CIA-17> ffmpeg: ffv1: allow selecting 1.3 by using -level [23:49] <CIA-17> ffmpeg: Note, 1.3 is not finalized and the bitstream will still change [23:49] <CIA-17> ffmpeg: do not use it yet. This option is just to make playing with it [23:49] <CIA-17> ffmpeg: easier, otherwise one would have to edit the source [23:49] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:49] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rfe5bc46f60 10ffmpeg/libavcodec/ffv1.c: [23:49] <CIA-17> ffmpeg: ffv1: support vlc mode in 1.3 [23:49] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Wed Apr 25 2012

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ffmpeg.log.20120424
by burek 25 Apr '12

25 Apr '12

[00:22] <kami8785> quick question: can i cut .flv files without re-encoding them? approximation through file size would be good enough for me [00:30] <burek> kami8785, try ffmpeg -i input.flv -t 10 -codec copy out.flv [00:30] <burek> that will produce 10 seconds of original without reencoding [00:31] <kami8785> i tried that, while the video codec remains the same, some settings still change and youtube barfs if i upload [00:31] <kami8785> i can head -c 10M test.flv and the headers remain intact [00:31] <kami8785> cutting from the front, no such luck apparently [00:32] <burek> no no [00:32] <burek> head will just cut the file off [00:32] <burek> rendering it invalid [00:33] <kami8785> works fine for me [00:33] <kami8785> im getting it from rtmpdump [00:33] <kami8785> so probably not a perfectly valid file to begin with [00:34] <taqattack> Could someone explain to me why my output fps drops when I'm trying streaming to RTMP? http://pastebin.com/7Yd2X94e [00:37] <taqattack> The encoded file is at 30 fps H264/MP3. Checked with Gspot. [00:39] <burek> kami8785, I don't know, but -vcodec copy -acodec copy -scodec copy should make an identical copy of the input [00:39] <burek> combined with -t you should get just a part of it [00:46] <taqattack> Is librtmp being truncated in favour of native rtmp? [00:46] <kami8785> burek, didnt know about the scodec, seems to work now, i will test on some other flvs, thanks [00:46] <burek> :beer: [01:05] <kami8785> burek, yup there it is again, youtube gives "Failed (unable to convert video file)" after ffmpeg -i own2.flv -ss 5 -t 20 -acodec copy -vcodec copy -scodec copy own2out.flv, it failed on 2/4 uploads [01:07] <kami8785> ffmpeg output of one of the files http://pastebin.com/S6Titey4 [01:07] <burek> kami8785, maybe your 10 seconds contains little or no keyframes [01:07] <burek> or something [01:08] <burek> anyway, if you use just -vcodec copy [01:08] <burek> there is no point placing -ss after the -i [01:08] <burek> because you are not re-encoding your video [01:08] <burek> so it doesn't help you if ffmpeg actually gets your position more accurate [01:08] <burek> if it is in the middle of 2 keyframes [01:09] <burek> it won't be able to successfully do -vcodec copy [01:09] <kami8785> i will try again with 60 seconds and -ss before -i [01:09] <burek> try ffmpeg -ss 5 -i own2.flv -t 60 -acodec copy ... [01:09] <burek> yes, exactly [01:10] <burek> also, try opening your video in a video editor to find out where are your keyframes [01:10] <burek> so you can see where you can cut it without reencoding [01:10] <kami8785> it needs to be automated [01:11] <burek> good luck with that, then [01:11] <kami8785> haha [01:12] <kami8785> its part of a much larger app, but it simplify things a lot if i can load off video cutting to ffmpeg [01:12] <kami8785> ive also thought about just doing the cutting in youtube, since it has a video editor [01:12] <burek> or you can always reencode it and get it as accurate as you need :) [01:12] <kami8785> but id have to figure out the session token and it also takes a good amt of processing time before its ready [01:13] <kami8785> hmm with the new settings it seems to work again [01:13] <teratorn> why would ffmpeg need to do anything other than re-encode a single key-frame, and all the inter-frames before the NEXT keyframe, where it could resume vcodec=copy ? [01:13] <kami8785> more testing! [01:14] <kami8785> how can i re-encode to make sure youtube accepts it? [01:14] <kami8785> as a last-resort if copy doesnt work [01:14] <teratorn> burek: does that sound feasible to you? [01:15] <burek> teratorn, because you explicitly told it not to do that? [01:15] <teratorn> burek: well that's fine, I'm just wondering if it's possible in theory [01:16] <burek> well, I agree that's the best approach, but encoding process needs parameters for quality/speed/rate [01:16] <burek> so, it would have to be specified somehow, or detected from input, etc [01:16] <teratorn> yeah.. [01:17] <burek> kami8785, install mediainfo and check youtube files' properties [01:17] <teratorn> BUT, can you actually instruct ffmpeg to follow those semantics... re-encode until first keyframe then vcodec=copy? [01:17] <burek> and then use various params to encode your video and check with mediainfo if they match :) [01:17] <teratorn> I assume re-encoding will always re-encode all the frames [01:17] <burek> teratorn, I personally don't know, but you can always ask in devel channel #ffmpeg-devel [01:18] <kami8785> burek, hmm that seems to be the most solid solution [01:18] <burek> or you can read the source code [01:18] <kami8785> let me see if someone has found out the params that marry yt + ffmpeg [01:18] <burek> ok :) [01:18] <teratorn> yeah, I'll read source :) - I have need of some editing features soon, so I'll probably have to answer this question, or implement support if it's not already there [01:18] <teratorn> assuming we' don't end up using GStreamer Editing Services instead :/ [01:19] <burek> teratorn, that would be a great feature [01:19] <burek> I can tell already [01:19] <burek> a lot of people have downloaded some videos encoded with, say, -gop 250 at 25fps [01:19] <burek> so you get a key frame every 10 seconds [01:20] <burek> and then they come asking why doesn't -ss work properly.. [01:20] <burek> (without reencoding) [01:20] <teratorn> heh [01:22] <teratorn> personally I never go over 9.8 seconds/KF [01:22] <burek> I had to do -gop 25 once [01:22] <burek> @25fps [01:22] <burek> :/ [01:22] <teratorn> heh [01:22] <burek> because stupid web flv player wouldn't wanna play video immediately [01:22] <burek> even with grey/discolored image [01:23] <burek> it would wait until a keyframe, so people often turned it off, thinking it doesn't work [01:23] <teratorn> in my webm-streamer, I just don't send any damn data until a keyframe comes up [01:23] <teratorn> which is pretty quick, so it's not a problem [01:24] <teratorn> chrome chokes like hell if you don't start it on a keyframe [01:26] <burek> it would really be cool to have ffmpeg as a web browser plugin :) [01:26] <burek> instead of flash and such :) [01:34] <kami8785> yeah ffmpeg is amazing [01:35] <kami8785> im a fabrice bellard fanboy ever since i found out he created it [01:46] <kami8785> burek, these new settings i cant shake a stick at, everything just works :) thanks alot [01:46] <kami8785> any way i can use -ss to get to cut to a keyframe? [01:47] <kami8785> woops [01:47] <kami8785> back [02:11] <Hugo5863> I have a problem, can anyone here help? [02:12] <Hugo5863> Start a stream with ffmpeg I get an error that says "Unrecognized option 'lag-in-frames'" [02:13] <Hugo5863> "/usr/local/share/ffmpeg/libvpx-360p.ffpreset: Invalid option or argument: 'lag-in-frames=16', parsed as 'lag-in-frames' = '16' [02:13] <Hugo5863> " [02:35] <iive> Hugo5863: that file is text one, remove the line containing it. [02:35] <iive> it is likely libvpx own option and the version you have had changed/removed it. [02:36] <Hugo5863> Ok, I'll try but last time I tried to edit it, it tikd me I couldn't [02:36] <Hugo5863> Told [02:36] <iive> Hugo5863: you may try to find out if they just renamed it, and change it to the new name. [02:37] <iive> Hugo5863: probably wants root access :) [02:37] <Hugo5863> How do I give myself root access to something I'm modifying in the text editor? [02:38] <Hugo5863> (I'm just going to comment the line out) [02:39] <Hugo5863> I wouldn't know how to find out if they changed the name of the file. [02:44] <Hugo5863> Nvm, did a search and found it [02:45] <Hugo5863> Alt+f2 to open a seach command line then enter 'gksudo nautilus' then after type my password fot a root admin window to do my edits with [02:49] <Hugo5863> I commented out everything that was causing an error, now it's a new error. [02:50] <Hugo5863> Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height [02:51] <Hugo5863> I'm trying to use a quality preset for the stream. If I remove the preset the computer doesn't vomplain. [02:51] <Hugo5863> complain [02:54] <Hugo5863> The odd thing is I'm using the libx264 encoder, why do I have to use presets for libvpx? [02:54] <Hugo5863> I don't see the libx264 presets [02:58] <Hugo5863> Ok, I had the thought that I'm missing the presets instead of that being what I'm supposed to use.. [02:58] <Hugo5863> Downloading the missing presets [02:59] <Hugo5863> Going to put them in the file it was referencing for the incorrect preset [02:59] <phunyguy_> hello everyone, I have a specific situation I need help with. I have a video with 24bit DTS-HD Master Audio, and I would like to convert to raw wav file to encode as 24 bit 7.1 FLAC. Currently ffmpeg (avconv?) only sees it as 16bit DTS Core. Is there an alternative means to make this happen? [03:00] <phunyguy_> TrueHD works perfect, just cant get DTS-HD MA to work [03:01] <iive> Hugo5863: hum? i guess there is more than one wrong thing in your command line ;) [03:05] <phunyguy_> the ffmpeg -i video.mkv produces output with "Stream #0.1(eng): Audio: dca (DTS-HD MA), 48000 Hz, 5.1, s16, 1536 kb/s (default)" as input audio stream. [03:06] <phunyguy_> should be DTS-HD MA, 48000Hz, 7.1, s32. [03:08] <Hugo5863> It's not letting me extract the presets from the zip file [03:09] <Hugo5863> Each onr has a lock symbol next to it in the zip file [03:14] <Hugo5863> I'm pretty sure at this point I just need to get these files from the zip into my presets folder but ubuntu isn't letting me [03:55] <Hugo5863> Ok, I figured it out myself. The problem WAS that ffmpeg is missing the presets for libx264. [03:56] <Hugo5863> I had to use the iPod presets that are for libx264 to get the stream to work. [04:38] <tlvb> is it possible to stream a prerecorded intro seamlessly followed by a realtime input (webcam, ipcam)? [05:52] <intangir> how do i change only the container format? and not the video/audio streams (just move from mp4 to avi or something?) [05:52] <intangir> without actually changing the video encoding [05:54] <intangir> i think i found out, brb gonna test it on blueray ;) [05:59] <intangir> no dice [06:57] <aphid> trying to figure out the rtmp path to get the main feed from http://www.media.mit.edu/events/medialabtalk/ (in the source, this stuff is defined in the loadMidSD() function) so i can rip this thing tomorrow. everything thusfar i've tried via ffmpeg has thrown "Closing connection: NetStream.Play.StreamNotFound" [08:20] <ciao> ciao [12:14] <mkoch> hi! how can I find out if my ffmpeg uses vda on Mac? according to the config.h vda support is on, but when I try to play a video ffplay uses 10 times more CPU then QT, so it seems that actually it doesn't use video decode acceleration [12:16] <Mavrik> as far as I know ffplay doesn't use hardware acceleration yet [12:16] <Mavrik> other apps that use ffmpeg libs do [12:18] <mkoch> then what is this? http://ffmpeg.org/doxygen/trunk/vda_8h-source.html [12:18] <mkoch> and I'm actually building an app which uses ffmpeg az a lib, so is there a way then to use hw accel? [12:18] <Mavrik> this is a hook for those apps so they get data pre-packaged for acceleration [12:18] <mkoch> (or should I go to the dev channel with this question?) [12:19] <Mavrik> so they can pass it over to HW decoding APIs [12:19] <Mavrik> probably yes [12:19] <Mavrik> or even better - libav-users mailing list [12:19] <mkoch> okay, thanks anyway! :) [16:02] <praedo> i'm trying to encode a NTSC DV file and i only get sound when played in windows media [16:02] <praedo> this is the command i use: [16:02] <praedo> ffmpeg -i file.avi -vf yadif=0:1 -vcodec libx264 -preset veryfast -crf 22 -threads 0 -acodec libfaac -aq 100 file.mp4 [16:02] <praedo> it's a project encoded to .avi (NTSC DV profile) in premiere [16:03] <praedo> and i want to encode the .avi into .mp4 [16:03] <praedo> if i encode it with mpeg streamclip, it works with audio+video [16:03] <praedo> but with ffmpeg i get blank video [16:03] <praedo> it works with PAL and the only difference is that it's NTSC [16:03] <praedo> do i need any extra option to that command line? [16:15] <TimNich> praedo: can you put the complete uncut console output somewhere? [16:42] <b_jonas> thanks for ffmpeg! [16:42] <b_jonas> I've just decoded a video successfully [16:44] <praedo> TimNich, what do you mean by uncut? [16:49] <TimNich> as in don't be tempted to trim it down, everything from the header message to the end of the coding stats.. [16:51] <kai> hi folks [16:53] <kai> I'm trying to stream live video from a webcam to another PC with as low a latency as possible. what would be the ideal way to do this? [16:54] <kai> I'll have 100Mbit ethernet available, and a dual-core centrino system connected to the web cam [17:32] <ckb> hey guys, is there a way I can encode a video with adobe's h264 codec? I'm trying to add DRM to some video files, but apparently it doesn't support libx264 [17:45] <iive> ckb: drm to video files? [17:46] <ckb> iive: yes, I'm trying to use DRM with my video& but EzDRM doesn't support libx264, only FLV.. [17:46] <iive> flv is container, x264 is encoder... [17:47] <ckb> yeah, I'm aware [17:47] <ckb> I have a fly file encoded with libx264 and libfaac [17:48] <ckb> but when I try to pack the file for DRM, it tells me the video format isn't supported [17:48] <ckb> flv*** [17:55] <barhom> anyone that uses ffserver, please check ticket: https://ffmpeg.org/trac/ffmpeg/ticket/1249 - have you ever noticed this problem? [18:04] <ckb> Apparently, I have to use Adobe to encode my videos.. so I can't use ffmpeg to encode ;( [18:06] <Rik316> I'm curious. What's the opaque member in libavformat? [18:27] <ckb> APPARENTLY& I can only use Adobe's codecs with what I'm trying to do [18:31] <barhom> does anyone know any web server that can accept ffmpeg output "-f flv http://webserver/somefile.flv" [18:31] <barhom> and then stream that live stream via HTTP GET standard port 80 (wget http://webserver/somefile.flv) [18:34] <aphid> ...adobe has codecs? [18:34] <ckb> aphid: that's what I don't understand about this [18:34] <ckb> aphid: you think you could help me wrap my head around this? [18:35] <ckb> EzDRM says that the files need to be in a f4v container h264/aac [18:35] <ckb> in order to put the DRM header content in there [18:36] <ckb> everything is great and dandy, as I can encode videos into f4v& but the problem lies with me needing 5 or 6 different outputs for the video (bitrates/resolutions) [18:36] <aphid> the time I spend working with flv is usually spent trying to get into open formats. wouldn't even know where to start trying to get it into it. [18:36] <ckb> sigh [18:37] <aphid> like all this public domain gov't produced video they're using rtmp to stream cruddy flvs... i guess i should be relieved only a few of them are still doing RealVideo [18:39] <ckb> why can't ffmpeg output to f4v? [18:39] <aphid> ckb - http://en.wikibooks.org/wiki/FFMPEG_An_Intermediate_Guide/h.264 mentions f4v as a valid output [18:40] <ckb> but... [18:40] <aphid> "h.264 is arguably the cutting edge of video compression codecs. Unfortunately it is copyright protected." just hurt my brain. [18:41] <ckb> Unable to find a suitable output format for 'output.f4v' [18:41] <ckb> yep [18:42] <aphid> copyrights, patents, whatever [18:47] <ckb> just don't understand any of this [18:47] <ckb> DRM is the worst idea ever [18:47] <aphid> why are you trying to use it? [18:47] <phunyguy_> that is an opinion. [20:02] <taqattack> I have a question. How do I add timestamp to ffmpeg. IT says -timestamp now has been deprecated [22:27] <praedo> TimNich, still here? [22:35] <praedo> [15:02] <praedo> i'm trying to encode a NTSC DV file and i only get sound when played in windows media [22:35] <praedo> [15:02] <praedo> this is the command i use: [22:35] <praedo> [15:02] <praedo> ffmpeg -i file.avi -vf yadif=0:1 -vcodec libx264 -preset veryfast -crf 22 -threads 0 -acodec libfaac -aq 100 file.mp4 [22:35] <praedo> [15:03] <praedo> it's a project encoded to .avi (NTSC DV profile) in premiere [22:35] <praedo> [15:03] <praedo> and i want to encode the .avi into .mp4 [22:35] <praedo> [15:03] <praedo> if i encode it with mpeg streamclip, it works with audio+video [22:35] <praedo> [15:03] <praedo> but with ffmpeg i get blank video [22:35] <praedo> [15:04] <praedo> it works with PAL and the only difference is that it's NTSC [22:35] <praedo> [15:04] <praedo> do i need any extra option to that command line? [22:35] <praedo> ... [22:35] <praedo> can i give you a chunk of the dv file for you to test please? [22:39] <praedo> i see it can be played in vlc but not in windows media [22:39] <praedo> i don't think it's an encoding problem [22:39] <praedo> maybe something related to ntsc playback in windows media... [00:00] --- Wed Apr 25 2012

