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December 2011

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ffmpeg.log.20111203
by burek 04 Dec '11

04 Dec '11

[00:32] <defaultro> can ffmpeg do the slow motion that twixter does? [05:24] <stev3> hey - im rendering the same file, with the same commandline options on both windows and ubunutu, but the linux version returns a file much larger - im thinking they may have different default values for options im not specifying maybe? [05:24] <stev3> ffmpeg -i test.avi -f flv -ab 128k -ar 44100 -b 800k test.flv [10:02] <sjuxax> Hello. I have: ffmpeg -i 00001.MTS -threads 0 -s 1024x576 -c:v wmv2 -b:v 3000k -c:a libmp3lame -aq 0 -y out.wmv, but I get standard incorrect parameters error when I attempt to encode. Pasting full output momentarily... [10:03] <sjuxax> http://dpaste.com/665337/ [10:04] <sjuxax> DEVSD wmv2 Windows Media Video 8, so the codec support is present [10:08] <relaxed> sjuxax: remove -threads 0 [10:09] <sjuxax> relaxed: Thanks. [10:09] <sjuxax> That works now. [10:11] <sjuxax> Is there a reason ffmpeg wants to error there when it could just ignore the request for automatic threading, since it's not supported? I know you could say that you don't want someone to mistakenly assume that the computer is automatically deciding how many threads to use, but that doesn't seem like it's worth the cost. It seems ffmpeg is much stauncher than everything else and that leads to a lot of errors with its usage (parameter order, for instance, i [10:24] <uclinux> is it possible to fade a watermark/image in? [10:29] <dango> what is the difference between stereo and joint stereo [10:32] <relaxed> uclinux: yes, look at the overlay filter in the man page or website. [10:33] <uclinux> okay cool, I can get the whole video to fade in and out but I am having trouble getting just the watermark to fade [10:33] <relaxed> oh, hmm [10:34] <uclinux> i think the overlay filter will fade the whole video...correct? [10:36] <relaxed> I'm not sure what you mean. What are you doing? [10:37] <uclinux> video plays, water mark fades in then fades out, [10:37] <uclinux> i can do it with avisynth [10:38] <uclinux> but i would much rather approach this problem with ffmpeg for a number of reasons [10:39] <relaxed> the filter system is powerful but I'm not sure how do that [10:40] <uclinux> yup, pretty much the consensus. theres a piece of c code floating around to do it but it was for an older version of ffmpeg. [10:40] <uclinux> pip.c [10:42] <relaxed> there's the fade filter [10:45] <uclinux> I'm having trouble applying the fade filter to "only" the watermark [10:48] <relaxed> uclinux: It might be something like: movie=logo.png [logo]; [in][logo] overlay=10:main_h-overlay_h-10,fade=.... [out] [10:49] <uclinux> thats like exactly what i'm using. I am fairly familiar with chains [10:49] <uclinux> relaxed: it actually applies the fade to the clip, not "only" the watermark [10:51] <uclinux> relaxed: its a tricky one, this would save like 200 man hours probably [10:52] <relaxed> did you try movie=logo.png,fade=... [logo]; [in][logo]... [10:54] <uclinux> relaxed: When I do that I get no image overlay [10:54] <uclinux> http://pastebin.com/ttWrimHy is what i tried [10:55] <mshadle_> I am trying to convert some really old syntax/ffmpeg commands and such that someone developed a while ago. bounty is available for help. :p http://pastebin.com/DpjqLVFd [10:56] <uclinux> mshadle_: i can do this after I finished what I'm working on [10:57] <uclinux> I'm pretty booked at the moment lol [10:57] <mshadle_> awesome! the vhook pieceis the trickiest i think. and i might fall asleep. i left my email in the pastebin though. so if i don't reply, email me and let me know :) [10:58] <uclinux> yeah, I'm dealing with a vhook/watermark issue right now [10:58] <uclinux> idk if there is a solution to mine [10:58] <mshadle_> cool [10:58] <mshadle_> i *think* this i a simple one [10:58] <uclinux> i'll shoot you an email in the AM [10:59] <mshadle_> thanks. these are 8 core webservers. all they do is nginx/php and encoding when something is in the queue. so they can be given plenty of CPU resources [11:00] <uclinux> relaxed: thanks for help...I think i will end up with a custom solution :/ [11:04] <relaxed> uclinux: Use two commands and on the first one loop the input of the png adding the fade with the output of %06d.png. Then use %06d.png as your overlay input. [11:05] <relaxed> assuming you can use %06d.png as the input [12:25] <burek> dango, the diff is compression only [12:25] <burek> no loss of quality [12:25] <dango> huh? all lossy compression = loss of quality [12:26] <JEEB> yes, to a level :) [12:26] <dango> what is the difference between stereo and joint stereo? and why is it only option in mp3 [12:27] <burek> http://www.richardfarrar.com/what-is-joint-stereo/ [12:27] <burek> there is also lossless compression, called gzip, you know? [12:28] <JEEB> of course [12:29] <JEEB> just like ffvhuff, x264's H.264's lossless profile and such for video, and FLAC, wavpack and such for audio [12:30] <dango> is mp3 the only lossy audio codec that use joint stereo [12:31] <JEEB> it can be implemented in many formats, but it just means that ffmpeg / whatever only has a specific option for it [12:31] <burek> dango, why do you not like joint stereo? [12:32] <JEEB> not to mention that joint stereo can mean a lot in the ways of implementation >_> [12:32] <dango> burek no idea [12:32] <burek> dango, did you read that url above I left for you? [12:32] <dango> i never seen joint stereo in AAC or vorbis wma [12:32] <dango> yes i did [12:33] <JEEB> http://en.wikipedia.org/wiki/Joint_%28audio_engineering%29#Joint_stereo [12:33] <JEEB> "Two forms are described here, both of which are implemented in various ways with different codecs, such as MP3, AAC and Ogg Vorbis." [12:33] <burek> so, why are you asking such questions then? [12:33] <JEEB> just that there are no options for it in the stuff you see doesn't mean the coding techniques aren't used elsewhere [12:34] <JEEB> but I'd guess that most encoders use these techniques automatically as they become more needed [12:34] <JEEB> and/or from the beginning depending on the format/encoder -- if it gives a perceivable compression benefit while not affecting audio quality [12:34] <dango> what is better aac-lc or aac-he [12:35] <burek> wav is the best [12:35] <JEEB> The latter uses compression techniques that are good for low bitrates, but you can end up with a worse off end result if you use more bitrate than where it is needed [12:36] <burek> JEEB, why would you spend more money than you actually need to? :) [12:36] <JEEB> which is why most encoders that can do both set the mode depending on the bitrate you're encoding with [12:37] <JEEB> burek, as far as I've heard the HE-AAC stuff specifically kills some information that would otherwise be left around. Not to mention that I usually encode with 128-192kbps regions anyway [12:37] <dango> why is there version 2 of aac-he [12:37] <JEEB> I don't need the extra compression [12:37] <JEEB> if you can give me a compression algorithm that isn't limited to lower bitrate scenarios because of the herp derp it does to the stream, I'm fine with that [12:37] <burek> dango, read the wikipedia about AAC, they even have a picture for you, regarding that question [12:38] <JEEB> HE-AAC, though, seems to be exactly for low bitrate things and isn't meant for relatively accurate compression [12:38] <dango> and what about AAC-LTP? why does so many players have problem with this [12:38] <JEEB> which is also the reason why encoders only stick their HE-AAC modes on with lower (<=64kbps or so) bitrates [12:38] <burek> aac+ is just like mp3, made for best quality with as low bitrates as possible [12:39] <dango> i don't even see aac-LTP in wiki [12:40] <JEEB> burek, tl;dr if HE-AAC is always better than LC-AAC, why do encoders switch their algorithms depending on the scenario :) [12:40] <burek> dango [12:40] <JEEB> that's the only thing that icks me in the whole LC/HE-AAC thing [12:41] <burek> JEEB, because they are not updated ^^ [12:42] <JEEB> sigh... [12:42] <burek> I mean, if your goal is quality, regardless of the size (bitrate), you wont use AAC in the first place, you'll use flac, right? [12:42] <JEEB> no [12:43] <JEEB> I'd use wavpack for lossless as it has specified channel mappings [12:43] <JEEB> but that's the "lossless" area [12:43] <JEEB> which is usually 800kbps+ [12:43] <JEEB> I'm not comfortable with using 800kbps+ on audio, and thus we enter the roughly 128-256kbps zone. [12:44] <burek> so, you do care about size/bitrate? [12:44] <dango> is wavpack even playable? [12:45] <JEEB> of course, libavcodec's decoder was fixed by uau and friends [12:45] <JEEB> I care about size but I don't care to use algorithms that specifically lower the quality of audio -- like HE-AAC does. Not