1 0

ffmpeg-devel.log.20120423
by burek 24 Apr '12

24 Apr '12

[07:43] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2012-April/123616.html [07:43] <ubitux> http://lists.libav.org/pipermail/libav-devel/2012-April/026542.html [07:44] <ubitux> funny to see again some ostrich policy [07:44] <ubitux> and funny to see that even the reviews are the same [07:46] <ubitux> i want my internet back to commit stuff :( [08:01] <bryno> ubitux: interesting, indeed. [08:05] <ohsix> ostritch policy? [08:06] <ubitux> certainly a french expression; i meant "restraining from looking around" policy [08:06] <ubitux> (ostritch with the head in the floor) [08:06] <ohsix> i got that, but i'm not sure what you mean, that the typo was missed? [08:07] <ubitux> i don't see credits to nicolas, or luca taking into account the typo reimar fixed at first [08:08] <ohsix> ah [08:08] <ubitux> the patch was ignored from ffmpeg again, and "re-invented" [08:08] <ohsix> well they said they won't take anything from it, but they probably have someone looking to duplicate the work :] [08:09] <ubitux> no, when it happens they at least keep the credits [08:09] <ohsix> luca might have seen someone apply it to gentoo [08:10] <ubitux> anyway, not the first time this stuff happens, it's too bad [08:10] <ubitux> but well, too bad for them. [09:53] <ubitux> a lot of mp3 overread in this sample, can anyone confirm http://kagerow.net/mp3/sakura.mp3 ? [09:56] <ubitux> playback is fine though [09:56] <ubitux> i guess it's been badly encoded [10:29] <VooDooNOFX> When examining the magic headers of some recent r3d files, it appears they changed their magic number from DRED1 to fRED2. Does anyone know if this has ever been brought up? [10:33] <TimNich> fRED2 or DRED2 ? [10:34] <av500> Red Said Fred [10:34] <VooDooNOFX> hexedit shows 00 00 01 66 52 45 44 32 05 04 52 32 [10:34] <VooDooNOFX> ...fRED2..R2 [10:35] <ubitux> lavf demuxer checks for "RED1" [10:36] <ubitux> looks like they just bump the version to introduce new stuff [10:36] <VooDooNOFX> yep, r3d_probe tests for RED1 [10:37] <ubitux> 00 00 01 66 is the size [10:38] <ubitux> the fact it is 'D' or 'f' doesn't matter, just a coincidence [10:40] <VooDooNOFX> is that a U32 size? or R32? [10:41] <ubitux> u32 afaict [10:41] <ubitux> (big-endian) [10:41] <TimNich> I remember ARM's 'misuse' of hex to put 'DEADBEEF' to indicate a defunct file..... [10:43] <VooDooNOFX> interesting. this file is around 3.1gb [10:48] <ubitux> it doesn't look like the red1 fmt supports 64 bits atom sizes [10:48] <ubitux> maybe the red2 added that [10:52] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r4d87001096 10ffmpeg/libavcodec/vp8.c: [10:52] <CIA-17> ffmpeg: vp8: fix crash due to skiped update_dimensions(). [10:52] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [10:52] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [10:54] <VooDooNOFX> i'll update the magic string, recompile ffprobe and run it on the files i've got. Maybe it can still be read. [13:16] <CIA-17> ffmpeg: 03Nicolas George 07master * r73bc019baf 10ffmpeg/libavformat/utils.c: lavf: print a warning if probesize seems not enough. [14:30] <CIA-17> ffmpeg: 03Nicolas George 07master * r78ac49c233 10ffmpeg/libavfilter/avfilter.h: avfilter: document request_frame return codes. [14:30] <CIA-17> ffmpeg: 03Nicolas George 07master * re0761feec4 10ffmpeg/libavfilter/ (avfilter.h avfiltergraph.c): avfilter: add a pointer from links to graph. [14:30] <CIA-17> ffmpeg: 03Nicolas George 07master * r2d6522bab0 10ffmpeg/libavfilter/ (vsrc_buffer.c vsrc_buffer.h): vsrc_buffer: count the number of failed requests. [14:30] <CIA-17> ffmpeg: 03Nicolas George 07master * re11110dee4 10ffmpeg/libavfilter/ (buffersink.h sink_buffer.c version.h): buffersink: add AV_BUFFERSINK_FLAG_NO_REQUEST. [14:30] <CIA-17> ffmpeg: 03Nicolas George 07master * r99f29108d3 10ffmpeg/ffmpeg.c: [14:30] <CIA-17> ffmpeg: ffmpeg: directly request frames from filters. [14:30] <CIA-17> ffmpeg: It allows ffmpeg filtering logic to work with filters [14:30] <CIA-17> ffmpeg: that do not implement the poll_frame method, [14:30] <CIA-17> ffmpeg: such as split or tile. [14:30] <CIA-17> ffmpeg: 03Nicolas George 07master * r2ce7972779 10ffmpeg/libavfilter/ (5 files): [14:30] <CIA-17> ffmpeg: avfiltergraph: add avfilter_graph_request_oldest(). [14:30] <CIA-17> ffmpeg: Keep a heap of all sink links ordered by timestamps. [14:30] <CIA-17> ffmpeg: 03Nicolas George 07master * rf75ee4031e 10ffmpeg/libavfilter/vsrc_buffer.c: [14:30] <CIA-17> ffmpeg: vsrc_buffer: return EAGAIN if no frame is available. [14:30] <CIA-17> ffmpeg: This is not an erroneous condition, do not print a warning. [14:36] <ubitux> oh, awesome :) [14:51] <CIA-17> ffmpeg: 03Nicolas George 07master * r9246896f77 10ffmpeg/ (3 files in 3 dirs): parseutils: add FATE test. [14:51] <CIA-17> ffmpeg: 03Nicolas George 07master * r9b84f8a987 10ffmpeg/ (libavutil/parseutils.c tests/ref/fate/parseutils): parseutils: add av_parse_time() test. [14:51] <CIA-17> ffmpeg: 03Nicolas George 07master * r43b7068070 10ffmpeg/libavutil/parseutils.c: [14:51] <CIA-17> ffmpeg: parseutils: make av_parse_time() check for failure. [14:51] <CIA-17> ffmpeg: Until now, av_parse_time() would accept "1:00" as "1" [14:51] <CIA-17> ffmpeg: and silently ignore ":00". [14:51] <CIA-17> ffmpeg: This patch also includes a few cosmetic changes. [18:16] <CIA-17> ffmpeg: 03Nicolas George 07master * r1e43786b4c 10ffmpeg/doc/print_options.c: print_options: warn that the files are generated. [20:14] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r068d0b4e25 10ffmpeg/libavcodec/h264_sei.c: [20:14] <CIA-17> ffmpeg: h264: some fields in SEIs are longer than 25 bits thus use get_bits_long() [20:14] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [20:14] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:15] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r53253a0b83 10ffmpeg/tests/ (Makefile fate/mapchan.mak): (log message trimmed) [21:15] <CIA-17> ffmpeg: Simplify asynth sample generation. [21:15] <CIA-17> ffmpeg: One rule can be used to generate all asynth files. [21:15] <CIA-17> ffmpeg: Requires renaming the mapchan files though. [21:15] <CIA-17> ffmpeg: Also switch to using the .wav variants for mapchan [21:15] <CIA-17> ffmpeg: while changing the name anyway, this allows getting rid [21:15] <CIA-17> ffmpeg: of the explicitly specified format. [23:03] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r2954574126 10ffmpeg/libavcodec/indeo4.c: [23:03] <CIA-17> ffmpeg: indeo4: check quant_mat more fully. [23:03] <CIA-17> ffmpeg: quant_mats valid range depends on the block size. [23:03] <CIA-17> ffmpeg: This fixes a global array overread. [23:03] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [23:03] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:26] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rc799e46231 10ffmpeg/tests/ (Makefile codec-regression.sh regression-funcs.sh): [23:26] <CIA-17> ffmpeg: Get rid of tests/data/asynth-%.sw rule. [23:26] <CIA-17> ffmpeg: This also avoids an issue with parallel make in some [23:26] <CIA-17> ffmpeg: cases never building asynth-16000-1.sw. [23:26] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [23:49] <CIA-17> ffmpeg: 03Nicolas George 07master * r2e8f19d26f 10ffmpeg/libavfilter/avfiltergraph.c: avfiltergraph: free the sink_links heap. [00:00] --- Tue Apr 24 2012

1 0

ffmpeg.log.20120423
by burek 24 Apr '12

24 Apr '12

[00:10] <SpaceBass> hey folks, may be a very basic question, but racking my brain on this... [00:10] <SpaceBass> where can I download the preset files? [00:11] <Mavrik> preset files for what? [00:11] <SpaceBass> a software package I'm using is looking for presets such as "slow" and "medium" [00:11] <SpaceBass> Mavrik, ffmpeg? [00:11] <SpaceBass> .ffpreset files specifically [00:11] <Mavrik> well, preset files are just stored command-line parameters [00:12] <Mavrik> so... the answer is "anywhere" [00:12] <Mavrik> what are you trying to do? [00:13] <SpaceBass> Mavrik, I'm trying to get a package called MythExport to work, basically it transcodes recordings made by mythtv. all of the docs and forum posts refer to a few specific presets like they are part of the ffmpeg install, and not provided by the myth export package [00:13] <Mavrik> ok. [00:14] <Mavrik> let's try this again: TO WHAT are you trying to transcode your videos and where did you hear about those presets? [00:15] <SpaceBass> guessing I didn't answer the right question, although think I'd likely answer it the same way given the way you've re-posed it [00:15] <SpaceBass> I am trying to convert to m4v, h264 files and I heard about them in the docs and forum posts related to the mythexport package [00:16] <SpaceBass> those docs/posts refer to presets such as "slow" "slowfirstpass" and "medium" and all I have are these: http://pastebin.com/XXz34k3k [00:16] <Olson> i've got some presets there http://olson.hopto.org/ffmpeg [00:16] <Mavrik> SpaceBass: ah, I see [00:17] <SpaceBass> Mavrik, I'm pretty novice with this stuff, if it's not obvious :) appreciate the hand holding :) [00:17] <Mavrik> SpaceBass: there are no more separate presets for libx264 (H.264) anymore - ffmpeg now invokes them directly from x264 library [00:17] <Olson> for me they live in /usr/local/share/ffmpeg [00:17] <SpaceBass> Olson, that's where mine are too [00:17] <Mavrik> SpaceBass: you have to use "-preset <preset>" parameter instead of "-vpre <preset>" [00:17] <Mavrik> SpaceBass: you also don't need any preset files anymore [00:17] <SpaceBass> Mavrik, good to know, thanks [00:18] <Olson> cool [00:18] <Mavrik> (but that's libx264 specific, that's why I wanted to know what your goal is) [00:18] <Mavrik> SpaceBass: this might help you: http://www.virag.si/2012/01/web-video-encoding-tutorial-with-ffmpeg-0-9/ [00:19] <SpaceBass> to make matters even more complicated, the ffmpeg arguments are generated by, what I believe is, a perl script herE: http://pastebin.com/kcY9d0jZ [00:20] <SpaceBass> Mavrik, thanks, I'll read up [00:20] <Mavrik> SpaceBass: you'll have to replace "-vpre" with "-preset" [00:20] <SpaceBass> Olson, thanks for your d/l, it appears some of the ones the MythExport package is looking for are in that zip. I'll experiment with altering MythExport's pearl as well as using the ones you provided [00:20] <Mavrik> probably lower the CRF to ~25 if you want any quality [00:21] <Mavrik> and lose the "-f ipod" if you're not encoding for (now really obsolete) iPods [00:21] <SpaceBass> Mavrik, thanks, will try both. Target is a h264 encode for AppleTV and iPad, so no need for iPod support [00:22] <SpaceBass> I was trying with -vpre libx264-lossless_medium, since I have that preset in the filesystem. now that I know I need -preset, is that still a valid approach, or should I use the medium preset from Olson 's zip? [00:23] <Mavrik> use the medium preset [00:23] <Mavrik> (and you don't need any .ffpreset files - presets are "baked" into libx264 now) [00:23] <SpaceBass> thanks much! [00:23] <Mavrik> (I'm guessing your ffmpeg is reasonably new) [00:23] <Olson> I haven't updated and build ffmpeg for a while.. so.. is that right when I do I need change my command lines a bit [00:23] <SpaceBass> ah, so should I just 86 that parameter all together? [00:24] <SpaceBass> yep, using latest git clone for both ffmpeg and x264, both compiled today [00:24] <Mavrik> ah ok, just pass in "-preset medium" and don't worry about the files [00:24] <Mavrik> Olson: well if you're still using preset files for H.264, your ffmpeg is pretty ancient [00:25] <Olson> it could be uhm.. 6 months? [00:25] <Mavrik> there were quite a few changes from that time [00:25] <Olson> more perhaps [00:25] <Mavrik> more. at least your build is older :) [00:25] <Olson> I kinda.. because I do screencasting if it aint broke I dont fix it kinda thing [00:25] <Mavrik> mhm [00:26] <Olson> is say built 22 aug 2011 [00:26] <Olson> so.. yeah :P [00:26] <Olson> i hadn't realised quite it be that long [00:27] <SpaceBass> just tried running by hand with: bigsky:~$ ffmpeg -i /local/media/LiveTV/recordings/1351_20120416210000.mpg -y -vcodec libx264 -vpre medium -crf 28 -threads 0 -s 960x540 -acodec libfaac -ab 96k -ac 2 -aspect 16:9 ~/test.m4v [00:27] <SpaceBass> and got "File for preset 'medium' not found" [00:27] <Mavrik> SpaceBass: replace "-vpre medium" with "-preset medium" as I said. ;) [00:27] <SpaceBass> ugh! Can't believe I missed that again [00:27] <SpaceBass> :) [00:28] <SpaceBass> worked! thanks Mavrik! [00:29] <Olson> soo.. if I updated via GIT I would be getting what FFmpeg 0.10.2? [00:30] <Olson> oh well.. I guess I could checkout that version eh [00:30] <Olson> but.. [00:30] <Olson> nevermind I didn't think it through [00:30] <SpaceBass> ffmpeg version N-40057-g2ea5f86 [00:32] <Mavrik> Olson: you'd be getting something newer than 0.10.2 :) [00:33] <Olson> yeah :P [00:54] <SpaceBass> inspired to lean the arguments better? reading up on crf ?. is lower better? [00:58] <Olson> depends what you mean by "better" [00:58] <SpaceBass> image quality [00:58] <JEEB> the idea of crf is to find the highest number that still looks good for you [00:58] <JEEB> lower numbers try to go closer to the source (at the expense of bit rate) [00:58] <SpaceBass> helpful, thanks! [00:59] <JEEB> (and higher ones in theory go "further" from the source) [00:59] <Olson> I use erm.. [00:59] <JEEB> (and thus use less bit rate) [00:59] <Olson> 22 [00:59] <JEEB> well, if that looks good for you that would sound like you would be getting relatively good compression :) [01:23] <SpaceBass> wow, getting 267 fps on a 1080p source file using -preset superfast [01:26] <JEEB> yes, libx264 can be as quick or slow as you wish [01:27] <JEEB> thus the general guideline is to use the slowest preset you can bear [01:27] <SpaceBass> just not used to seeing those kind of frame rates when I use handbrake ? anxious to see the quality of this file when it's done, guessing its going to be pretty poor [01:28] <JEEB> if it's crf and the value is around your liking it should be fine, what could be bad is possibly the compression ratio [01:31] <SpaceBass> ah [02:17] <Rik316> char *start = strchr(uc->filename, ','); [02:17] <Rik316> Can someone explain to me why there would be a comma in the middle of a url? right after the protocol specification... [02:17] <Rik316> and if there was a comma specifically... why would that be important [02:30] <Rik316> I disced, did anyone answer my question [02:57] <elkng> I did built "ffmpeg-0.6.3" version and now trying to build "ffmpeg-0.10.2" version using same options for "configure" but it says "Unknown option:" "--enable-avfilter-lavf" "--enable-libfaad", where are they ? how can I enable "libfaad" ? [03:18] <iive> elkng: i think libfaad have been removed in favor of the internal aac decoder [03:19] <iive> not sure what the first option is. [03:19] <elkng> no need for faad2-2.7 package in system ? [03:20] <iive> other programs may use it, but ffmpeg won't. [03:44] <elkng> one time I run "configure" it prints: "License: LGPL 2.1" another time it prints: "License: nonfree and unredistributable" [03:44] <elkng> why are they differ ? [04:14] <elkng> what about "libfaac" was it replaced by internal version ? [04:19] <taqattack> I'm getting this error when I'm trying to merge two audio sources into one using my build of ffmpeg: http://pastebin.com/Jbi1vZ1J . But Using Zeranoe build of FFmpeg I don't. Am I missing any libraries? [04:21] <taqattack> i believe its "libvo_aacenc" now. You need to enable-gpl and enable-nonfree to compile those. [05:15] <diracdelta> Hi, the documentation for ffmpeg claims the existence of a -tile filter, but when i run ./configure --list-filters, it does not appear as an option. Is this a planned feature? [08:04] <elkng> when I use version of ffmpeg 0.6.3 while converting video in a CPU with one core but with hyperthreading the command "top" shows there is less than 95-98% of CPU's load, [08:04] <elkng> now using 0.10.2 version on same CPU the command "top" shows its 160-180% load, is this version used more threads ? [08:04] <elkng> when I did used version "ffmpeg 0.6.3" I did run 2 parallel ffmpeg processes for recoding 2 video files be cause there is hyperthreading and it should be faster, but if now one ffmpeg procees used hyperthreading ability is it worth to run 2 ffmpeg processes ? [08:12] <elkng> there are features listed in changelog added after 0.6 they are: "VP8 frame-multithreading" "Frame multithreaded decoding" and "H264/MPEG frame-level multi-threading" is that it ? [08:23] <toulouse> Question: I want to write a Java library I can embed in an application to take a mp4 and emit an HLS playlist - but I don't know the first thing about mpeg 2 transport streams. How involved and how resource-intensive is segmenting on the fly? Is lazily generating the segments tractable? [08:30] <toulouse> welp, doesn't look like i'll get an answer here [08:31] <elkng> "This channel is publicly logged", where are the logs ? [16:02] <l-b> hello [16:08] <l-b> I have tried this script to save the first frame of a video file: http://stackoverflow.com/a/10042169 [16:08] <l-b> my command line is like this: [16:09] <l-b> /cygdrive/c/programs/ffmpeg/bin/ffmpeg.exe -i avideo.mp4 -ss 0 -vframes 1 -vcodec mjpeg -f image2 avideo.jpg [16:10] <l-b> I get the error "video.mp4: No such file or directory" - why is it cutting off the first letter of the name? [16:17] <l-b> it might be a quirk of cygwin... I call that in a loop generating lines on the fly and it failed ~80% of the time. after piping the generated commands into a file and executing that, everything's fine. a cygwin/bash question then, perhaps. [17:11] <FiSHNUGGET> I am trying to get video duration of a file on s3. FFMPEG works on all files on my local machine, but not on files stored in s3. The output of the command is seen here http://pastebin.com/tnRHisgF . Any help would be great& [17:12] <Mavrik> FiSHNUGGET: you're using really obsolete version of ffmpeg [17:12] <Mavrik> plus, for file info, "ffprobe" is used now [17:13] <Mavrik> and thus ffmpeg executable demands output [17:14] <FiSHNUGGET> it does work on local files though [17:14] <FiSHNUGGET> but you're saying use ffprobe instead off ffmpeg -i? [17:14] <Mavrik> yes. [17:14] <Mavrik> you probably have different versions installed locally and on S3 [17:15] <FiSHNUGGET> it doesn't need to be installed on s3. I just need it to read the file from s3 and give me back the duration. There is an example of it here http://stackoverflow.com/questions/2677317/how-to-read-remote-video-on-amaz… but I can't get it to work. [17:16] <FiSHNUGGET> let me see what happens with ffprobe [17:22] <FiSHNUGGET> hmm&. no such file reported from ffprobe. But I do see it was configured w/ --disable-network. Would this prevent communication w/ S3? [17:26] <Mavrik> most likely ;) [17:26] <Mavrik> considering you have to use https. [17:26] <Mavrik> *http [17:31] <andrew__> If I am piping the output of ffmpeg to a video player, would 3gp or mp4 be a suitable type? [17:32] <andrew__> Are there extra flags that I need call on the cmd line to make the output suitable? [17:33] <Mavrik> hmm, mp4 probably won't be [17:33] <Mavrik> since it has file info atoms at the end [17:33] <Mavrik> don't know much about 3gp structure though [17:34] <andrew__> ok, so I should investigate 3gp you think? [17:41] <FiSHNUGGET> ffprobe works on the remote server. I guess they have a newer version of ffmpeg. [17:41] <FiSHNUGGET> thanks for the tip [20:28] <praedo> hello [20:29] <praedo> i'm trying to encode a NTSC DV file and i only get sound when played in windows media [20:29] <praedo> no video [20:29] <praedo> but if i encode it with mpeg streamclip, it works with audio+video [20:30] <praedo> this is the command i sue: [20:30] <praedo> use [20:30] <praedo> ffmpeg -i file.avi -vf yadif=0:1 -vcodec libx264 -preset veryfast -crf 22 -threads 0 -acodec libfaac -aq 100 file.mp4 [20:30] <praedo> it worked with PAL DV [20:31] <praedo> what do i have to change to work with NTSC DV? [20:36] <Mavrik> uhhm [20:37] <Mavrik> praedo: you're creating an mp4 file, not a .dv file [20:37] <Mavrik> basically you're encoding into H.264/AAC in MP4 [20:37] <Mavrik> there's nothing DV connected here [20:38] <Mavrik> NTSC DV would be created by ffmpeg -i file.avi -vf yadif=0:1 -target ntsc-dv file.dv [20:38] <praedo> no, input is already dv [20:38] <praedo> i don't need to export to dv [20:39] <praedo> it's a project already encoded to .avi (NTSC DV profile) in premiere [20:40] <Mavrik> I see. [20:40] <Mavrik> So what exactly do you want to do with it? [20:42] <praedo> encode the .avi into .mp4 [20:42] <praedo> as i usually did with that command for PAL DV projects [20:43] <praedo> i get blank video [20:43] <praedo> and the only difference is that it's NSTC [20:43] <Mavrik> command as you pasted it gives no reason for a difference between PAL and NTSC [20:44] <Mavrik> can you paste ffmpeg output when transcoding? [22:04] <burek> how to tell ffmpeg to use either full stereo or joint stereo for mp3 files output? [22:38] <caleress> hey guys, does any1 know if wmp12 can play a mpeg2video in an avi container ? [22:39] <Mavrik> I'm 90% sure it can. [22:39] <JEEB> wmp12 can handle MPEG-2, yes. The avi container could be the problem [22:39] <JEEB> (wmp12 = win7) [22:40] <JEEB> unless you are dealing with the euro version without the multimedia stuff [22:40] <caleress> yup it can play from the mpg container [22:40] <caleress> but muxed it to avi with ffmpeg and i only get audio :/ [22:41] <JEEB> recommendation of the day: don't put MPEG-2 in avi [22:41] <caleress> heh :p [22:41] <caleress> so it doesnt work ? [22:41] <Mavrik> huh, now that's downright silly behaviour [22:42] <JEEB> Mavrik, IIRC there was some funny stuff related to MPEG-2 in AVI [22:42] <JEEB> many things have some kind of specific support for it or whatever [22:42] <JEEB> I just guess MS didn't do that [22:42] <JEEB> but to be honest, it's just better off not to use AVI for MPEG-2 [22:42] <caleress> well [22:43] <caleress> i tried putting into mp4 [22:43] <caleress> same story [22:43] <Mavrik> mp2? :) [22:43] <caleress> yeah mpeg-ps work :p [22:43] <JEEB> yeah, mp4 should be theoretically fine [22:43] <JEEB> but I guess they didn't think of that [22:43] <JEEB> in other words, didn't implement it [22:44] <caleress> pretty weird, why would only mp2 work and avi+mp4 not, it's the same goddamn stream and the spec supports it [22:45] <Mavrik> if (file.ext == "mp4") { open_decoder(H.264) } [22:45] <caleress> hehe :) [22:46] <JEEB> probably their mp4 splitter just doesn't support splitting MPEG-2 [22:46] <caleress> and avi splitter doesnt as well :/ [22:46] <JEEB> as for avi, I'm not exactly sure on details but I'm not sure if the MPEG-2 decoder can do it [22:47] <JEEB> could be a splitter or decoder thing with AIV [22:47] <JEEB> *AVI [22:47] <caleress> well if i put it into mpeg-ps it plays fine [22:47] <caleress> so definitely splitter [22:49] <JEEB> with AVI there was something special [22:49] <JEEB> because a lot of decoders had options for MPEG-2 in avi [22:49] <JEEB> back in the day [22:49] <JEEB> but yeah, either splitter or decoder [22:50] <caleress> so there's no special option or something that i would need to add to ffmpeg when transforming it to avi? [22:51] <JEEB> nope [22:51] <JEEB> it should be taking care of that automagically [22:52] <JEEB> so yeah, MS's stuff will limit you to mpeg-ps and mpeg-ts [22:52] <caleress> hmm ok [22:53] <caleress> thought their players would support their own formats, but ok :) [22:54] <caleress> tnx guys [22:56] <burek> how to tell ffmpeg to use either full stereo or joint stereo for mp3 files output? [22:56] <JEEB> I would guess there's a setting for that in LAME? [22:57] <JEEB> since LAME is what you should be using when encoding mp3 [23:13] <cvprog> hello world how ccan i make a single frame into a movie i tried ffmpeg -f image2 -i ~/Pictures/here.png -i smoke.mp3 -acodec copy -sameq out.flv [23:13] <cvprog> does not work, and ffmpeg built on Feb 15 2012 07:17:25 with gcc 4.6.1 does not seem to have loop_input [23:16] <cvprog> :) [23:17] <burek> you need -loop 1 [23:18] <cvprog> burek: ill try ... [23:21] <cvprog> burek: worked :D [23:28] <burek> :beer: :) [00:00] --- Tue Apr 24 2012