to mention that my comfort zone is over the rates where HE-AAC is useful [12:45] <dango> isn't monkeyaudio or flac better than wavpack [12:46] <JEEB> compression-wise monkey's audio is better than wavpack IIRC, but FLAC is worse off. Also, monkey's audio is a mess in many ways which is why it's not on the selection list to begin with. [12:46] <burek> well, as I said before, I tested aac+ on classical music and the sound was perfect, so I don't see that part of "lower the quality of audio" [12:46] <dango> why you say it's a mess [12:46] <burek> and classical music has really wide range/spectar of freq to encode [12:46] <JEEB> dango, non-open format that is not spec'd [12:46] <dango> jeeb who cares if it's not open or not [12:47] <JEEB> because, you know, you want to know if the decoders etc. are doing it right? [12:47] <dango> and what does not-open have to do with being a mess [12:47] <JEEB> because the spec is not open [12:47] <dango> jeeb what about ac3/dts [12:47] <JEEB> and lacks info, it is a mess -- let's just begin with channel mappings for example [12:47] <burek> use 7zip on a wav file and that's it ^^ [12:47] <JEEB> AC3 and DTS are both old [12:48] <dango> jeeb are ac3/dts open? [12:48] <JEEB> I don't think so, but the specs at least exist and open source developers to a level have acquired them [12:48] <JEEB> not to mention that stuff like audio channel mappings have been specified [12:49] <JEEB> thus, albeit f.ex. DTS is complete shit compression-wise [12:49] <JEEB> it is still less of a mess than monkey's audio [12:49] <JEEB> both are terrible formats in different classes [12:49] <dango> why if flac more popular than wavpack then [12:50] <dango> is [12:50] <JEEB> it's a viable alternative, and the difference between wavpack and it are small [12:50] <dango> and is wavpack open? [12:50] <JEEB> yes, it is [12:51] <JEEB> the problems only come up with around 5.1 and over, where the channel mappings aren't specified at FLAC, so for most music it's just fine [12:51] <dango> jeeb best lossless compression is actually this one: http://www.lossless-audio.com/comparison.htm [12:52] <JEEB> sure, there are lossless compression formats that compress better than flac/wavpack [12:52] <JEEB> I'm just not sure I want to use them :D [12:52] <JEEB> you can just check out the 'enctime' and 'dectime' areas [12:53] <JEEB> there are spots that are way over my comfort level :) [12:53] <dango> jeeb true [13:00] <burek> there is always spide-quality-size triangle of restrictions [13:01] <burek> you can influence at most the two of them at anytime [13:01] <burek> the third will simply be the result of your choice for those two [13:11] <burek> spide :)))))))) speed* :) [14:30] <brontosaurusrex> any decent assumptions about the multithreading status of prores decoder/encoder ? [15:03] <cbsrobot> brontosaurusrex: patch welcome :) [15:40] <coony> hi [15:41] <coony> it possibol to convert un file .mov on other format having the same quality [15:42] <JEEB> -acodec -vcodec copy [15:42] <JEEB> :P [15:43] <coony> ok [15:43] <coony> but it's important have a svn ffmpeg? [16:05] <cbsrobot> coony: ... git ffmpeg [16:27] <symtab> hi [16:27] <symtab> what get i use to get the same results as -vpre baseline [16:27] <symtab> http://pastebin.com/QxW7Pthe [16:27] <symtab> ? [16:28] <symtab> i need to get this output: [16:28] <symtab> http://pastebin.com/jRTBQwWB [16:28] <symtab> Constrained Baseline level 1.3 [16:45] <burek> symtab, use -x264opts [16:49] <burek> also, read this: http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Profiles [17:06] <coony> how can add different file .mov? [17:20] <burek> coony? [17:21] <coony> so, i have a more single file.mov and i want unity on one file.avi [17:35] <burek> can you use google translate? [19:55] <zorbi> I have a question about streaming live transcoded content in mp4 format to an iPad. [19:56] <zorbi> I am trying to transcode an MPEG2 file into MPEG4 directly to stdout and send it through a web server into my waiting ipad [19:56] <zorbi> alas, mp4 is reporting not to support non-seekable formats (as can be seen in the implementation of movenc.c) [19:56] <zorbi> Apparently from the need to update the MOOV atom. [19:57] <zorbi> Does that mean one cannot produce live streaming (for content not pre-generated previously into a file)? [19:57] <zorbi> For an mpeg4 container format, that is. [20:59] <funyun> hi. is it possible to convert a mkv with DTS