1 0

ffmpeg-devel.log.20120422
by burek 23 Apr '12

23 Apr '12

[04:06] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rb21ba20cc8 10ffmpeg/libavcodec/wmaprodec.c: [04:06] <CIA-17> ffmpeg: wmaprodec: tighter check for num_vec_coeffs [04:06] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [04:06] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:52] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r2a59abf1a8 10ffmpeg/libavformat/smacker.c: [11:52] <CIA-17> ffmpeg: smackerdemux: Allocate padding for extradata [11:52] <CIA-17> ffmpeg: Fixes slight overread. [11:52] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [11:52] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [11:52] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rf7c67536fe 10ffmpeg/libavcodec/svq1dec.c: [11:52] <CIA-17> ffmpeg: svq1dec: Fix overread on very small input [11:52] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [11:52] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [12:30] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rfa5dacce14 10ffmpeg/libavcodec/indeo5.c: [12:30] <CIA-17> ffmpeg: indeo5: check against scaleable frames in non scaleable streams. [12:30] <CIA-17> ffmpeg: Fixes a null ptr dereference. [12:30] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [12:30] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [12:31] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r605f2b6b00 10ffmpeg/libavcodec/asv1.c: [12:31] <CIA-17> ffmpeg: asv1dec: check extradatasize before reading. [12:31] <CIA-17> ffmpeg: Fixes null ptr dereference [12:31] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [12:31] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:09] <CIA-17> ffmpeg: 03Nicolas George 07master * r23a6e4749f 10ffmpeg/ffmpeg.c: [13:09] <CIA-17> ffmpeg: ffmpeg: fix validity test for file index. [13:09] <CIA-17> ffmpeg: Also fix a typo in the next line. [15:05] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rabec6549ae 10ffmpeg/libavcodec/ffv1.c: [15:05] <CIA-17> ffmpeg: ffv1dec: Require a valid keyframe for decoding non keyframes. [15:05] <CIA-17> ffmpeg: Before this the context could become inconsistent, this lead to a null ptr [15:05] <CIA-17> ffmpeg: dereference. [15:05] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [15:05] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:05] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r9a4f5b7616 10ffmpeg/libavcodec/ (mjpegbdec.c mjpegdec.c): [15:05] <CIA-17> ffmpeg: mjpegbdec: check SOS/SOF ordering. [15:05] <CIA-17> ffmpeg: Fixes null ptr dereference [15:05] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [15:05] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:00] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rfd2127ad53 10ffmpeg/libavformat/wtvdec.c: [16:00] <CIA-17> ffmpeg: wtvdec: Check that stream private context has been allocated before use. [16:00] <CIA-17> ffmpeg: This fixes a null ptr dereference with attachments [16:00] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [16:00] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:00] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8aa57b7b5e 10ffmpeg/libavformat/au.c: [16:00] <CIA-17> ffmpeg: audemux: Check channels isnt 0 [16:00] <CIA-17> ffmpeg: Fixes a division by 0 [16:00] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [16:00] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:00] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r52fdaf27f7 10ffmpeg/libavformat/au.c: [16:00] <CIA-17> ffmpeg: audemux: Fix potential integer overflow leading to a division by 0 [16:00] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:26] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r22a6a23a45 10ffmpeg/ffmpeg.c: [20:26] <CIA-17> ffmpeg: ffmpeg: assert the refcount of allocated frames, [20:26] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:26] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rbe2b927a6f 10ffmpeg/ffmpeg.c: [20:26] <CIA-17> ffmpeg: ffmpeg: stricter refcount check in unref_buffer() [20:26] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:27] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r41abc9da50 10ffmpeg/libavcodec/iff.c: [20:27] <CIA-17> ffmpeg: iff: fix null ptr dereference [20:27] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [20:27] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:27] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r951cbea56f 10ffmpeg/libavcodec/mpeg12.c: [20:27] <CIA-17> ffmpeg: mpeg12dec: reset data size after parsing extradata. [20:27] <CIA-17> ffmpeg: This ended up corrupting data structures and may possibly [20:27] <CIA-17> ffmpeg: lead to a double free. [20:27] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [20:27] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:27] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rcf09496cf9 10ffmpeg/ffmpeg.c: [20:27] <CIA-17> ffmpeg: ffmpeg: assert against creation of cycles in the pools linked list. [20:27] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:27] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r2ea5f866e6 10ffmpeg/ffmpeg.c: [20:27] <CIA-17> ffmpeg: ffmpeg: assert the refcount of decoded frames [21:37] <cbsrobot> wow that's a nice bug [21:37] <cbsrobot> happens only 1 time out of 4 [21:37] <cbsrobot> http://ffmpeg.org/trac/ffmpeg/ticket/1097 [21:38] <cbsrobot> and with -threads 1 I hit it all the time [21:38] <cbsrobot> with -threads 4 only every other time [21:39] <cbsrobot> and when it succeeds, only the bottom row seems to have issues (red / green dots) [22:08] <CIA-17> ffmpeg: 03Carl Eugen Hoyos 07master * r2e07f42957 10ffmpeg/libavcodec/flicvideo.c: [22:08] <CIA-17> ffmpeg: Support flicvideo with 904 bytes extradata. [22:08] <CIA-17> ffmpeg: Fixes ticket #1234. [22:38] Action: mystica555 asks here as well, it could be considered a development question: is there something materially different between -vf filter; and -filter:v filter; ? -'vf yadif' produces expected results; '-filter:v yadif' does not- the output video is still interlaced. [22:41] <CIA-17> ffmpeg: 03Mans Rullgard 07master * r2bcbd98459 10ffmpeg/ (34 files in 6 dirs): [22:41] <CIA-17> ffmpeg: Remove lowres video decoding [22:41] <CIA-17> ffmpeg: This feature is complex, of questionable utility, and slows down [22:41] <CIA-17> ffmpeg: normal decoding. [22:41] <CIA-17> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [22:42] <CIA-17> ffmpeg: 03Yusuke Nakamura 07master * rba9869311f 10ffmpeg/libavformat/mov.c: [22:42] <CIA-17> ffmpeg: mov: Take stps entries into consideration also about key_off. [22:42] <CIA-17> ffmpeg: Splitted files don't start always from a sync sample. [22:42] <CIA-17> ffmpeg: Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [22:42] <CIA-17> ffmpeg: 03Mans Rullgard 07master * rd526c5338d 10ffmpeg/ (20 files in 4 dirs): [22:42] <CIA-17> ffmpeg: ARM: allow runtime masking of CPU features [22:42] <CIA-17> ffmpeg: This allows masking CPU features with the -cpuflags avconv option [22:42] <CIA-17> ffmpeg: which is useful for testing different optimisations without rebuilding. [22:42] <CIA-17> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [22:43] <CIA-17> ffmpeg: mov: Treat keyframe indexes as 1-origin if starting at non-zero. [22:43] <CIA-17> ffmpeg: Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [22:43] <CIA-17> ffmpeg: 03Mans Rullgard 07master * rd7458bc8c6 10ffmpeg/libavcodec/dsputil.c: [22:43] <CIA-17> ffmpeg: dsputil: remove unused functions [22:43] <CIA-17> ffmpeg: These functions were left unused by the lowres removal. [22:43] <CIA-17> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [00:00] --- Mon Apr 23 2012

1 0

ffmpeg.log.20120422
by burek 23 Apr '12

23 Apr '12

[00:00] <krobins> i have it working. i converted yuv420p to gray pixel format, but i'm looking for a little more information [00:01] <krobins> i was wondering why the video size only went from 22MB to 15MB when the pixel information went from 3x12 to 1x8 [00:02] <krobins> i would have expected the size of the video to drop by about 75% [00:03] <krobins> also, i was curious if anyone has any experience in getting the video into 4bit grayscale [00:03] <krobins> none of the pix_fmts listed seemed to have that, but it seems like it would be easily doable [00:37] <burek> elkng, hi "scientist" :) would you be so kind to offer a better text for the manual, regarding the section about -map option? :) [00:38] <burek> how difficult it is to just type -map <i want this stream> -map <i want that stream too> [05:39] <brad0> Being a bit of a git n00b can anyone assist me in trying to pull down what looks like a branch but I don't think it is? I'm trying to get the 0.7 branch. [05:44] <Olson> im wondering how I would grab system sounds via x11grab on a, shall we say standard linux system with ALSA. . if -i hw:1,0 is my mic, then is there something also I could put there to grab those sounds, say if I was playing back some video or so and I wanted that audio to be in the capture also? [05:46] <Olson> I hope it's not too much to ask but also.. It appears I could use multiple inputs with -i .. ie at the moment we have video and microphone so adding another that accepts system sounds could be done at the same time perhaps [05:48] <Olson> as I side note.. I have tried to use JACK.. for the life of me I find it confusing.. if that is the solution then.. ok i'll have to crack on with it [06:09] <Olson> it's not clear to me how to x11grab and take the mixer output in there.. mic is ok, though I not sure how to get sound from video playing on the system or so [07:13] <yjacket> I am trying to convert an MKV file to an MP4 file, but ffmpeg fails... details here: http://pastebin.com/qBK9e92y [07:16] <yjacket> **convert MKV to MP4, using the same audio and video tracks [07:16] <yjacket> I was trying the instructions here: http://forum.doom9.org/archive/index.php/t-124877.html [07:41] <briggers_> http://pastebin.com/qBK9e92y [07:41] <briggers_> bugger, sorry about that. [08:02] <yjacket> briggers_ any idea why it won't work? [08:12] <briggers_> Nope, sorry. Maybe you could upload a sample of the input file. [08:32] <briggers_> yjacket: try, ffmpeg -i old_video.mkv -sameq new_video.mp4 [10:53] <jondot> is there a way to use gstreamer to output a series (20-30) of video thumbnails from a given video, in even spacing? [10:54] <jondot> oops. ffmpeg, im trying both :) [11:03] <briggers_> jondot: You mean an animated gif? [11:04] <jondot> briggers_, either that or a series of images. just like what youtube now does (and the less politically-correct youporn) [11:05] <jondot> i can do a single frame with ffmpegthumbnailer, so i can make a loop that takes X frames and calls it each time [11:05] <jondot> since i'll be doing it on the serverside i wouldn't like to have to do that [11:06] <jondot> i'd prefer that to happen entirely inside ffmpeg or other binary [11:06] <briggers_> Maybe you can use the -t switch and the -ss switch [11:08] <jondot> -t is duration but -ss is? [11:08] <briggers_> ffmpeg -i $INPUT_VID -ss $STARTTIME -t $DURATION $OUTFILE.GIF [11:08] <jondot> but this is a single frame right? [11:08] <briggers_> -ss is the start time offset [11:09] <briggers_> You should get a gif that starts at $STARTTIME of the video and contains $DURATION lenght of the video. [11:10] <jondot> ah i see. well i wasn't intending to do that.. i was intending to make a gif or a series of images, in constant intervals in the movie [11:10] <jondot> so for example, snap a frame every 1 minute for a 60minute movie, makes a 60second preview of it [11:11] <briggers_> oh, okay. [11:12] <briggers_> If you can read php, this migh help http://www.alberton.info/video_preview_as_animated_gif_with_ffmpeg_and_spl.… [11:13] <jondot> briggers_, yep this is what i'm defaulting to do. however it involves N shell calls per video [11:15] <briggers_> Well, I dont see anything in the command line options that suggests you can do it in only one call. [11:15] <jondot> thanks.. i'll try and see if there's a way to write the glue code in C [11:17] <briggers_> well, for an N frame gif, you'll need to do only about N+1 calls, maybe that isnt too bad. [11:22] <jondot> yes. i'm kind looking at sources, like this https://github.com/lajos/iFrameExtractor/blob/master/Classes/VideoFrameExtr… [11:22] <jondot> looks like if I wanted to do such a thing, there are quite a few new things I need to be familiar with from the video/ffmpeg domain [11:23] <jondot> now i'm wondering - in terms of codecs, does ffmpeg cover all or most codecs? is there a list somewhere? [11:24] <briggers_> Try the -r switch. I think that's the one you want if you want even spacing. [11:25] <jondot> ah. thats an interesting take actually [11:25] <jondot> reduce the frame rate.. [11:25] <briggers_> I dont have access to ffmpeg right now, but the man page says the -r accepts fractional values. [11:25] <jondot> so if i go with 1/60 im looking at what i want [11:26] <briggers_> Not sure, i suppose you'll need to do a bit of trial and error. [11:26] <jondot> yup, thanks thats a good start [11:27] <briggers_> also see, http://ffmpeg.org/general.html for supported formats [12:35] <jondot> briggers_, fyi, here is the command im using: ffmpeg -i sample_iPod.m4v -r 0.2 -sameq -f image2 thumbs%02d.jpg [12:35] <jondot> works nicely [13:14] <Olson> it's not clear to me how to x11grab and take the mixer output in there.. mic is ok, though I not sure how to get sound from video playing on the system or so [13:15] <Olson> as I side note.. I have tried to use JACK.. for the life of me I find it confusing.. if that is the solution then.. ok i'll have to crack on with it [13:30] <Olson> mmm.. [14:14] <qarpfen> is there a difference in using "-c:a vorbis" and "-c:a libvorbis"? [14:18] <sacarasc> Yes. [14:23] <qarpfen> sacarasc: can you tell me about them? And do you know good options to convert DVD VOBs to libxvid? [14:24] <qarpfen> sacarasc: (except that vorbis also supports decoding and direct rendering method 1) [14:25] <sacarasc> vorbis is the internal ffmpeg one, libvorbis is the library libvorbis. [14:27] <qarpfen> sacarasc: so the code in vorbis is not written by xiph but by ffmpeg devs? [14:29] <sacarasc> Yes. [14:29] <sacarasc> I think. [14:30] <qarpfen> does anyone have good options to convert DVD VOBs to xvid? [14:32] <Olson> :( [15:02] <qarpfen> how do I set frame rate to same as source? [15:02] <qarpfen> (Why, at all, defaults frame rate to 25 and not to same as source?) [15:06] <JEEB> umm, so far as long as the source had some notion of a frame rate or time stamps it has used the source frame rate for me by default? [15:07] <JEEB> (or more correctly "source time stamps" as most formats have time stamps instead of a single "frame rate") [15:07] <Olson> how do I make x11grab record also system sounds from playing video or so? do I need JACK? linux. [15:59] <qarpfen> what is the grouping (-g) good for? Should I use it when converting to xvid? [17:05] <taqattack> Is there a way to capture sound output in windows without Stereo Mix? [17:17] <qarpfen> is it rather advisable to use vorbis or libvorbis? [18:15] <ZodiacMentor> Converting three PNGs with alpha to huffyuv with this: http://pastebin.com/MuVq09B4 and then trying to read that and write them back to png's with this code: http://pastebin.com/Dd80Jix0 avcodec_decode_video2() returns 0 in the got_picture_ptr. the code works with other avi's, like xvid. Any help appreciated! [18:26] <Olson> I make screencasts using ffmpeg and x11grab, I was wondering if whilst recording the mic I could also add something on the commandline to capture system sounds, be it some audio track or video or whatever. Using linux with ALSA [18:48] <iive> Olson: i remember when the sound cards used to have a lookback capture stream. I guess thank's to MAFIAA this functionality have been slowly phased out. [18:49] <iive> no idea if alsa or pulse support that kind of functionality. [18:51] <Olson> it'll be possible using JACK [18:51] <Olson> but that thing confuses the heck out of me [18:52] <Olson> i was hoping for a -i hw:mixer or so :P [18:55] <ZodiacMentor> I managed to get some frames out, by looping avcodec_decode_video2() until it returns 0, but for some odd reason the frames are all the same frames... [19:07] <iive> ZodiacMentor: try to disable threaded decoding and see if you get the same result. [19:08] <iive> in that case it shuold not delay more than 2 frames. [19:08] <iive> 2 in case of h264 with b-pyramid, and 1 in case of b-frames. [19:10] <ZodiacMentor> how I do disable threaded decoding from within the code? and the codec used is huffyuv with pxfmt bgra, ffplay play's the file perfectly [19:28] <ZodiacMentor> I got it fixed by doing the avcodec_decode_video2() loop like in ffmpeg.c [19:28] <ZodiacMentor> putting that inside the loop {pkt.data += pkt.size; pkt.size -= pkt.size; [19:28] <ZodiacMentor> } [19:28] <ZodiacMentor> now it works as it should [19:30] <Rik316> When should , if ever, CONFIG_MEMALIGN_HACK be defined? [20:07] <iive> Rik316: when the OS doesn't have native memalign() [20:26] <elkng> the latest git of ffmpeg is unstable ? [20:28] <sacarasc> It's probably pretty stable. [22:27] <mystica555> i seem to be having issues making the yadif filter engage properly.. [22:28] <mystica555> hm well thats interesting; it wasnt working before.. [22:29] <mystica555> ah. its how im trying to instantiate it. [22:29] <mystica555> -filter:v yadif ; does not seem to work. [22:29] <mystica555> -vf yadif ; does [22:30] <mystica555> is there something inherently different about those ? [22:31] <mystica555> (enable, if not instantiate...damn english) [22:38] <vbgunz> I ran this about 3 hours ago "ffmpeg -i /x.mkv -vcodec libx264 -acodec ac3 -ab 448k -ss 00:00:00 -t 01:00:00 ~/y.1.mp4" ... on a 9GB mkv movie. It's still going and I have no idea when it's going to finish *but* I feel it won't be today. It might be a little exaggerated but damn, for the love of god or kittens, there has to be a faster way? [22:40] <JEEB> first of all, do you get any asm optimizations used with x264? It should output a list [22:40] <JEEB> SSE, SSE2(Slow), etc. etc. [22:40] <vbgunz> JEEB: I can paste what I see so far, I'll do it to a pastebin, one moment [22:41] <JEEB> I'm just starting out with the check if your x264/ffmpeg compiles seem sane'ish :P [22:42] <vbgunz> hmm, trying to copy what I see but it seems to get reset most likely by the last line [22:43] <mystica555> vbgunz: ctrl-s to pause terminal output [22:43] <mystica555> then copy [22:43] <mystica555> then ctrl-q to continue [22:44] <vbgunz> this is what I see from calling ffmpeg with those options http://dpaste.com/735717/ [22:44] <vbgunz> mystica555: that's a gem. thanks [22:44] <JEEB> ok [22:44] <JEEB> using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [22:44] <JEEB> at least it's built with optimizations, so that part is fine [22:44] <JEEB> a bit older revision of course, but that's not major [22:44] <Mavrik> 200 kb/s? [22:45] <vbgunz> hmm, mystica555, Ctrl-s didn't pause anything. I thought that was going to be awesome :) [22:45] <Mavrik> I've found some very wierd behaviour when you force an encoder so low at such resolution. [22:45] <JEEB> Mavrik, I wonder if it really is doing that [22:45] Action: JEEB checks the libx264 line [22:45] <mystica555> vbgunz: hm; that should have worked with most terminal implementations on *nix. [22:45] <JEEB> rc=crf mbtree=1 crf=23.0 [22:45] <JEEB> yes, it is using libx264 defaults [22:46] <JEEB> not 200kb/s [22:46] <Mavrik> hmm [22:46] <Mavrik> point there [22:46] <vbgunz> thats the last line > frame=78025 fps= 7 q=28.0 size= 1374423kB time=00:54:12.49 bitrate=3461.7kbits/s [22:46] <Mavrik> you're about done then ^^ [22:46] <Mavrik> vbgunz: what's your CPU? [22:47] <JEEB> vbgunz, first of all, let me ask what you're doing? [22:47] <JEEB> what is it that you're trying to do? [22:47] <JEEB> I can see that you probably know that you want AC3 audio [22:47] <vbgunz> JEEB: turning a 9GB mkv into 3 mp4 files for playback on a PS3 [22:47] <JEEB> ok [22:47] <vbgunz> Mavrik: I have an AMD Quad 965 [22:47] <JEEB> depending on how the source you're using is encoded, you wouldn't need to re-encode [22:47] <JEEB> you could just -vcodec copy [22:48] <vbgunz> JEEB: yeah, I tried that, vcodec and acodec and what happens is, I get no audio on the PS3 and the video seems to be playing in slow motion [22:49] <vbgunz> sorry, I meant I tried vcodec copy and acodec ac3 [22:50] <JEEB> try a newer ffmpeg (and update libx264 while you're at it) [22:50] <JEEB> it should work, if the stream plays on the PS3 that's already good enough [22:51] <vbgunz> I'm on Fedora 16 x264 using pretty much default repos and rpm-fusion [22:51] <JEEB> it shouldn't be hard, do you have yasm and git? [22:51] <vbgunz> Yeah, this should definitely run on the ps3, just shocking in a sense I've spent 3 hours so far [22:52] <vbgunz> I have git, not sure what yasm is [22:52] <JEEB> well, you were trying to re-encode the video stream :P [22:53] <JEEB> that takes time [22:54] <vbgunz> I just had no idea it would take so long, I expect for this to all be done in about another 6 hours. I'm happy after spending all night trying to figure this out to get something going but I just didn't know it would take so long to convert [22:55] <JEEB> I have no idea what exactly currently is limiting you, but libx264 does have presets (-preset name-of-x264-preset) if you absolutely need to re-encode :V [22:55] <JEEB> but if the damn stream plays with a PS3, I'd say that a copy of the stream would work fine [22:55] <JEEB> and you'd just re-encode the audio [22:55] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [22:56] <vbgunz> I tried using -vpre ultrafast and a 5 minute slice of a test turned out to be something like 3GB. It should have been called ultracowsize or something :) [22:56] <JEEB> well, d'uh [22:57] <JEEB> the fastest and the slowest presets aren't recommended for general usage [22:57] <JEEB> also I'm not sure if -vpre does the same as -preset [22:57] <JEEB> (the main thing being that it should access libx264's presets, not the ffpreset files) [22:57] <JEEB> basically ultrafast is for speed tests :P (to show that x264 can be fast) [22:58] <vbgunz> -preset is a new option isn't it? because I read about this online but everytime I go to the manual, there is no -preset option [22:58] <JEEB> and placebo is for compression testing [22:58] <JEEB> possible [22:58] <JEEB> it's what should be used now for setting presets [22:58] <JEEB> as long as the option accesses libx264's internal presets, we're fine [22:58] <JEEB> whatever the option is called [22:59] <Mavrik> (it's -preset) [22:59] <JEEB> yes, in current ffmpeg [22:59] <JEEB> that comment was in relation to the fact that the guy's ffmpeg doesn't have it :P [22:59] <JEEB> but yeah, I would first of all check out the current ffmpeg git HEAD [22:59] <JEEB> and try vcodec copy + audio encoding [22:59] <JEEB> and then fall back to re-encoding video with the slowest preset you can take if you need to re-encode it [23:00] <vbgunz> I'll wait and hopefully the beefy miracle will include a much later version of ffmpeg [23:00] <CoJaBo> o cool, theres a channel for it .. [23:00] <CoJaBo> ..Is there anything that can trascode/cut/edit/anything a .rm file on linux? Already tried the obvious (avidemus wont decode the video, mencoder just freezes, and ffmpeg can't seem to produce an output file with audio less than 2 hours out of sync) [23:00] <JEEB> if current git HEAD ffmpeg can't handle it, then you're out of luck for the time being :P [23:01] <JEEB> (it's worth it to make a trac issue about it with a sample if it doesn't work with current git HEAD tho) [23:02] <CoJaBo> hm. How do I get the very latest version then? [23:02] <JEEB> git? [23:02] <JEEB> http://git.videolan.org/?p=ffmpeg.git [23:02] <JEEB> you'll probably need yasm, too [23:03] <JEEB> but that should be it [23:03] <CoJaBo> gah, i hate compiling stuff :/ [23:03] <JEEB> I'm sorry, but multimedia stuff usually goes forward so quickly that you can't rely on distros :P [23:05] <JEEB> and if I could compile ffmpeg, so can you. You can just set --prefix= to ${HOME}/ownapps or something [23:05] <JEEB> or heck, you don't even have to install ffmpeg to try it out after compilation [23:05] <mystica555> hrm got a slightly non-ffmpeg question but it'd ease my use of ffmpeg: anyone know a decent dvd ripping program that is GOP Exact (mplayer -dumpstream is leaving a B-frame with no second reference...) and allows for chapter extraction? [23:06] <mystica555> trying to segment up individual videos for Two Mix: White Reflection The Movie ... [00:00] --- Mon Apr 23 2012

1 0

ffmpeg-devel.log.20120421
by burek 22 Apr '12

22 Apr '12

[01:17] <teratorn> hey, can anyone recommend a video editing library based on ffmpeg? [01:22] <Compn> avidemux works pretty well [01:22] <Compn> if thats what you are looking for [01:23] <teratorn> well, a library [01:23] <teratorn> we want to implement our own GUI frontend [01:23] <Compn> heh [01:24] <teratorn> of course we can use ffmpeg directly, but was hoping for something higher level and geared to editing [01:24] <Compn> most editors use libavcodec and libavformat and make their own programs using those to open and decode files [01:25] <Compn> since those can be built as libs [01:25] <teratorn> I suppose it's relatively trivial to cut a video directly, yeah [01:25] <teratorn> transition effects are another thing [01:25] <Compn> i dont recall any editor libs [01:25] <Compn> there might be one and i just dont know of it [01:25] <Compn> wait around :) [01:25] <teratorn> cool thx anyway :) [01:29] <CIA-17> ffmpeg: 03Roland Scheidegger 07master * ra812b599b5 10ffmpeg/libavcodec/ (h264_cabac.c x86/cabac.h x86/h264_i386.h): (log message trimmed) [01:29] <CIA-17> ffmpeg: h264: assembly version of get_cabac for x86_64 with PIC (v4) [01:29] <CIA-17> ffmpeg: This adds a hand-optimized assembly version for get_cabac much like the [01:29] <CIA-17> ffmpeg: existing one, but it works if the table offsets are RIP-relative. [01:29] <CIA-17> ffmpeg: Compared to the non-RIP-relative version this adds 2 lea instructions [01:29] <CIA-17> ffmpeg: and it needs one extra register. [01:29] <CIA-17> ffmpeg: There is a surprisingly large performance improvement over the c version (more [02:22] <CIA-17> ffmpeg: 03Andrew Wason 07master * re9b1d5ae5e 10ffmpeg/libswresample/swresample.c: (log message trimmed) [02:22] <CIA-17> ffmpeg: fix swr_convert buffering of packed audio [02:22] <CIA-17> ffmpeg: swr_convert is not properly buffering packed input audio when the [02:22] <CIA-17> ffmpeg: output is not large enough, and when resampling is not actually needed [02:22] <CIA-17> ffmpeg: (same samplerate and no SWR_FLAG_RESAMPLE). [02:22] <CIA-17> ffmpeg: buf_set() is only handling the first channel and leaving the others as-is. [02:22] <CIA-17> ffmpeg: Sample program to reproduce the problem is here https://gist.github.com/2431768 [02:22] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r9849515214 10ffmpeg/libavcodec/ (h264_cabac.c x86/cabac.h x86/h264_i386.h): [02:22] <CIA-17> ffmpeg: Revert "h264: assembly version of get_cabac for x86_64 with PIC (v4)" [02:22] <CIA-17> ffmpeg: This broke compilation on darwin, revert until a better solution is found. [02:22] <CIA-17> ffmpeg: This reverts commit a812b599b504b39a8021827da89d5e23fb361cc9. [02:38] <funman> i thought only darwin/x86 was problematic, not x86_64 [02:42] <iive> funman: the commit in questing is 1 hour old, you can see the description above. [04:25] <funman> ah it changed x86 too [10:16] <CIA-17> ffmpeg: 03Tim Nicholson 07master * rc97201dd29 10ffmpeg/ (3 files in 2 dirs): [10:16] <CIA-17> ffmpeg: lavfi/setfield: add "progressive" option [10:16] <CIA-17> ffmpeg: Add "prog" parameter value, and deprecate numeric values. [10:16] <CIA-17> ffmpeg: Signed-off-by: Stefano Sabatini <stefasab(a)gmail.com> [10:32] <northrup> anyone awake in the developers chat? [12:33] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * r459414b4d9 10ffmpeg/libavfilter/vf_setfield.c: [12:33] <CIA-17> ffmpeg: lavfi/setfield: rename variable top_field_first to mode [12:33] <CIA-17> ffmpeg: The new name better reflects the meaning of the variable, after the [12:33] <CIA-17> ffmpeg: recent change. [12:33] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * rdd3043d1c0 10ffmpeg/libavfilter/vf_setfield.c: [12:33] <CIA-17> ffmpeg: lavfi/setfield: remove stray argument in av_log() [12:33] <CIA-17> ffmpeg: Fix warning: [12:33] <CIA-17> ffmpeg: libavfilter/vf_setfield.c: In function init: [12:33] <CIA-17> ffmpeg: libavfilter/vf_setfield.c:64:20: warning: too many arguments for format [-Wformat-extra-args] [14:04] <CIA-17> ffmpeg: 03Nicolas George 07master * rb27383e453 10ffmpeg/ffprobe.c: [14:04] <CIA-17> ffmpeg: ffprobe: do not try to decode empty packets. [14:04] <CIA-17> ffmpeg: Fixes sporadic decode failures and trac ticket #997. [17:29] <ods15> any way to tell gcc that "(const int[]){1,2,3}" should go in text segment and not in data?... if I put it on a seperate line it works, but not within an expression :/ [17:32] <michaelni> ods15, probably best to ask gcc devels [17:34] <ods15> it's alright, I gave up, two lines it is... means I need to declare all the arrays first, and then I intialize the big array which uses them all [17:35] <ods15> yey, now this file has zero data/bss size [19:55] <CIA-17> ffmpeg: 03Jakub Stachowski 07master * r93bf1aac6a 10ffmpeg/libavcodec/wmalosslessdec.c: [19:55] <CIA-17> ffmpeg: wmalossless: Ensure that last frame is not written again if nothing was decoded in current packet. [19:55] <CIA-17> ffmpeg: Reviewed-by: Mashiat Sarker Shakkhar <mashiat.sarker(a)gmail.com> [19:55] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:55] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r5a35bd92ad 10ffmpeg/libavcodec/cook.c: [19:55] <CIA-17> ffmpeg: cook: check subacket count [19:55] <CIA-17> ffmpeg: Fixes out of array writes. [19:55] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [19:55] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:55] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r13381577d1 10ffmpeg/libavformat/xmv.c: [19:55] <CIA-17> ffmpeg: xmvdemux: dont let current_stream become invalid. [19:55] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [19:55] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:55] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r80d51680b2 10ffmpeg/libavformat/xmv.c: [19:55] <CIA-17> ffmpeg: xmvdemux: prefer av_freep() to deallocate audio stream structs. [19:55] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:57] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8c59e0c362 10ffmpeg/libavcodec/xl.c: [21:57] <CIA-17> ffmpeg: xldec: move buffer size check up, it can be done before allocating a frame [21:57] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:58] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r94b42da696 10ffmpeg/libavcodec/xl.c: [21:58] <CIA-17> ffmpeg: xldec: Check that width is a multiple of 4 [21:58] <CIA-17> ffmpeg: Fixes out of array reads [21:58] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [21:58] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:58] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r6c0027bb39 10ffmpeg/libavcodec/dnxhddec.c: [21:58] <CIA-17> ffmpeg: dnxhddec: check that the indicated bit depth matches the tables. [21:58] <CIA-17> ffmpeg: Fixes crash [21:58] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [21:58] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:33] <CIA-17> ffmpeg: 03Carl Eugen Hoyos 07master * r2f06b56382 10ffmpeg/libavcodec/ (v210dec.c v210dec.h): [22:33] <CIA-17> ffmpeg: Support broken v210 files with 64 byte padding. [22:33] <CIA-17> ffmpeg: Fixes ticket #743. [22:33] <CIA-17> ffmpeg: Reviewed-by: Paul B Mahol [23:00] <CIA-17> ffmpeg: 03Ronald S. Bultje 07release/0.10 * r265a628f16 10ffmpeg/libavcodec/x86/cabac.h: [23:00] <CIA-17> ffmpeg: h264: use struct offsets in get_cabac_bypass_sign_x86(). [23:00] <CIA-17> ffmpeg: (cherry picked from commit db025929f202bc32459a1278ee06920a06564762) [23:24] <CIA-17> ffmpeg: 03Diego Biurrun 07master * rb1563d0cf9 10ffmpeg/libavcodec/ivi_common.c: [23:24] <CIA-17> ffmpeg: ivi_common: Initialize a variable at declaration in ff_ivi_decode_blocks(). [23:24] <CIA-17> ffmpeg: This simplifies the code a bit and avoids an uninitialized variable warning. [23:24] <CIA-17> ffmpeg: 03Diego Biurrun 07master * rf973a85d4a 10ffmpeg/libavcodec/xxan.c: [23:24] <CIA-17> ffmpeg: xxan: Remove write-only variable in xan_decode_frame_type0(). [23:24] <CIA-17> ffmpeg: libavcodec/xxan.c:293:13: warning: variable corr_end set but not used [23:24] <CIA-17> ffmpeg: 03Anton Khirnov 07master * r2994913d70 10ffmpeg/avconv.c: [23:24] <CIA-17> ffmpeg: avconv: fix a segfault when default encoder for a format doesn't exist. [23:24] <CIA-17> ffmpeg: Fail earlier and with a more descriptive error message. [23:25] <CIA-17> ffmpeg: 03Alex Converse 07master * r7eacd70fea 10ffmpeg/libavcodec/aacdec.c: aac: Handle HE-AACv2 when sniffing a channel order. [23:25] <CIA-17> ffmpeg: 03Jan Ekström 07master * rb5c3f0b994 10ffmpeg/libavcodec/utvideo.c: [23:25] <CIA-17> ffmpeg: utvideo: general cosmetics [23:25] <CIA-17> ffmpeg: General cosmetics, such as keeping lines under 80 characters, [23:25] <CIA-17> ffmpeg: fixing a couple of typos (predition -> prediction) and a [23:25] <CIA-17> ffmpeg: general style fix that was pointed out by Derek when I was having [23:25] <CIA-17> ffmpeg: my sliced multithreading patch in review by him. [23:25] <CIA-17> ffmpeg: Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [23:25] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rc047afb80c 10ffmpeg/: (log message trimmed) [23:25] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/master' [23:25] <CIA-17> ffmpeg: * qatar/master: [23:25] <CIA-17> ffmpeg: avcodec: remove AVCodecContext.dsp_mask [23:25] <CIA-17> ffmpeg: avconv: fix a segfault when default encoder for a format doesn't exist. [23:25] <CIA-17> ffmpeg: utvideo: general cosmetics [23:25] <CIA-17> ffmpeg: aac: Handle HE-AACv2 when sniffing a channel order. [23:25] <CIA-17> ffmpeg: 03Mans Rullgard 07master * r95510be8c3 10ffmpeg/libavcodec/ (avcodec.h pthread.c version.h x86/dsputil_mmx.c): [23:25] <CIA-17> ffmpeg: avcodec: remove AVCodecContext.dsp_mask [23:25] <CIA-17> ffmpeg: it for eviction at the next version bump. It has been superseded [23:25] <CIA-17> ffmpeg: by av_set_cpu_flag_mask() which, unlike this field, works everywhere. [23:25] <CIA-17> ffmpeg: Signed-off-by: Mans Rullgard <mans(a)mansr.com> [23:25] <CIA-17> ffmpeg: 03Alex Converse 07master * rdc878b96a7 10ffmpeg/libavformat/movenc.c: [23:25] <CIA-17> ffmpeg: movenc: Support high sample rates in isomedia formats by setting the sample rate field in stsd to 0. [23:25] <CIA-17> ffmpeg: Libisomediafile appears to always set this field to zero. [00:00] --- Sun Apr 22 2012

1 0

ffmpeg.log.20120421
by burek 22 Apr '12

22 Apr '12

[00:03] <iive> DelphiWorld: decoder - yes. there is an encoder, but you don't want to use it :) [00:04] <DelphiWorld> iive: LOL :) [00:04] <DelphiWorld> iive: a bit pissing question:P please suggest a low biterate audio / video codec that i can use with ffmpeg, by default [00:06] <iive> lower than mp3? not sure there is one. mp2 and ac3 use higher bitrate. [00:07] <DelphiWorld> iive: ok, higher audio ? good one? ;) [00:10] <stf> is there any limitation for copy map streams? [00:11] <barilo> how do I convert some images into a video? [00:13] <Mavrik> DelphiWorld: I usually use AAC+ with libaacplus when I have to encode really low bitrate audio [00:13] <Mavrik> just make sure your playback devices support it [00:13] <DelphiWorld> Mavrik: vlc [00:14] <DelphiWorld> Mavrik: please may i pm you? [00:19] <stf> hm i think i have found the problem: ffmpeg do not like to copy more then one Audio stream& [00:23] <bahar> what will it take for ffmpeg to support OpenCL or CUDA? [00:30] <DelphiWorld> good night folks [00:31] <JEEB> bahar, it took a company with close ties to a GPU maker and an actual development team, as well as months' worth of time to develop something useful'ish for x264 (not ready yet) [00:31] <bahar> what about VPDAU? i take it you can compile ffmpeg w/ support for VPDAU? that will offload some work to the GPU? [00:31] <bahar> VDPAU* [00:34] <JEEB> VDPAU/CUVID are APIs to use nvidia's dedicated hardware on their GPUs to decode stuff, yeah. I don't know if ffmpeg supports them, though. [00:34] <bahar> http://forums.fedoraforum.org/showthread.php?t=260198 [00:34] <bahar> apparently some guy compiled ffmpeg with VDPAU support [00:34] <JEEB> (the GPU has a decoding ASIC on it, and you just feed stuff to it and get decoded pictures back) [01:15] <district> hi. is there something like avformat_open_input() that works with an already opened file descriptor? [01:17] <teratorn> hey, can anyone recommend a video editing library based on ffmpeg? [02:23] <freeminds> I have the feeling that the -t option is pretty inaccurate is there a better option? [03:53] <taqattack> I'm having some issues with piping output in flv into another ffmpeg process. Does anyone know how to do this? [04:11] <connelly> could someone help me out with some trouble I'm having with ffserver? It appears to keep taking info from ffmpeg until it's feed file fills up and then just hangs [07:31] <taqattack> I have a question. If I have a software call FFmpeg through CLI that has built in mp3 library and x264 library, do I need to release my source code. I don't intend to sell it though. [07:41] <mystica555> taqattack: do you link to libffmpeg? or simply call it via commandline? [07:41] <mystica555> if entirely through commandline, you are simply utilizing a tool in userspace; this needs no licensing provision at all imho (i am not a lawyer, but im pretty sure its legally sound) [07:41] <mystica555> if you linked to it, then you are making a derivative so then the licensing comes into play [07:42] <mystica555> taqattack: its also my understanding (again, i am not a lawyer, i could be wrong on this) but if you did use it for internal use, and not distribute it, you technically shouldnt have to either. [07:43] <mystica555> but, as soon as you distribute modifications, then again the licensing comes into play [08:33] <spyzer> hey everyone, i converetd a video from mov to mp4 from ipod using this command [08:33] <spyzer> fmpeg -i IMG_0146.MOV -vf "hflip,vflip" my.mp4 [08:33] <spyzer> but the quality of video has degraded [08:34] <spyzer> Input is this -> Stream #0.0(und): Video: h264 (Baseline), yuv420p, 640x480, 3526 kb/s, 30.02 fps, 30 tbr, 600 tbn, 1200 tbc [08:34] <taqattack> mystica555: Thanks. I did compile it using GPL with external libraries. And I guess I will be fine since I am calling it via CLI only. [08:34] <spyzer> Output is this -> Stream #0.0(und): Video: mpeg4, yuv420p, 640x480, q=2-31, 200 kb/s, 30 tbn, 30 tbc [08:34] <spyzer> I do not understand these numbers actually [08:35] <Mavrik> spyzer, well, you didn't tell the encoder anything about what quality you want [08:35] <spyzer> So may anyone please help me to raise up the video quality [08:35] <spyzer> okay so by default what doe sit do, Mavrik? [08:35] <Mavrik> as you see, set bitrate to 200 kb/s :) [08:35] <mystica555> taqattack: indeed, it should be fine if its just calling from commandline [08:35] <spyzer> okay so use -b option? [08:35] <Mavrik> spyzer, mhm [08:35] <Mavrik> spyzer, see this: http://www.virag.si/2012/01/web-video-encoding-tutorial-with-ffmpeg-0-9/ [08:35] <spyzer> thanks :) [08:36] <Mavrik> spyzer, also encoding with libx264 instead of mpeg4 would probably help [08:36] <spyzer> Mavrik: does it matter that i specify the bitrate in powers of 2? [08:36] <Mavrik> nop, bitrate can be anything [08:36] <Mavrik> higher the number, the bigger the file [08:37] <spyzer> ahh i see [08:37] <taqattack> mystica555: What if I don't show the commandline interface? As in I'm using my software as a GUI for FFmpeg backend. [08:37] <mystica555> again its all about linking [08:37] <mystica555> if you yourself are simply calling ffmpeg via a commandline, its not license related imho [08:37] <mystica555> if you link to libffmpeg, then it becomes a license issue [08:38] <mystica555> just like k3b calls cdrecord, mkisofs, etc [08:38] <mystica555> it doesnt link to any of them [08:38] <mystica555> just calls them and pipes data [08:39] <taqattack> Yeah that's what I'm trying to do essentially. But what about libmp3 and libx264? [08:39] <taqattack> Since ffmpeg calls them internally. I dont have to worry about those too, right? [08:40] <Mavrik> why would you? You're calling ffmpeg binary, not linking to them [08:40] <mystica555> exactly [08:40] <mystica555> its only if your program itself links [08:40] <Mavrik> ffmpeg has those issues cleared (when you compile ffmpeg you see under which license it is license... and that's dependent on what you link to) [08:41] <taqattack> Since it was external libraries and not included in ffmpeg. I assumed it was kind of like linking ot to the databases [08:41] <taqattack> But yeah I understand now. [08:41] <mystica555> nah, you are calling a program; that program (ffmpeg) can be a black box to you if you only execute it and not link to a shared/static library called from within your pogram [08:42] <mystica555> *program [08:42] <taqattack> So why don't all these commercial applications make use of Ffmpeg then? [08:42] <Mavrik> which commercial applications? :) [08:42] <Mavrik> alot of apps use ffmpeg internally [08:43] <Mavrik> heck we used it at my IPTV provider to transcode video channels on the fly :) [08:43] <taqattack> Interesting. [08:44] <taqattack> Okay. Well that clears up a lot of things. Thanks again. [08:45] <mystica555> indeed, ffmpeg gets used a *lot* of places [08:45] <mystica555> look at youtube [08:46] <mystica555> potentially the largest ffmpeg user ever [08:51] <djdduty> hello! [08:52] <djdduty> I am trying to run the command ffmpeg -f image2 -r 25 -i *.jpg -vcodec libvpx -vpre 1080p -ar 22050 -crf 22 video.mp4 to compile a bunch of pictures into a timelapse, but I get the error "Could not write header for output file #796 (incorrect codec parameters ?) and I have 796 photos that will compile D: [08:56] <djdduty> hmm, I tried a different command but vlc won't display the video [08:56] <djdduty> :( [08:57] <djdduty> well I think it is playing, but it is only 0.0001 sec long D: [09:00] <djdduty> hmmm [09:00] <djdduty> :( [09:10] <taqattack> I'm having some issues with ffmpeg streaming to RTMP. I'm getting low framerates when uploading to the server. http://pastebin.com/gVw3EdZ1 [09:12] <taqattack> I mean I've tried to save it to harddrive which works flawlessly. [09:12] <djdduty> so, ffmpeg -f image2 -r 20 -i *.jpg -sameq -crf 20 video.mp4 is not working for me to make a bunch of images taken 10 seconds apart into a video at 20 fps :( [09:13] <taqattack> I think i might have figured it out [09:14] <taqattack> when you put 20 fps you're displaying 20 pictures in 1 second [09:14] <djdduty> taqattack: oh, sorry let me look [09:14] <djdduty> I have a streaming script that works great, allthough it is EXTREMELY cpu intensive [09:14] <djdduty> taqattack: yes I know [09:14] <djdduty> taqattack: I have 7980 pictures [09:16] <djdduty> taqattack: what is your script? [09:16] <taqattack> I'm calling FFmpeg [09:16] <taqattack> with [09:16] <taqattack> -y -report -re -i ffmeg.flv -vcodec copy -threads 0 -acodec copy -f flv rtmp://live.justin.tv/app/live_20700021_hqykFR8HfNfH0tIqK4faPu73mk2req [09:17] <taqattack> That's my second thread of ffmpeg [09:17] <djdduty> you just gave out your live key -.- [09:17] <taqattack> It's a test livekey [09:17] <djdduty> I see, mine is ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0+1200,896 -ab 96k \ [09:17] <djdduty> -f alsa -ac 2 -i pulse -vcodec libx264 -crf 30 -preset "$QUAL" -s "1280x720" \ [09:17] <djdduty> -acodec libmp3lame -ar 44100 -threads 0 \ [09:17] <djdduty> -f flv "$URL" [09:17] <djdduty> my script that is [09:18] <djdduty> URL is my rtmp://live.justin.tv... [09:18] <djdduty> INRES is my monitor res [09:18] <djdduty> OUTRES is my target res to stream as [09:18] <djdduty> FPS is my target fpsQUAL is simply "fast" right now [09:18] <taqattack> I have something similar [09:19] <djdduty> it is _very_ slow though [09:19] <djdduty> well not slow [09:19] <djdduty> it runs fine [09:19] <djdduty> but 40% cpu usage on all threads D: [09:19] <djdduty> just from it [09:19] <djdduty> I don't know how to fix that [09:19] <taqattack> Does it stream well? [09:19] <djdduty> it actually causes my temperature threshold to trigger, freezing my machine and shutting down [09:19] <taqattack> is librtmp compiled? [09:20] <djdduty> I'd say it works good... www.justin.tv/djdduty [09:21] <taqattack> u dont set the vbitrate or maxrate or bufsize? [09:21] <djdduty> =/ [09:22] <djdduty> taqattack: I don't it appears D: [09:22] <djdduty> taqattack: do you think that will help? :D [09:24] <taqattack> This is mine: ffmpeg -r 30 -f dshow -i video="Dxtory Video 1":audio="Stereo Mix (VIA High Definition" -vcodec libx264 -preset ultrafast -vb 384k -maxrate 500k -bufsize 384k -acodec libmp3lame -ab 128k -f flv rtmp://live.justin.tv/app/ [09:24] Action: djdduty uses linux [09:24] <djdduty> some of those wouldn't work :) but I see what you mean [09:25] <djdduty> on windows, why not just use xsplit? [09:25] <taqattack> the quality is not so good [09:25] <taqattack> for the free version [09:25] <taqattack> Especially audio is limited to 42k bitrate [09:26] <taqattack> Plus it is a huge resource hog on the machine. [09:26] <djdduty> taqattack: it used less on my windows that ffmpeg uses on my linux :D [09:26] <djdduty> and they have the same exact hardware [09:27] <taqattack> I havent tried it recently though. [09:28] <taqattack> So you could be right. But the limited audio quality is still an issue for me and I needed a freeware alternative. [09:31] <djdduty> maybe someone could enlighten me as to why livestreaming takes so much cpu? [09:33] <taqattack> Real-time encoding would be my guess. [09:34] <taqattack> I'm reading over the chat logs. I'm not sure what you're trying to do exactly. [09:34] <djdduty> hmmm [09:34] <djdduty> =/ [09:34] <djdduty> but I mean [09:34] <djdduty> you will have to look at the htop stream I am doing right now [09:34] <djdduty> xD [09:34] <djdduty> www.justin.tv/djdduty [09:35] <djdduty> stopped it [09:35] <taqattack> Interesting [09:37] <taqattack> It seems like you're trying to output image2 sequence [09:37] <taqattack> ffmpeg -r 20 -i *.jpg -sameq -crf 20 -f mp4 video.mp4 [09:37] <taqattack> Try that maybe [09:38] <djdduty> ok, turn your audio down, thhis is all screwed up! http://www.justin.tv/djdduty/b/315665819 [09:38] <djdduty> I have no idea what is wrong with the audio [09:38] <djdduty> and no idea why it uses all 4 cores/threads at 40% [09:40] <taqattack> That's some insane CPU usage [09:40] <djdduty> yeah D: [09:40] <djdduty> hmm [09:40] <djdduty> I did not write this script [09:40] Action: djdduty is ffmpeg noob [09:41] <djdduty> how do I remove audio from streaming? [09:41] <djdduty> http://www.justin.tv/djdduty/b/315665819 [09:41] <djdduty> oops [09:41] <djdduty> wrong thing [09:41] <djdduty> ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0+1200,896 -ab 96k \ [09:41] <djdduty> -f alsa -ac 2 -i pulse -vcodec libx264 -crf 30 -preset "$QUAL" -s "1280x720" \ [09:41] <djdduty> -acodec libmp3lame -ar 44100 -threads 0 \ [09:41] <djdduty> -f flv "$URL" [09:41] <taqattack> Try setting preset to veryfast [09:41] <djdduty> like this? ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0+1200,896 -vcodec libx264 -crf 30 -preset "$QUAL" -s "1280x720" \ [09:41] <djdduty> -acodec libmp3lame -ar 44100 -threads 0 \ [09:41] <djdduty> -f flv "$URL" [09:42] <djdduty> oops [09:42] <djdduty> no not that [09:42] <djdduty> I left in the avode and -ar [09:42] <taqattack> Try [09:43] <taqattack> I think u might need to set your output fps [09:44] <djdduty> taqattack: wtf? veryfast brings everything up to the 58% [09:45] <taqattack> I get 61% usage to 51% going from fast to veryfast [09:45] <djdduty> =/ [09:45] <taqattack> Try limiting your output fps [09:45] <djdduty> also, ffmpeg -r 20 -i *.jpg -sameq -crf 20 video.mp4 is my new timelapse compile script, havnt tested it [09:45] <taqattack> That seems about right [09:46] <djdduty> same problem on timelapse video [09:46] <taqattack> output header? [09:46] <taqattack> or 1 second video [09:46] <djdduty> it is only like 0.001 sec long or something :( [09:47] <djdduty> how do I add out fps? [09:47] <taqattack> ffmpeg -r 20 -i *.jpg -sameq -crf 20 -r 20 -f mp4 video.mp4 [09:48] <taqattack> Are your files numbered? [09:48] <djdduty> es [09:48] <djdduty> with a timestamp [09:50] <djdduty> didnt fix it [09:50] <djdduty> I just realized [09:50] <djdduty> how does it know which image to use first? D: [09:50] <taqattack> Yes exactly [09:50] <djdduty> my images are names like this 20120420132639.jpg [09:50] <taqattack> can u give me example of the naming convention? [09:51] <djdduty> 20120420110752.jpg is the first image [09:51] <taqattack> what would the second image in the sequence be named [09:51] <djdduty> 20120420174911.jpg is the last [09:51] <djdduty> second is [09:52] <taqattack> ....53.jpg? [09:52] <djdduty> 20120420110780.jpg [09:52] <taqattack> I think u might want to rename them for ffmpeg to read them correctly [09:52] <djdduty> sorry wait [09:52] <djdduty> the first one is messed up [09:52] <djdduty> all the ones after are incrememnt by 1 [09:53] <taqattack> Can you run this [09:53] <taqattack> ffmpeg -re -i video.mp4 -vcodec ljpeg -f image2 "output_%05d.jpg" [09:54] <taqattack> It should extract out all the jpg used in video.mp4 [09:54] <djdduty> yes [09:54] <taqattack> Were u able to extract all the images? [09:55] <djdduty> it gave me output_00001.jpg [09:55] <taqattack> and nothing else? [09:55] <djdduty> nope [09:55] <taqattack> Yeah so ffmpeg is only reading 1 jpg input [09:55] <taqattack> and outputing that frame in mp4 [09:57] <taqattack> You might have to rename the files [09:58] <taqattack> so that the first image in the sequence would start from 0 [09:58] <taqattack> http://ffmpeg.org/trac/ffmpeg/ticket/247 [09:59] <djdduty> taqattack: so, my script to take images every 10 seconds is as follows... [09:59] <djdduty> while [ 1 ]; do scrot -q 100 $(date +%Y%m%d%H%M%S).jpg; echo "screen taken"; sleep 10; done [10:04] <taqattack> I'm just not sure if ffmpeg will be able to recognize that sequence. [10:08] <taqattack> You might need to change 20120420132639.jpg to something like input_00001.jpg, so on and so forth... [10:08] <taqattack> 20120420110752.jpg to input_00001.jpg, 20120420110753.jpg to input_00002.jpg, etc. [10:09] <taqattack> There might be a script to do this. [10:10] <taqattack> But as an experiment, you could try with 2 files. [10:11] <taqattack> and then try to run: ffmpeg -r 20 -i "input_%05d.jpg" -sameq -crf 20 -r 20 -f mp4 video.mp4 [10:11] <taqattack> TO see if it works. [10:12] <taqattack> Anyways, its 4Am in the morning right now. I must go to bed. I hope you resolve this issue. [10:13] <djdduty> you can do this in script, just have a varialbe initialize as 00001 and increment it by 1, name the picture frame_VARIABLE [10:22] <djdduty> taqattack: k, now I have them named frame_(incremeting variable) [10:22] <djdduty> so frame_1.jpg [10:22] <djdduty> frame_2.jpg [10:22] <djdduty> etc [10:23] <djdduty> taqattack: and it still doesn't work [10:23] <djdduty> with ffmpeg -r 20 -i *.jpg -sameq -crf 20 -r 20 -f mp4 video.mp4 [10:28] <djdduty> wait! [10:28] <djdduty> I got it too work! [10:28] <djdduty> \o/ [10:28] <djdduty> changed it too frame_%01d.jpg :) [10:29] <djdduty> but I don't know if it will grab the double digit ones :( [10:31] <northrup> anyone have any experience of getting ffmpeg to compile with CrystalHD support? [10:33] <djdduty> taqattack: nvm it doesnt work [10:33] <djdduty> it makes a video, of seemingly appropriate length, but of only the first screenshot D: [10:43] <djdduty> RRAGGHHH wtf?! [10:44] <djdduty> why is it making the video, but only with the very first frame spanned out the entire thing? O.o [10:46] <djdduty> there we go! [10:46] <djdduty> yay! [10:46] <djdduty> \o/ [13:49] <burek> <djdduty> but I don't know if it will grab the double digit ones :( [13:49] <burek> <djdduty> changed it too frame_%01d.jpg :) <---- it won't [13:49] <burek> you specified that your images have 1 digit after the text "frame_" and before ".jpg" [13:50] <burek> if you want double digits, like 01, 02, ... 10, 11, 12, ... use %02d [14:06] <kami8785> i upgraded from .7 to .10 and now if i do something like rtmpdump | ffmpeg, it waits for rtmpdump to finish before starting to decode [14:07] <kami8785> help would be much appreciated [14:12] <Kakadu> can you give me a tip what options should I add? http://paste.in.ua/4191/#35 [14:18] <Kakadu> it seems I have wrong presets... [14:19] <kami8785> any way i can force ffmpeg to start decoding right away instead of waiting? [14:23] <Mavrik> Kakadu: you have obsolete command line for new ffmpeg versions [14:24] <Mavrik> http://www.virag.si/2012/01/web-video-encoding-tutorial-with-ffmpeg-0-9/ [14:24] <Mavrik> kami8785: what do you mean by "waiting"? [14:25] <Kakadu> Mavrik: I've looked at http://michaelchinen.com/2011/06/27/justin-tv-streaming-in-linux/ but now I'll investigate your link, Thanks. [14:25] <Mavrik> Kakadu: new ffmpeg versions (in combination with libx264) now have a slightly different command-line parameters [14:25] <kami8785> i used to use .7 and using this command "rtmpdump | ffmpeg -i - -r 1 -sameq -f image2 /path/%06d.jpg" it would instantly start decoding [14:25] <Mavrik> there are no preset files anymore, the presets are now built-in [14:26] <kami8785> now it takes >60 seconds to start [14:26] <kami8785> presumably buffering the input [14:26] <Mavrik> kami8785: ugh, using sameq probably doesn't do what you think it does :P [14:26] <Mavrik> kami8785: yeah, setting analze duration or explicitly setting input formats will cut on that time [14:27] <kami8785> sorry i dont understand, how can i set that stuff? [14:27] <kami8785> id love to get back to having instant decoding like w/ the old version [14:28] <kami8785> since my app relies on it [14:29] <Mavrik> if you know your input format you pass "codec:v" and "codec:a" before the -i [14:29] <Mavrik> or you check in documentation how to set analyze duration [14:31] <kami8785> yeah ok analyzeduration -0 works [14:31] <kami8785> ^_^ [14:31] <kami8785> thanks Mavrik [14:32] <Mavrik> that MIGHT not detect all your streams ;) [14:33] <kami8785> k i will keep that in mind if i encounter problems [15:34] <mystica555> hm.. where would i start trying to diagnose a sigfpe in libavfilter [15:35] <Mavrik> build it with debug filters, run with gdb [15:35] <Mavrik> *debug parameters [15:41] <mystica555> heh; seems the better option would be 'rebuild form git, maybe it'll work the next time' ;) [15:41] <mystica555> as in, its not crashing anymore [15:42] <mystica555> *from [15:43] <mystica555> how much overhead (if any appreciable amount) is caused by the debug symbols included? [15:43] <mystica555> ie, how much slowdown? [15:43] <Mavrik> binaries are bigger, not to mention you have to build without optimizations to have any use of debugging [15:43] <Mavrik> so yeah, very noticable slowdown [15:44] <mystica555> hm didnt realize the lack of optimizations was part of it [15:44] Action: mystica555 recompiles without debug [15:51] <Kakadu> I'm a little bit confused by strange error message. http://paste.in.ua/4192/#21 [15:58] <mystica555> Kakadu: itd seem that the jtv script is using randomly wrong commandline options for your version of ffmpeg. [15:59] <mystica555> your version > newer versions (likely; as ffmpeg keeps changing things) [16:00] <Kakadu> hmmm, some minute ago sound was OK.... [16:00] <Kakadu> mystica555: how to find solution? [16:02] <mystica555> not certain; if you are saying that exact script worked not long ago, then i don't know what would have changed. [16:05] <Kakadu> not the exact I was playing with video settings but now I can't revert it [16:05] <Kakadu> Why ffmpeg's error message is so useless? [16:06] <mystica555> you'd probably want to diff the original script for your new copy [16:06] <mystica555> as for useless error messages... im getting some annoying ones with libx264 [16:08] <mystica555> [libx264 @ 0x2530c50] bad option 'preset': 'slow' [16:08] <mystica555> to me, that seems like a perfectly valid option based on x264 --longhelp [16:09] <mystica555> passing it as -x264opts preset=slow [16:13] <mystica555> i bet my libx264 is too old... [16:13] <mystica555> the difference 4 months makes >.> [16:29] <theophilus> Hey guys... I want to use libavfilters to do a watermark logo and overlay text on some videos... I'm having trouble figuring out the documentation for drawtext and using it along side the movie option for the watermark logo... any chance someone could give me a sample command that uses both of these, so I could have something to go off of? The documentation is great and everything, but if I could see an example of it, that would be [16:29] <theophilus> golden! [16:33] <mystica555> [libx264 @ 0xb09c50] bad option 'preset': 'slower' [16:33] <mystica555> any ideas? [16:33] <mystica555> -x264opts crf=18:preset=slower [16:35] <Mavrik> why aren't you just using "-crf 18 -preset slower" ? [16:38] <mystica555> at least one point during the reading of the ffmpeg manual i came across options that were deprecated so i was potentially under the wrong assumption that others would be to; such as, level [16:40] <mystica555> granted ive reinstalled ffmpeg today, so if a newer manual appeared within the last few weeks, that changed the wording, it no longer shows it for anything but level.. [16:41] <mystica555> also, the crf= option only seems to appear in libvpx; and -x264opts crf=18 worked fine [16:41] <mystica555> rather the -crf option [16:41] <mystica555> ... [16:42] <mystica555> not sure how much of that you got.. [16:43] <mystica555> i think it was mainly strangely deprecated and non-deprecated things in the manual confusing me. [16:43] <mystica555> and, -crf seems to only be for libvpx as -x264opts crf=18 works fine [16:43] <mystica555> *shrug* [16:49] <theophilus> (bump) Anybody think they can help with some libavfilter stuff? [17:08] <burek> <Kakadu> Why ffmpeg's error message is so useless? <--- because you don't rtfm :) [17:08] <burek> mystica555 [17:08] <burek> can you please use pastebin.com, to show your command line and its output? [17:09] <mystica555> afaik the relevant info was -x264opts crf=18:preset=slow causes '[libx264 @ 0x2530c50] bad option 'preset': 'slow'' [17:09] <mystica555> which is bypassed with -preset slow [17:10] <mystica555> (taking out :preset=slow in x264opts) [17:10] <burek> ok, then, help yourself :) [17:10] <mystica555> i apparently did :) [17:11] <mystica555> at one point i believe the manual stated that certain options were deprecated; then i compiled a new version today and it may have changed; or im just reading it wrong [17:11] <mystica555> ie, i could have sworn all of 'level' 'preset' and 'profile' were 'deprecated, in favor of x264opts' [17:11] <mystica555> now i only see 'level' as being on that list [17:12] <mystica555> then again my brain is holier than a piece of swiss cheese so i could be entirely wrong [17:12] <mystica555> but, its encodin g;) [17:12] <mystica555> *encoding [17:13] <mystica555> http://paste.pocoo.org/show/585234/ [17:14] <mystica555> if you're entirely curious at what i got working [17:14] <mystica555> (ive since tweaked the crf for quality) [18:28] <luc4_mac> Hi! I'm trying to use the ffmpeg command to simply rotate a video, so I'm using ffmpeg -i file1.avi -vf "transpose=1" file2.avi. What I see is that, whatever I do, quality is not kept intact, but it decreases. Can I simply keep everything unaltered (codec, container, etc&)? [19:00] <burek> luc4_mac, it depends on the encoder used [19:00] <burek> can you please use pastebin.com, to show your command line and its output? [19:04] <luc4_mac> burek: you mean that ffmpeg encoder might not support the original encoding? I'm trying to use h264 now as output codec, give me a second to pastebin the original output. [19:06] <burek> no, I mean, for example acdsee32 (image program for windows) could rotate jpg images without loosing the quality [19:07] <burek> since the encoding process was such that rotation of the image can be done without reencoding [19:07] <burek> but only if you rotate for 90, 180, 270 degrees [19:07] <burek> it is highly unlikely you will be able to do the same with video stream [19:08] <burek> so I suppose you'll have to reencode it anyway [19:08] <burek> can you control the player, to instruct it to rotate the video? [19:09] <luc4_mac> burek: yes, I understand this. Yes, I can rotate with vlc for instance using post processing. [19:09] <luc4_mac> burek: this is what I tried first: http://pastebin.com/xus6FG1w. [19:09] <luc4_mac> burek: I was just supposing that this should just rotate and reencode using the same exact parameters of the source video. [19:10] <burek> no, it just uses default encoder [19:10] <burek> with default settings, so it is more likely that it will be something you don't want :) [19:11] <burek> if you want h264 as an output [19:11] <burek> try this [19:11] <luc4_mac> burek: oh now I get. Yes, h264 has just finished and seems correct. [19:11] <burek> ffmpeg -i DSCN0214.AVI -acodec copy -vcodec libx264 -crf 20 -vf "transpose=1" DSCN0214_transposed.mp4 [19:13] <luc4_mac> burek: the only param I was missing is cry :-) thanks! [19:13] <luc4_mac> cry = crf :-) [19:13] <burek> :beer: :))) [19:14] <luc4_mac> burek: should I enable cry? [19:14] <luc4_mac> burek: autocorrection& cry = crf :-) [19:18] <burek> luc4_mac, crf is constant rate factor [19:18] <burek> something like "i want constant quality" [19:19] <burek> 23 is the default, and if you go up, the quality gets worse, if you go down, the quality gets better (so the filesize gets bigger also) [19:19] <burek> crf=0 is near lossless [19:20] <sacarasc> crf 0 IS lossless. [19:20] <sacarasc> With x264. [19:20] <luc4_mac> burek: yes, but so not mandatory right? And in case it was not specified? [19:29] <burek> <burek> 23 is the default [19:44] <luc4_mac> burek: so it is constant quality anyway? Not variable. [19:45] <JEEB> more or less [19:45] <JEEB> constant rate factor is possibly the closest thing that an algorithm can get to such a thing [19:45] <JEEB> (that's what crf means) [22:24] <elkng> I want to convert "mkv" video into "avi" but there are 2 audio tracks, what is the option to set audio track's id to use ? [22:25] <Guest92268> -map [22:29] <elkng> "ffprobe" shows: "Stream #0.0(lang): Video: h264 Stream #0.1(lang): Audio: aac Stream #0.2(lang2): Audio: ac3" how can I use "-map" option ? [22:30] <elkng> "-map 0.2 -map 0.0" ? [22:39] <elkng> I want to convert video starting from 2 minute and convert 20 seconds, I use option "-ss 0:2:0 -t 0:0:20", but it doesnt make anything [22:40] <elkng> just prints strings "frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit" again and again [22:56] <elkng> how mapping working ? [22:59] <elkng> there is an option "-aid" in mencoder, its quite straightforward to use [23:16] <elkng> better to use mencoder [23:18] <mmu> RTFM :p [23:18] <mmu> it's a bit awkward but it works [23:23] <elkng> they forgot to write a little warning into ffmpeg's man page: "we have an option -map but one need to be a scientist to uderstand how it works nahahaha" [23:33] <taqattack> I have a question about piping. If i have ffmpeg outputting something into another ffmpeg as input. Where does the output from the first get stored? [23:36] <cbsrobot> elkng: be more verbose [23:36] <cbsrobot> it's hard to help you if you do not show the whole command [23:40] <microchip_> taqattack: it doesn't get stored but it gets transfered to the second ffmpeg instance [23:41] <microchip_> taqattack: piping is transfering the output of one program to be the input of the other [23:42] <elkng> cbsrobot: I have video file, the information ffprobe prints is: "http://sprunge.us/cUde" and I want to use the command: "ffmpeg -i 1.mkv -vcodec mpeg4 -vb 2000k -s 650x276 -ab 128k -ar 44100 -ac 2 -acodec libmp3lame -y -f avi 1.avi" but that command gets second audio stream: "Stream #0.2(hin): Audio" I want first audio stream: "Stream #0.1(eng): Audio" [23:45] <cbsrobot> elkng try: ffmpeg -ss 120 -t 20 -i 1.mkv -map 0:0 -map 0:1 -vcodec mpeg4 -vb 2000k -s 650x276 -ab 128k -ar 44100 -ac 2 -acodec libmp3lame -y -f avi 1.avi [23:45] <cbsrobot> but I do not know how old your ffmpeg is ..... [23:46] <cbsrobot> so maybe some options will not work anymore , or could be written differently [23:46] <elkng> "Resampling with input channels greater than 2 unsupported." [23:46] <elkng> "Can not resample 6 channels @ 48000 Hz to 2 channels @ 44100 Hz" [23:46] <cbsrobot> and btw. the above command encodes only 20s - just to make sure you do not have to wait too long [23:47] <cbsrobot> try [23:47] <cbsrobot> ffmpeg -ss 120 -t 20 -i 1.mkv -map 0:0 -map 0:1 -vcodec mpeg4 -vb 2000k -s 650x276 -ab 128k -ar 48000 -ac 2 -acodec libmp3lame -y -f avi 1.avi [23:47] <elkng> same [23:48] <elkng> FFmpeg version 0.6.3 [23:48] <cbsrobot> waht version of ffmpeg ? [23:48] <cbsrobot> way too old ! [23:48] <cbsrobot> install from git [23:49] <elkng> problems with mencoder too, it generates file but mplayer doesnt play video only audio, MEncoder 1.0rc4-4.4.4 [23:51] <elkng> ffmpeg-0.10.2 latest stable ? [23:53] <cbsrobot> yes I guess [23:55] <cbsrobot> It should work - I just tested it [23:55] <cbsrobot> but I had an issue cutting and encoding - maybe you have to do it in two steps [23:59] <krobins> hi room. is someone available to answer a few quick questions about turning raw video into grayscale? [00:00] --- Sun Apr 22 2012

1 0

ffmpeg-devel.log.20120420
by burek 21 Apr '12

21 Apr '12

[00:29] <burek> is this a mistake on the website of ffmpeg: [00:29] <burek> March, 17, 2012, FFmpeg 0.10.1 [00:29] <burek> April 12, 2012, FFmpeg 0.7.12 / 0.8.11 [00:29] <burek> or ar there several version of ffmpeg [00:29] <burek> that are being maintained as "official" or something? [00:30] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r455fcf29b8 10ffmpeg/libavformat/avidec.c: [00:30] <CIA-17> ffmpeg: avidec: zero extradata padding [00:30] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:32] <burek> oh, I got my answer :) just ignore me :) [01:03] <CIA-17> ffmpeg: 03Robert Nagy 07master * r65fa7bc1f8 10ffmpeg/ (4 files in 2 dirs): [01:03] <CIA-17> ffmpeg: lavfi: add avfilter_unref_bufferp() [01:03] <CIA-17> ffmpeg: Signed-off-by: Stefano Sabatini <stefasab(a)gmail.com> [01:03] <CIA-17> ffmpeg: 03Stefano Sabatini 07master * r9750254885 10ffmpeg/ (5 files in 3 dirs): [01:03] <CIA-17> ffmpeg: lavfi/mp: remove libmpcodecs mirror filter [01:03] <CIA-17> ffmpeg: The native libavfilter hflip filter does the same thing. [11:03] <Tjoppen> any particular reason why libswscale doesn't consider alpha when going RGBA -> RGB? [11:04] <Tjoppen> I'd expect the background to become black, like with most other software [11:18] <Tjoppen> right.. adding support for that would be tricky [11:19] <michaelni> if you want to do it i dont mind applying a patch that adds optional blend to black support [11:19] <michaelni> it should be optional because it will be slower ... [11:22] <Tjoppen> indeed. after some thinking I realize it needn't be too hard, at least for common 8-bit formats [11:49] <Tjoppen> pixfmtdesc needs a bit in flags för alpha I think [15:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rd7a4c43f18 10ffmpeg/libavcodec/ffv1.c: [15:43] <CIA-17> ffmpeg: ffv1: Add a CRC check to the global header with version 1.3 [15:43] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r32883c0667 10ffmpeg/libavcodec/ffv1.c: [15:43] <CIA-17> ffmpeg: ffv1: split init_slice_state() out so individual slices can be inited [15:43] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rc7a435aab2 10ffmpeg/libavcodec/ffv1.c: [15:43] <CIA-17> ffmpeg: ffv1: split clear_slice_state() out so individual slices can be cleared. [15:43] <CIA-17> ffmpeg: This allows us to clear outside of the main thread for example. [15:43] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r9408316a85 10ffmpeg/libavcodec/ffv1.c: [15:43] <CIA-17> ffmpeg: ffv1: refactor slice decoding init loop so that the first is less a special case [15:43] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r044f7275d3 10ffmpeg/libavcodec/ffv1.c: [15:43] <CIA-17> ffmpeg: ffv1: add optional per slice CRCs to detect undamaged slices. [15:43] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * rb4fc53856d 10ffmpeg/libavcodec/ffv1.c: [15:43] <CIA-17> ffmpeg: ffv1: move most fields from the frame header to the slice headers with ffv1.3 [15:43] <CIA-17> ffmpeg: This will allow decoding a single undamaged slice even if all others are lost [15:43] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:43] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r8456089f50 10ffmpeg/libavcodec/ffv1.c: [15:43] <CIA-17> ffmpeg: ffv1: clear slice state in decode_slice() [15:43] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:42] <CIA-17> ffmpeg: 03Nicolas George 07master * r34c426be4a 10ffmpeg/ffmpeg.c: [16:42] <CIA-17> ffmpeg: ffmpeg: check encoder earlier. [16:42] <CIA-17> ffmpeg: Fixes a segfault with Ogg output, libtheora not compiled in [16:42] <CIA-17> ffmpeg: and no codec specified. [17:24] <michaelni> ubitux, the "-enable-shared + libs " box is yours ? it failed with "Xfixes not found" [17:25] <ubitux> yeah i know i just saw that [17:25] <ubitux> i messed with my packages [17:25] <michaelni> ahh ok [17:25] <ubitux> should be fixed in the next run [17:25] <ubitux> no worries :) [17:25] <ubitux> sorry about that [17:25] <michaelni> np [17:26] <ubitux> btw, my isp decided to completly fuck my connection, so no internet atm, and it may long for a few more days; it's likely i won't be much active [17:27] <ubitux> (yes as usual, i'm pretty lucky; and of course i'm the only one in the building) [17:37] <TimNich> No internet, yet still on IRC! ;) [17:37] <ubitux> hell yeah [18:29] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r1df49142ba 10ffmpeg/libavcodec/avs.c: [18:29] <CIA-17> ffmpeg: avsdec: Set dimensions instead of relying on the demuxer. [18:29] <CIA-17> ffmpeg: This fixes out of array writes. [18:29] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [18:29] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:29] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r2c22701c37 10ffmpeg/libavcodec/ac3dec.c: [18:29] <CIA-17> ffmpeg: ac3dec: Check number of output channels. [18:29] <CIA-17> ffmpeg: Fixes out of array write. [18:29] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [18:29] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:29] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r4a80ebe491 10ffmpeg/libavcodec/indeo3.c: [18:29] <CIA-17> ffmpeg: indeo3: Fix reallocation code so that it doesnt become inconsistent. [18:29] <CIA-17> ffmpeg: Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind [18:29] <CIA-17> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:59] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * rdd5e1fc22e 10ffmpeg/libavformat/mxfdec.c: [20:59] <CIA-17> ffmpeg: mxfdec: fix memleak on read error/EOF. [20:59] <CIA-17> ffmpeg: This also matches the rest of the demuxer which will return [20:59] <CIA-17> ffmpeg: partial packets. [20:59] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [21:16] <CIA-17> ffmpeg: 03Reimar Döffinger 07master * r9b1f776d75 10ffmpeg/libavutil/x86/x86inc.asm: [21:16] <CIA-17> ffmpeg: Fix compilation with NASM. [21:16] <CIA-17> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [22:32] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r0c0d1bce7c 10ffmpeg/ (5 files in 2 dirs): [22:32] <CIA-17> ffmpeg: avutil: add audio fifo buffer [22:32] <CIA-17> ffmpeg: The functions operate on the sample level rather than the byte level and work [22:32] <CIA-17> ffmpeg: with all audio sample formats. [22:32] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r870165e217 10ffmpeg/tests/Makefile: FATE: add a pattern rule for generating asynth wav files [22:32] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r010943c6ce 10ffmpeg/tests/audiogen.c: FATE: optionally write a WAVE header in audiogen [22:32] <CIA-17> ffmpeg: 03Justin Ruggles 07master * ra6c8cca2a8 10ffmpeg/tests/ (6 files in 4 dirs): [22:32] <CIA-17> ffmpeg: FATE: replace current g722 encoding tests with an encode/decode test [22:32] <CIA-17> ffmpeg: Avoids resampling and channel mixing. This only tests the behavior [22:32] <CIA-17> ffmpeg: with respect to input and output audio rather than also testing changes [22:32] <CIA-17> ffmpeg: to the encoder or muxer that do not affect the resulting decoded output. [22:32] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r03caef1bed 10ffmpeg/tests/ (8 files in 5 dirs): [22:32] <CIA-17> ffmpeg: FATE: replace the acodec-g726 test with 4 new encode/decode tests [22:32] <CIA-17> ffmpeg: Avoids resampling and channel mixing. This only tests the behavior [22:32] <CIA-17> ffmpeg: with respect to input and output audio rather than also testing changes [22:32] <CIA-17> ffmpeg: to the encoder or muxer that do not affect the resulting decoded output. [22:32] <CIA-17> ffmpeg: 03Justin Ruggles 07master * rb0e9edc44f 10ffmpeg/ (5 files in 2 dirs): [22:32] <CIA-17> ffmpeg: avcodec: add a cook parser to get subpacket duration [22:32] <CIA-17> ffmpeg: Fixes jittery video playback of rm files with cook audio. [22:32] <CIA-17> ffmpeg: 03Justin Ruggles 07master * r5052980400 10ffmpeg/tests/ (5 files in 5 dirs): [22:32] <CIA-17> ffmpeg: FATE: allow lavf tests to alter input parameters [22:32] <CIA-17> ffmpeg: Change some lavf tests to avoid resampling and channel mixing. [22:32] <CIA-17> ffmpeg: 03Michael Niedermayer 07master * r3194ab78a6 10ffmpeg/: (log message trimmed) [22:32] <CIA-17> ffmpeg: Merge remote-tracking branch 'qatar/master' [22:32] <CIA-17> ffmpeg: * qatar/master: [22:32] <CIA-17> ffmpeg: avcodec: add a cook parser to get subpacket duration [22:32] <CIA-17> ffmpeg: FATE: allow lavf tests to alter input parameters [22:32] <CIA-17> ffmpeg: FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test [22:32] <CIA-17> ffmpeg: FATE: replace the acodec-g726 test with 4 new encode/decode tests [22:39] <Compn> there was jittery real video with cook audio ? [22:39] Action: Compn hasnt noticed any reports nor files [00:00] --- Sat Apr 21 2012

1 0

ffmpeg.log.20120420
by burek 21 Apr '12

21 Apr '12

[00:00] <burek> your input stream might be damaged, or we are doing something wrong [00:00] <burek> try ffplay /dev/dvb/adapter0/dvr0 [00:00] <burek> do you see video normally [00:00] <taypen> burek, i suppose audio codec should be AAC and video H.264 [00:00] <burek> taypen, try typing just ffmpeg -i input.mkv [00:01] <burek> and use pastebin to copy the output [00:01] <burek> so we can tell you exactly [00:01] <xandy> burek i've updated ffmpeg now and i'll try it now again with the defaults [00:01] <burek> ok [00:02] <loo> burek, that's a server machine, so I cant do that [00:02] <loo> if I cat that stream [00:02] <burek> i see [00:02] <loo> I can open it on my windows machine [00:02] <burek> if you cat it to a test.ts file [00:02] <loo> burek, sample - http://beberry.lv/up/u/sample [00:02] <burek> and then use ffmpeg -i test.ts -vcodec copy -acodec copy test2.ts [00:02] <burek> would it be the same size? [00:02] <burek> filesize* [00:03] <loo> will try [00:03] <taypen> burek, http://pastebin.com/aekXpzsY [00:04] <burek> Stream #0.0: Video: h264 (High) [00:04] <burek> and 2 ac3 audios [00:04] <burek> eng/rus so I guess you'll have to convert it to aac [00:04] <burek> but try only with that remux command [00:04] <loo> burek, outcome is greater in size [00:04] <burek> and see if that works [00:05] <taypen> burek, ok thanks [00:05] <loo> burek, were you saying that to me? [00:05] <xandy> size= 3188416kB time=309096:19:12.00 bitrate= 0.0kbits/s [00:05] <xandy> hm didn't worl [00:05] <xandy> l [00:05] <basilgohar> burek: http://paste.basilgohar.com/view/86958547 [00:05] <taypen> burek, but in case i'll have to convert the audio, how i do that? [00:05] <burek> taypen, if it doesn't, then try: ffmpeg -i input.mkv -acodec aac -strict experimental -ar 44100 -ab 96k -ac 2 -vcodec copy -scodec copy out.mp4 [00:06] <burek> loo, if test.ts and test2.ts don't match at filesize, then test.ts is not a valid mpegts file I guess [00:06] <burek> the test was to see if the -vcodec copy would produce the same thing (which it should) [00:07] <burek> but if it doesn't that means something is not right with the input I guess [00:07] <loo> burek, the test2 is greater,, Burek, can you take a look at the sample file, please? [00:07] <burek> you can try uploading a sample to the ffmpeg's ftp [00:07] <burek> for developers to investigate [00:07] <burek> and submit a bug report, pointing to that sample [00:07] <burek> read that loo [00:08] <loo> ah, I added a ticket, and added the link to sample, but no response yet. I'm just burning, because I can't solve this for a quite a long time.. [00:08] <xandy> burek tryed ffmpeg -i video.flv -acodec copy -vn audio.flv now and got [flv @ 0x21e39a0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 461352 >= 461306 [00:08] <loo> I assume, that you are more engaged in that community, so you could maybe post something there [00:09] <burek> basilgohar, the problem with simple concatenation of video streams is that it works mostly with mpg streams [00:09] <burek> all the other streams have some problems with it (if you just use cat) [00:09] <burek> and mpegts container is such a container that it breaks the stream into packets [00:10] <burek> so maybe that's why all the packets played back correctly, but when you use some format like mkv [00:10] <burek> then those concatenated streams make problems [00:10] <burek> try using some simple video editor to first concatenate those streams into one [00:11] <burek> and then use ffmpeg to encode it (or remux) to whatever you like [00:11] <burek> loo, I know the feeling :( [00:11] <burek> but try to be patient [00:11] <burek> no report is being ignored [00:11] <burek> it's just the lack of free time to manage all of them [00:11] <burek> also, update your report with any new investigations that you do [00:12] <burek> which might help speed up its resolving [00:12] <burek> xandy [00:12] <burek> can you please use pastebin.com, to show your command line and its output? [00:12] <xandy> ok [00:12] <loo> burek, ok, but this is too frustrating, a feeling that no one cares.. ++ can I add as a new post, or should I edit old ones? [00:14] <burek> dizzylizzy, if you yourself don't stretch/extend it, then you're ok with that logic :) just extract, make changes (and make sure that duration is about the same) and then just mux it back together [00:14] <burek> loo, if it is related to an old one, then update [00:14] <burek> and leave the link here [00:14] <taypen> burek, is there any difference between m4v and mp4? [00:15] <loo> burek,ok, thank you, I'm tired, have to go. thanks again! [00:15] <burek> m4v is a video only and mp4 can contain audio too [00:16] <taypen> hmm, the movies i downloaded for ipad only ends with m4v [00:16] <xandy> burek ffmpeg -i video.flv -acodec copy -vn audio.flv -> http://pastebin.com/DJyEmJUj [00:17] <burek> well then use ffmpeg -i ... out.m4v [00:18] <taypen> ok :) [00:18] <burek> hm, xandy try just unpacking your audio, to see if it is not broken [00:18] <burek> like this [00:18] <burek> ffmpeg -i video.flv out.wav [00:20] <burek> xandy, your ffmpeg is still very old [00:20] <xandy> :( [00:20] <burek> 0.10.x is now something like latest [00:20] <burek> can you add debianmultimedia repos [00:20] <burek> and just install ffmpeg from there? [00:21] <burek> http://debian-multimedia.org/ [00:26] <xandy> burek thats what i did but the machine is squeeze [00:27] <burek> xandy, did you try just extracting the audio to wav [00:27] <xandy> yea. my tool still says it's broken but i'll try it now with vlc again [00:27] <burek> try extracting it with ffmpeg [00:28] <burek> and if it stops at the same point in time [00:28] <burek> 00:07:33.31 [00:28] <burek> then the input audio is broken I guess [00:50] <xandy> burek thanks for your help, i think ill give it a try again tomorrow [00:50] <burek> ok :) [00:50] <burek> :beer: [01:34] <biscii> does anyone know the best command to use for ffmpeg (mp3) -> segments for audio only ? [01:34] <biscii> I've found a few segmenting examples [01:34] <biscii> but its mostly for video [01:50] <saste> biscii: try the segmenter [01:51] <biscii> the apple seqmenter ? [01:51] <biscii> or ion cannon one ? [01:52] <saste> biscii: no the native ffmpeg segmenter muxer [01:52] <saste> also check this: http://gitorious.org/~saste/ffmpeg/sastes-ffmpeg/commits/misc-segment-fixes… [01:52] <saste> since the current implementation may be borken with audio only [01:52] <saste> feel free to report to me personally if the branch doesn't work [01:52] <saste> i want to merge it soon anyway [01:52] <biscii> cool will do. Thanks [02:07] <basilgohar> burek: Thanks for getting back to me. I'll try with different containers and see what I find. [03:46] <taqattack> when encoding to x264. My threads is set to 6 but I only have 4 cores. Does this hinder my encoding performance? [04:09] <dan3son> Hi [04:10] <dan3son> I like how your topic mentions x264 encoding, I'm at videolan's x264 codec website, can't find any documentation describing tune/preset/profile [04:11] <dan3son> I just want to transcode all my videos so they'll play on my TV w/out on-the-fly transcoding [04:18] <taqattack> This link helped me a lot: http://mewiki.project357.com/wiki/X264_Settings [04:18] <dan3son> I can't seem to find the plethora of x264 presets either, such as 'hq' [04:18] <dan3son> My copy of ffmpeg only has two, and they're for ipods [04:22] <taqattack> Which build are you using? [04:23] <dan3son> windows, N-39877-g4fa706a [04:23] <dan3son> The latest windows build, essentially [04:24] <taqattack> and you are trying to convert a file into h264 format? [04:25] <taqattack> essentially, you'll want something like "ffmpeg -i <inputfile> -vcodec libx264 -preset <ultrafast, superfast, veryfast, faster, fast, medium, slow, slower, veryslow, placebo> -acodec <libmp3lame,libvo_aacenc> -f <mp4,flv,mkv> <outputfile>" [05:32] <dan3son> I'm getting a bug, trying some predefined stuff [05:32] <dan3son> -flags2 +bpyramid+wpred+mixed_refs+dct8x8 [05:33] <taypen> hey, i've got some problem converting mkv to mp4 : http://pastebin.com/U2eiAv7g [05:33] <dan3son> http://pastebin.com/AU6uZcyp [05:34] <dan3son> I don't understand why I'm getting the error in processing my arguments [05:35] <taypen> taypen, well past post been marked as spam :( here's another one http://pastebin.com/RkmvwALh [05:36] <dan3son> [NULL @ 00000000003cc570] Unable to parse option value "bpyramid+wpred+mixed_ref [05:36] <dan3son> s+dct8x8" [05:49] <relaxed> dan3son: use a preset [05:52] <dan3son> If I use -s 1920x1080 it will scale of course [05:52] <dan3son> How does it do aspect ratio conversions by default? [05:53] <dan3son> Like if I'm using a 4:3 source what does it do? [05:54] <relaxed> stretch [05:55] <relaxed> ffmpeg now has video filters that have more advanced options, like keeping the aspect ratio. [05:55] <relaxed> look at the man page or read about it on the website. [05:57] <relaxed> taypen: 2> log [05:58] <taypen> relaxed, nah i just wanted to use pastebin program under linux [06:00] <taypen> relaxed, same error without "> log" [06:00] <relaxed> taypen: oh, use .... -f mp4 Contrabanda.m4v [06:01] <relaxed> or omit the subs [06:01] <taypen> same [06:01] <taypen> how do i omit them? [06:02] <relaxed> remove -scodec copy, add -map 0.0 -map 0.1 [06:03] <taypen> it means map the to nothing? [06:03] <relaxed> do you want english or russian audio? [06:03] <taypen> russian [06:03] <relaxed> it means map (use) only the streams you specify. [06:04] <taypen> ok [06:04] <taypen> i'm recompiling ffmpeg now, added enabled some flags [06:05] <taypen> what i'm writing... without added * ) [06:05] <relaxed> I hope you added support ofr another aac encoder such as libfaac or libvo-aacenc [06:06] <relaxed> you already have libvo-aacenc support [06:06] <relaxed> use -acodec libvo_aacenc instead, remove "-strict experimental" [06:06] <taypen> i'm adding v4l sdl theora and vorbis support [06:07] <relaxed> you're using gentoo, aren't you? [06:07] <taypen> yes [06:07] <relaxed> gentoo's default use flags for ffmpeg are fucking retarded. [06:08] <relaxed> you would be better off compiling ti by hand. [06:08] <taypen> :) [06:08] <relaxed> it* [06:08] <taypen> maybe if it wont work with this flags [06:10] <taypen> and the latest portage tree have old version of ffmpeg [06:10] <relaxed> One time a guy was in here and he had the "-encode" use flag set. [06:12] <taypen> yes but gentoo is not so bad :) [06:13] <relaxed> Yes it is. If it allows you to compile a butchered version of ffmpeg, how else has it allowed you to shoot yourself in the foot. [06:14] <relaxed> ? [06:14] <taypen> gentoo rely on user [06:14] <taypen> what linux are you using? [06:14] <relaxed> I'm aware of that. I'll stop trolling your choice of distro. [06:15] <relaxed> Debian Sid [06:18] <taypen> ok "use -acodec libvo_aacenc instead, remove "-strict experimental"" this line helped [06:19] <taypen> or maybe it's because i removed subs [06:20] <taypen> yes it's because i removed subs and added maps [06:20] <relaxed> both helped [06:21] <taypen> no i tried now with old acodec and with experimental [06:21] <relaxed> you'll get better quality with libvo_aacenc [06:23] <taypen> isn't those should be the same aac [06:24] <taypen> well changed to libvo_aacenc ) [06:24] <relaxed> different encoders of the same codec. libvo_aacenc > aac [08:37] <neonoe> Hi, howto mix tow video streams by incrusting one of them as reduced image overlaid on the other kept at full size ? [08:37] <neonoe> I have seen old topics on this subject on the web. I would know if you have some tricks for such purpose. [12:51] <loo> Any ideas how to change the buffer size of ffmpeg? http://ffmpeg.org/trac/ffmpeg/ticket/1224 Or is there a different way how to pass that stream? [14:57] <loo> what's the format of this sample http://beberry.lv/up/u/sample ? [15:00] <loo> burek, I don't know those problems were related to buffer, because once I got the ffmpeg to convert 25 seconds.. which was 2x the usual size.. [15:09] <basilgohar> burek: Returning to my issue, I noticed that the Matroska I made from remuxing the AVCHD files appears to have slightly different frame rate properties than the originals...namely, the tbc value is 59.92 instead of what should probably be 59.94 (i.e., 60000/1001). [15:09] <basilgohar> So, I think the issue I am observing has to do with the fact that Matroska has a limit precision when it comes to frame time lengths. [15:09] <basilgohar> *limited [15:09] <basilgohar> This is my suspicion, because I'd hear of this from another channel. [15:10] <basilgohar> It might also explain why I get two dropped frames. I wonder if I can specify the framerate when muxing, and if that would make any difference... [15:11] <basilgohar> I could also just be completely clueless and talking nonsense. :) [15:13] <Mavrik> basilgohar: I've noticed dropped frames when concating AVCHD files as wel [15:13] <Mavrik> l [15:13] <Mavrik> basilgohar: right on the file boundaries [15:14] <stowelly> Hi guys, does anyone know much about the Dirac / Schrodinger formats at all? [15:15] <Mavrik> basilgohar: my solution was to use Premiere to put them together and remux them to mp4 [15:15] <Mavrik> basilgohar: it seems that those other index files around MTS files were required for our Canon camera [15:16] <stowelly> im trying to assess for our game wether we could get a smaller filesize than Bink without sacrificing on quality, but in all my tests using ffmpeg the file sizes have been quite huge (however possibly im using the wrong encode params) [15:17] <loo> Mavrik, can you please take a look at this http://ffmpeg.org/trac/ffmpeg/ticket/1224 ? [15:22] <basilgohar> Mavrik: Can you suggest any other way to remux AVCHD files without using closed software? I never noticed glitches when playing back files muxed in this manner, and I do it all the time... [15:23] <basilgohar> I guess one dropped frame might slip by me, though. [15:23] <Mavrik> basilgohar: I didn't find any [15:23] <basilgohar> stowelly: There's a #dirac channel, and the implementer of libschroedinger (ds) is usually there. [15:24] <stowelly> thank you basilgohar [15:24] <basilgohar> If you just want to avoid licensing, then you might consider using VP8/WebM. [15:24] <basilgohar> I mean, licensing costs. [15:26] <stowelly> our legal team arent happy with that, particulary with the legal sh*t storm looming, think they estimated we would have to pay MPEG LA about 0.4 gbp per unit if they win [15:27] <loo> anyone? [15:39] <iive> stowelly: how about just using mpeg1 and good encoder? [15:40] <stowelly> dont you have to pay royalties for distributing a mpeg1 decoder? [15:41] <iive> i think I don't, and even if you do, it would soon become 20 years from the publishing of the standard. [15:42] <iive> "The standard was finished with the 6 November 1992 meeting" [15:47] <stowelly> ahh apparently after speaking to my lead, that mpeg 1 wasnt good enough for what we need, i.e file size to quality ratio [15:49] <iive> well... this is why I said "good encoder". One of big problems with mpeg1/2 is their inexact IDCT definition. This forces it to use frequent keyframes that ruin the compression. [15:50] <iive> mpeg4 main improvement was more strictier (i)dct constraints. h264 for example uses a bit-exact one. [15:51] <iive> I remember an iso working on mpeg1/2/4 "standard" like idct that is also defined to be bitexact. It may be viable "workaround". [15:54] <basilgohar> stowelly: You mean your legal team isn't happy with using WebM? [15:54] <taqattack> I have a question about licencing ffmpeg. I'm writing an GUI application for FFmpeg. I intend to keep the GUI code closed source. So does that mean I can't distribute FFmpeg with software? [15:55] <iive> basilgohar: webm/vp8 is known to use some things that are very similar to patented stuff in h264. [15:55] <Mavrik> taqattack: you can't even link to it. [15:56] <Mavrik> unless you make sure it's LGPL and your licence is LGPL compliant [15:56] <iive> Mavrik: are you kidding? [15:56] <Mavrik> (not all en/decoder licenses are LGPL compliant so you'll have to build your ffmpeg without those) [15:57] <Mavrik> you can, of course, invoke it as a process [15:57] <iive> (obviously I wrote my question after seeing only your first line). [15:57] <Mavrik> basilgohar: last time I checked MPEG LA was claiming patent royalties for WebM as well ;) [15:58] <taqattack> So if I build LPGL version of ffmpeg and create dlls. I can distribute it with my software? [15:58] <Mavrik> yes [15:59] <iive> imho (not legal) but lgpl should allow you to distribute the unmodified ffmpeg binary with any closed source program. As long as you follow the lgpl rules (provide offer to the source etc...). [15:59] <Mavrik> iive: you can always distribute the binary [15:59] <Mavrik> even if it's GPL [15:59] <Mavrik> (unless it's specifically not distributable) [15:59] <Mavrik> difference is if you can link against ffmpeg (libav*) libraries with your closed-source program [16:00] <Mavrik> calling ffmpeg.exe from your closed-source program does not violate GPL [16:00] <taqattack> Oh really [16:00] <iive> with gpl it may be a little bit more tricky, depending of interpretation of the "linking to". aka is the usage generic or specific. [16:00] <basilgohar> Mavrik: Nope. MPEG-LA made a public call for people interested in forming a pool around VP8. Nothing else followed, and that was over a year ago. [16:00] <taqattack> But I was wondering if I could redistribute the ffmpeg.exe with GUI application as package [16:00] <iive> but it should still allow you to distribute them separately. [16:00] <Mavrik> basilgohar: that and they made public claims VP8 infringes their patents [16:01] <Mavrik> basilgohar: as of yet that was not tested in court [16:01] <basilgohar> iive: Yes, very similar, and very obviously worked around issues known to be patented. [16:01] <basilgohar> Mavrik: Very few things are tested in court. [16:01] <Mavrik> well, it's enough to make our legal team nervous [16:01] <stowelly> either way, if it gets tried in court and loses it will cost us half a million! [16:01] <basilgohar> Yes, they've made plenty of FUD-like claims around VP8, because an unencumbered codec directly threatens their business model. [16:01] <iive> basilgohar: legal matters are like quantum physics. [16:02] <Mavrik> it'd be annoying if you'd have to pay ton of royalties for "free" WebM when MPEG-LA starts trolling around [16:02] <Mavrik> basilgohar: of course they did [16:02] <basilgohar> I'm not saying depend on Google to defend you, but a lot of companies have already adopted VP8 and have built software and hardware around it. [16:02] <Mavrik> basilgohar: but with US legal system being what it is [16:02] <Mavrik> trusting your business to the "logic" of non-infringement of VP8 is... stupid [16:03] <Mavrik> at least H.264 is a known evil [16:03] <iive> well, even if you pay mpeg-la, you could still be sued by 3'd parity. [16:03] <Mavrik> iive: cue Motorola vs. Microsoft :D [16:03] <iive> I think there was some such "claim" not long ago. [16:04] <Mavrik> yeah, Motorola came and sued Microsoft for H.264 - MPEG LA doesn't own whole H.264 patent pool :D [16:04] <Mavrik> Microsoft moved their datacenters from Germany to Nederlands because of it [16:04] <iive> Mavrik: to be precise, mpeg-la doesn't own the whole patent pool :) [16:05] <basilgohar> Mavrik: MPEG-LA doesn't provide indemnification, either. [16:05] <basilgohar> And, as pointed out, it doesn't provide protection from lawsuits outside of their pool either. [16:05] <basilgohar> VP8 was designed to avoid patented techniques, and it's backed and being actively used by Google and many software and hardware vendors. [16:06] <Mavrik> the most reasonable option is to just not do business with US if at all possible [16:06] <iive> the problem is WIPO [16:06] <Mavrik> opening your business to patent trolling can just bring too much damage to the company [16:06] <basilgohar> Mavrik: Even though I live in the US, I find your statement very reasonable! [16:06] <Mavrik> I just hope that part of law system gets revised asap :( [16:06] <loo> http://ffmpeg.org/trac/ffmpeg/ticket/1224 [16:07] <basilgohar> The US is trying to move to a more "IP"-oriented economy, because it's easier to make money off of something you don't need a factor to make and trains to ship. [16:07] <iive> Mavrik: they just made one patent reform. Don't you feel the improvements they brought? [16:09] <Mavrik> iive: oh? when was that? [16:09] <Mavrik> (we mostly focused on EU lately) [16:09] <Mavrik> ugh, gotta go, see ya guys :) [16:09] <iive> 1-2 years ago. They moved from invented first to filled for patent first (like it is in EU). And changed few minor other things. [16:35] <basilgohar> Okay, so I think I was able to remux the file from an h264 elementary stream using gstreamer. I have to play with the files now, but at least I have files! [17:02] <someone-noone> I want to synchronize audio to external clock(yes, yes i know, look at ffplay.c). I understand that i need to update my audio_clock every time i read audio packet and every time i read some audio samples. In my program i don't operate double values, instead i'm using uint64_t which represents timestamp in milliseconds. So my external clock is just current timestamp. The problem is: how to correct calculate audio_clock. Here is what [17:02] <someone-noone> 1) when program starts i make audio_clock = gettimeofday. [17:03] <someone-noone> 2) later, when i'm trying to decode and play audio data i need to update my audio_clock [17:04] <someone-noone> in "GetPacket" i do: [17:04] <someone-noone> if (_audio_pkt.pts != AV_NOPTS_VALUE) [17:04] <someone-noone> _audio_clock += _audio_stream->time_base.num*_audio_pkt.pts/_audio_stream->time_base.den; [17:04] <someone-noone> in "GetSamples" i do: [17:04] <someone-noone> n = 2 * _audio_stream->codec->channels; [17:04] <someone-noone> _audio_clock += _audio_chunk_size / (n * _audio_stream->codec->sample_rate*1000); [17:06] <someone-noone> the problem is i don't understand if i'm doing it correctly and if no, how should i do it? I don't understand what exactly "pts" represents, can someone explain? [17:06] <iive> presetnation timestamps. [17:06] <iive> aka the time when the sample should be presented. [17:07] <iive> however with audio the things are a little bit complicated. [17:07] <someone-noone> iive: in seconds, right? and it should be converted to human value with time_bas [17:07] <someone-noone> ? [17:07] <iive> the problem is that when you have 44100 samples per second, you can assume that playing 44100 would advance the clock with exactly 1 second. [17:08] <iive> however if you use wall clock (like get time of the day) you may find that it sometimes gets less than second, and sometimes more than a second. [17:09] <iive> some of the audiocards use separate oscilator thus they playback it slower of faster. [17:10] <iive> so for example with mpeg streams up to 3 clocks can be encoded. video, audio and system (of the system supposedly captured the a/v). [17:10] <iive> so, if there is audio only, you can do 2 things. [17:11] <iive> first use only the audio as clock. rely entirely on the samples played. [17:11] <iive> second... monitor both clocks and resample the input to compensate for the drift. [17:12] <iive> well. [17:12] <iive> audio clock should be relative to the audio stream, not the time of the day. [17:13] <iive> same goes for the usual timestamps. [17:15] <someone-noone> so, if i understand right, it should be 0 at beginning and should grow each time i get packet\decode frame, but what should be external clock than? If i make audio clock to gettimeofday()("in milliseconds") and than will add pts time to this value and use external clock as simple gettimeofday() ? [17:17] <iive> someone-noone: why you need external clock? [17:17] <rly> Is there any combination of ffmpeg + open-source viewer which allows me to create a screencast? [17:18] <rly> All ways I have tried result in excuses being thrown by ffmpeg or really bad output or segfaults when I watch it with mplayer or vlc. [17:18] <rly> It seems rather terrible that mplayer and vlc crash in both cases. [17:18] <rly> Somehow once rich media is involved it seems OK to crash. [17:19] <rly> Really, really questionable practices. [17:19] <someone-noone> iive: i need to synchronize to smth :) when i decode some avi file and want to play just audio it plays very fast. [17:19] <someone-noone> rly: you my try ffplay ? [17:19] <someone-noone> may* [17:19] <someone-noone> rly: and with which software are you capturing screen? platform? [17:19] <rly> someone-noone: ffmpeg -f oss -i /dev/audio -f x11grab -r 30 -s 1645x579 -i :0.0 -sameq screengrab.avi [17:19] <rly> someone-noone: I have tried that command for example. [17:20] <rly> someone-noone: either the output is wrong or all of the viewers don't work. [17:20] <rly> Or both. [17:20] <rly> someone-noone: Linux. [17:20] <someone-noone> rly: ffplay plays that screengrab? Is there any errors while encoding? [17:20] <iive> someone-noone: if you have to synchronize to something you don't have control over, then I guess you would have to resample of pitch the audio faster/slower to compensate for the sound card drift. [17:21] <rly> someone-noone: WARNING: library configuration mismatch [17:21] <iive> however "audio plays very fast" indicates there is some kind of another problem, not just clock drift. [17:21] <someone-noone> rly: try to rebuild your ffmpeg [17:21] <rly> someone-noone: I already did. [17:22] <rly> someone-noone: that's what I got when I did this. [17:22] <rly> What I don't get is why that error message exists in the first place. [17:22] <rly> It shouldn't be possible to get in this state in the first place. [17:22] <iive> rly: why are you scaling to odd dimensions? [17:22] <someone-noone> rly: clean old version of ffmpeg, download new version -> configure & make & make install [17:23] <rly> someone-noone: I have git ffmeg, define 'clean'. [17:23] <rly> someone-noone: make clean? [17:23] <rly> someone-noone: make uninstall? [17:23] <someone-noone> rly: you shouldn't get this warning [17:23] <rly> someone-noone: and yet someone made a mistake such that I do get it. [17:23] <rly> Sorry for being the messenger. [17:24] <iive> rly: that commands works for me. I think i have the last release installed. [17:24] <rly> I don't even get the error message. [17:24] <someone-noone> iive: what kind of problem it can be? I'm using WinMM to play LPCM, if i load some uncompressed wav file to it and play it in loop it works fine [17:24] <rly> A mismatch is always between two things. [17:24] <rly> It should tell me exactly what doesn't match. [17:24] <rly> And then it should also tell me why that is a problem. [17:25] <rly> Finally, it should tell me a sequence of steps that will solve that problem. [17:25] <rly> This is basic software engineering. [17:25] <rly> Software engineering shouldn't involve having to fix other peoples' broken software. [17:25] <rly> If everyone would have to do that, nothing would help. [17:26] <iive> rly: it said library. ffmpeg is probably build with shared libraries, and they have been overwritten. [17:26] <iive> it is likely ld.so error. [17:26] <rly> Ok, I see it. [17:27] <rly> ldconfig wasn't run. [17:27] <rly> But then I still get another error. [17:27] <rly> SDL_OpenAudio: [17:27] <rly> X Error of failed request: BadValue (integer parameter out of range for operation) [17:28] <iive> try without sound -an i think. [17:28] <rly> Is recording sound (possible with Windows 95) too advanced? [17:28] <iive> depends on libsdl. [17:30] <rly> X Error of failed request: BadValue (integer parameter out of range for operation) [17:30] <rly> Major opcode of failed request: 133 (XVideo) [17:30] <rly> Minor opcode of failed request: 19 () [17:30] <rly> Value in failed request: 0x90 [17:30] <rly> Serial number of failed request: 27 [17:30] <rly> Current serial number in output stream: 28 [17:31] <rly> iive: similar issue. [17:31] <rly> iive: this is without sound. [17:31] <taqattack> Ok so from what I gather, I can call ffmpeg.exe from my GUI application which I intend to be closed source. But I cannot build and redistribute the ffmpeg.exe with my application. Is this corect interpretation of the licensing? [17:31] <rly> taqattack: what's the ffmeg licence? LGPL? [17:31] <rly> (some parts are GPL) [17:31] <taqattack> I'm using the zeranoe build which is GPL [17:33] <iive> rly: are you running under X server? have you escalated your privilages (su) ? [17:33] <rly> iive: yes and no. [17:33] <iive> can you e.g. start a new terminal from the cli? [17:34] <rly> iive: I can run xterm from a terminal emulator. [17:34] <rly> iive: what do you mean? [17:35] <iive> just checking if the programs have correct display link. [17:35] <rly> iive: the problem lies with epic failure on the side of ffmpeg. [17:35] <iive> as I said... works for me. [17:36] <iive> so the problem is somewhere on your end. [17:37] <rly> iive: the software exits with an X error. [17:37] <iive> X means Xorg server. [17:37] <rly> iive: it doesn't say 'we tried to call this and that X function, which we know must work because of reasons X, Y and Z', but it didn't work. [17:38] <rly> In short, I have no reason to assume this isn't a programming error. [17:38] <rly> Because otherwise it would have worked. [17:38] <iive> rly: actually this is exactly what is says. X returned error the program didn't expect. [17:39] <rly> iive: that's an X error message. [17:39] <iive> try without the -s [17:40] <rly> [mpeg4 @ 0x141a080] Error, Invalid timestamp=40, last=40 [17:40] <rly> Video encoding failed [17:40] <rly> Yeah, that's much better. [17:40] <rly> ... [17:40] <rly> Also, it just says 'failed'. [17:41] <rly> It doesn't say 'byte 90' expected. [17:41] <rly> Or 'you have byte 90 10 30, but the specification says it has to be byte 90 10 31' or whatever. [17:41] <rly> This is just amateuristic crap software. [17:42] <iive> whatever. [17:42] <rly> I am not the only one who has these issues with ffmpeg. [17:42] <iive> try at #libav [17:49] <rly> iive: it does play with mplayer, but it is full of graphical bugs. [17:49] <rly> iive: so, that leads to the conclusion that not only ffplay stinks, but the recording part too. [17:50] <iive> I told you the last release works fine. [17:50] <rly> iive: so, you are saying that git is broken? [17:50] <rly> iive: fine, I will compile the latest release to prove you wrong. [17:50] <rly> iive: compilation options for ffmpeg please. [17:50] <rly> iive: it must be the exact same thing. [17:50] <iive> probably. one of the prices of using developer version software. [17:50] <rly> iive: otherwise it is still comparing apples to oranges. [17:51] <iive> rly: same options. [17:51] <rly> iive: no, I mean compile time options. [17:51] <iive> you mean, the ones I've used? [17:51] <rly> iive: ffmeg --version | someparstecommand [17:51] <rly> iive: you say it works. I say it doesn't. I would have to use the same options as you did to make it work. [17:52] <rly> iive: the Debian Stable version _also_ doesn't work, btw. [17:52] <iive> configuration: --prefix=/usr --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-pthreads --enable-static --enable-shared --enable-libfaac --enable-libtheora --enable-x11grab --enable-libmp3lame --enable-libvorbis --enable-libx264 --enable-libxvid --cpu=core2 --enable-debug=3 --disable-stripping [17:52] <rly> iive: so, it is not as if stable releases mean _anything_. [17:52] <iive> rly: debian doesn't even package ffmpeg. [17:52] <rly> iive: Debian Multimedia does. [17:53] <iive> 0.10 is the one that works. [17:54] <iive> and yes, the current git does flood my console with "invalid dropping st:0" [17:54] <iive> the resulting file still plays fine. [17:55] <rly> iive: I have an Intel Core cpu, how is that named in CPU? [17:55] <rly> (that's yet another bug in the configure --help output) [17:55] <rly> It should list all valid values for CPU. [17:55] <iive> it should query them from the gcc -march [18:00] <rly> iive: that release doesn't even come with --enable-gpl. [18:01] <rly> Hmm, or perhaps it does. [18:01] <rly> Odd [18:01] <iive> i just thought you have connected from parallel universe. Was going to ask for presidents names and such :E [18:02] <rly> Ok, compiling... [18:06] <rly> [swscaler @ 0xa2d2e0] 0x0 -> 640x480 is invalid scaling dimension [18:06] <rly> [swscaler @ 0xa2d2e0] 0x0 -> 640x240 is invalid scaling dimension [18:06] <rly> [swscaler @ 0xa4aec0] 0x0 -> 640x240 is invalid scaling dimension [18:06] <rly> Failed to inject frame into filter network [18:06] <rly> iive: then we get that. [18:06] <rly> iive: with command: ffmpeg -f x11grab -r 30 -i :0.0 -sameq screengrab.avi [18:09] <rly> Do you even ever test your software? [18:10] <rly> I think there have been 4 issues reported for this in the past 3 years and all of them didn't fix it. [18:10] <rly> Who reviews the patches? Santa Clause? [18:12] <iive> michaelni, so you have patches that are not applied? [18:12] <iive> actually elvis presley. :E [18:13] <relaxed> rly: you have to specify the frame size before the input and don't use -sameq [18:14] <rly> relaxed: before the input? [18:14] <rly> relaxed: just show a complete command that works. [18:14] <rly> relaxed: otherwise it's just comparing apples and oranges all over again. [18:14] <relaxed> ffmpeg -s WxH ... -i INPUT ... [18:15] <rly> relaxed: ffmpeg -f x11grab -r 30 -s 897x579 -i :0.0 -sameq screengrab.avi [18:15] <rly> relaxed: results in epic failure similar to above. [18:16] <relaxed> which version? [18:17] <rly> ffmpeg version 0.10.2 Copyright (c) 2000-2012 the FFmpeg developers [18:17] <rly> But I do get WARNING: library configuration mismatch _again_. [18:19] <relaxed> remove the old libs [18:19] <rly> relaxed: which ones? [18:19] <rly> relaxed: the error message should tell me what to do. [18:20] <rly> It created the problem, so it can solve it. [18:20] <relaxed> which distro? [18:21] <rly> relaxed: Debian, but I built it from source. [18:21] <rly> It shouldn't even install if there is some problem. [18:21] <rly> The theorem 'after installation it works' should always hold. [18:22] <relaxed> try, LD_LIBRARY_PATH=/usr/local/lib ffmpeg ... [18:22] <relaxed> stop spouting your nonsense [18:22] <rly> It's not nonsense. [18:22] <iive> or run ldconfig again... [18:22] <rly> I already ran ldconfig. [18:22] <rly> Didn't help this time. [18:22] <rly> No idea why not. [18:23] <iive> probably because my options are used to build my own package. [18:23] <iive> so they ended up on a different location (prefix=/usr) [18:23] <rly> iive: I changed the prefix. [18:23] <relaxed> to what? [18:24] <rly> Ok, never mind. That's the problem this time. [18:24] <rly> Still, I don't expect it to work. [18:24] <iive> of course not. [18:24] <iive> these liberal hippies can't even tie their shoes correctly. [18:25] <rly> That has been my experience, yes. [18:25] <rly> Most of them shouldn't be allowed near to a computer. [18:26] <iive> ~~ [18:26] <rly> Take 1000 [18:27] <rly> [buffer @ 0x1ce0740] Buffer video input changed from size:897x579 fmt:bgra to size:0x0 fmt:yuv420p [18:27] <rly> [swscaler @ 0x1cf1020] 0x0 -> 897x579 is invalid scaling dimension [18:27] <rly> [swscaler @ 0x1d19740] 0x0 -> 897x289 is invalid scaling dimension [18:27] <rly> [swscaler @ 0x1d19640] 0x0 -> 897x289 is invalid scaling dimension [18:27] <rly> Another prediction that came true. [18:28] <rly> I should become an oracle. [18:28] <rly> Any more excuses as to why it doesn't work? [18:30] <rly> Perhaps you should focus on having less features and for the ones that do ship make sure that it works. [18:31] <TimNich> me/ decides its time to start the weekend [18:31] <TimNich> away/ its the weekend... [18:44] <DelphiWorld> hey all [18:44] <DelphiWorld> finaly my project is compiled now [18:44] <DelphiWorld> but [18:44] <DelphiWorld> how to list ffmpeg actual linked codecs? [18:47] <relaxed> rly: try libav [18:47] <rly> relaxed: don't you ship that too? [18:48] <rly> relaxed: and obligatory: which version? [18:48] <rly> It also appears that all X capture programs stink. [18:48] <rly> E.g. they show large white blocks which are not visible when using it normally. [18:48] <rly> For example when there is scrolling terminal output. [18:54] <relaxed> rly: libav is the ffmpeg fork. libav.org [19:01] <rly> and what is this? [VD_FFMPEG] DRI failure. [19:01] <rly> This is not demonstrating DRI failure. [19:01] <rly> What it should so is output a test case for something which fails. [19:02] <rly> A minimal case at that. [19:02] <rly> No, but someone decided 'let's just scream it failed'. [19:06] <iive> DelphiWorld: -formats, the external codecs should start with lib* [19:06] <rly> relaxed: I built libav, now what? [19:07] <rly> relaxed: i.e. can libav also record? [19:07] <relaxed> rly: yes, libav has avconv which behaves just like ffmpeg. [19:09] <DelphiWorld> iive: thx [19:09] <rly> relaxed: which returns Unknown input format: 'x11grab' [19:09] <rly> relaxed: and yes I had --enable-x11grab [19:51] <RobertNagy> could someone explain the purpose of the sinkbuffer? [19:51] <RobertNagy> aren't frames always pulled in a filter graph? [19:51] <RobertNagy> a buffersink would only be useful if frames were push through the chain? [20:42] <ptrkmj> I would like to use ffmpeg to download bunch of wmv files served via mmsh. I don't want to bother with stream mappings - is there an option to just dump those streams without remuxing? [20:50] <brontosaurusrex> what would i need to do to compile ffplay as well? (osx) [21:08] <cbreak> brontosaurusrex: sdl? :) [21:08] <brontosaurusrex> cbreak: how? [21:08] <cbreak> what do you want ffplay for? Both VLC and mplayer are probably more capable [21:08] <cbreak> what how? [21:09] <brontosaurusrex> mplayer is not really working as it should [21:09] <cbreak> of course it is. [21:09] <brontosaurusrex> cbreak: do you have a bin i can try? [21:09] <cbreak> no. Compile it yourself. [21:09] <brontosaurusrex> i did [21:12] <taqattack> Okay I'm having some trouble with RTMP streaming with FFmpeg [21:12] <taqattack> http://pastebin.com/gVw3EdZ1 [21:16] <Mista_D> What lib is used to decode/encode s302m (BSSD) please? [23:03] <stf> i have a .ts file from a tv-recording which have different audio streams, i want to cut the file by now i try it this way: 'ffmpeg -ss 192 -t 6778 -vcodec copy -acodec copy -map 0.1:0.1 -map 0.3:0.2 -map 0.6:0.3 -i "test.ts" "test_1.ts"' i get the error "Number of stream maps must match number of output streams" < what does this means? [23:07] <stf> hello [23:07] <stf> ? [23:07] <stf> anyone who can help? [23:15] <sokak> Hello everyone. Im trying to experiment to convert a dual audio mkv with subtitles to .avi, to be played on a cheap dvd player (it plays avis, i want to try to see if it can play also double audio avis for fun) I think i messed it up pretty weird, there is the sample output of this ffmpeg -i input.mkv -map 0:0 -map 0:1 -map 0:2 -map 0:3 -map 0:4 -vcodec copy -acodec copy -acodec copy -scodec copy -scodec copy test.avi ht [23:15] <sokak> tp://pastebin.com/Z7c3JVAe [23:16] <iive> stf: counting begins from zero [23:16] <stf> ah [23:17] <stf> live: 0.0 is the video stream, or not? [23:17] <sokak> yes it is [23:17] <iive> stf: yep [23:17] <sokak> (ops sorry x3) [23:18] <stf> live: ehm but where is my failure? [23:18] <stf> do i need to -map 0.0:0.0 too? [23:19] <stf> live: ? [23:19] <iive> stf: you need to map something to the first output stream. [23:20] <stf> okay [23:20] <stf> live: could you give me an example? [23:22] <stf> ah wait i have to map 0.0:0.0 [23:24] <stf> what do i need to do if i only want to have streams 0.0, 0.1, 0.3 and 0.6 to the new file? [23:32] <stf> live: ? [23:33] <sacarasc> It's iive, not live. Double lower case I. [23:34] <stf> ups [23:34] <stf> iive: ? [23:35] <iive> i thought you had it figured out? doesn't it work? [23:36] <iive> btw, i'm not sure if you can copy subtitles into avi. [23:36] <iive> hum.. that was in the sokak, example. [23:36] <sokak> hehe yeah sorry i mixed up :x [23:37] <stf> ehm i want to copy from ts to ts [23:38] <stf> iive: i want to copy from ts to ts [23:42] <iive> so map 0.0:0.0 doesn'twork? [23:44] <stf> This works: "-map 0.0:0.0 -map 0.1:0.1" But do i add: "-map 0.2:0.2" therefore it is worried me [23:44] <stf> ehm no it do not worried me but i do not understand the problem [23:46] <DelphiWorld> iive: aac come by default with ffmpeg? [00:00] --- Sat Apr 21 2012

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