to AC3 with 5.1 surround sound at 640kbps with ffmpeg while keeping the video untouched? [21:01] <hi117> ffmpeg -i file.mkv -vcodec copy -acodec ac3 out.mkv < does this work? [21:02] <funyun> hi117: thanks. i'll try [21:06] <burek> zorbi you cant stream live video to iPad [21:06] <burek> buy a normal device :) [21:07] <burek> funyun, also please copy/paste your command + output to pastebin.com [21:07] <burek> so we can see what the media looks like [21:11] <JEEB> burek, uhh what [21:11] <JEEB> didn't ipad support mpeg-ts over http just fine? [21:11] <JEEB> which can be streamed in a few ways [21:12] <funyun> burek: http://pastebin.com/SSR42bgF [21:14] <burek> JEEB, i think not [21:15] <burek> if that was the case, people wouldn't bother with "progressive http streaming" [21:15] <burek> and stuff [21:15] <JEEB> burek, note: it does [21:16] <JEEB> progressive http streaming is just easier to set up :P [21:16] <JEEB> as you don't need something to be encoding it in real time [21:16] <JEEB> :P [21:16] <JEEB> http://en.wikipedia.org/wiki/HTTP_Live_Streaming [21:16] <burek> ok then [21:16] <burek> i was wrong :) [21:16] <burek> i don't use that crap anyway :) [21:16] <funyun> burek: i get that error when i try [21:17] <JEEB> burek, it's the easiest way of streaming to mobile devices to set up :P And is supported by both iOS and Android now, so herp derp. Would you rather have it, or some herpy derpy to set up rtsp stream? [21:17] <JEEB> :P [21:18] <hi117> id rather setup and manually code a way to UDP stream over the cell network [21:18] <burek> funyun, add -sn [21:18] <JEEB> suit yourself [21:19] <burek> or -scodec copy [21:19] <JEEB> I know http isn't the best way [21:19] <JEEB> but herp derp [21:19] <JEEB> you can relatively easily use VLC to do that thing [21:20] <burek> ok, i believe you :) [21:48] <funyun> burek: worked. thanks [22:20] <tiim> can anyone suggest a channel for general 'problems with video data' questions? [22:30] <burek> tiim, what kind of questions do you have in mind [00:00] --- Sun Dec 4 2011

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ffmpeg-devel.log.20111202
by burek 03 Dec '11

03 Dec '11

[00:03] <michaelni> Daemon404, :) [00:27] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * r8b08f81949 10ffmpeg/libavcodec/bmp.c: Support transparency in 32bit bmp files. [01:00] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r0dd704dbd4 10ffmpeg/libavcodec/vble.c: [01:00] <CIA-41> ffmpeg: vble: remove unused variable len. [01:00] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:00] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:00] <CIA-41> ffmpeg: 03Derek Buitenhuis 07master * re46abbcf77 10ffmpeg/libavcodec/vble.c: [01:00] <CIA-41> ffmpeg: vble: merge len and val arrays [01:00] <CIA-41> ffmpeg: There's no reason to use two arrays for this. [01:00] <CIA-41> ffmpeg: Based off commit 2fea60c60084c4e70d7cef128ea3bca5690ce465 [01:00] <CIA-41> ffmpeg: to FFmpeg by Michael Niedermayer. [01:00] <CIA-41> ffmpeg: Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> [01:00] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:00] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r317ea97bff 10ffmpeg/libavcodec/vble.c: [01:00] <CIA-41> ffmpeg: vble: remove flags copy, its not used in any speed relevant code. [01:00] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:00] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [01:01] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:01] <CIA-41> ffmpeg: 03Martin Storsjö 07master * r6af354436c 10ffmpeg/libavformat/md5proto.c: [01:01] <CIA-41> ffmpeg: md5proto: Remove the get_file_handle function [01:01] <CIA-41> ffmpeg: The private data pointer isn't a file handle, this protocol [01:01] <CIA-41> ffmpeg: doesn't have any file handle to return. [01:01] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:01] <CIA-41> ffmpeg: 03Martin Storsjö 07master * r02490bf358 10ffmpeg/libavformat/rtmpproto.c: [01:01] <CIA-41> ffmpeg: rtmp: Clean up properly if the handshake failed [01:01] <CIA-41> ffmpeg: This prevents memory leaks if this function returns an error. [01:01] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [01:01] <CIA-41> ffmpeg: 03Reimar Döffinger 07master * rf28070a123 10ffmpeg/ (5 files in 3 dirs): [01:01] <CIA-41> ffmpeg: Add PlayStation Portable PMP format demuxer [01:01] <CIA-41> ffmpeg: Not yet complete, for demuxing AAC the AAC header must be generated [01:01] <CIA-41> ffmpeg: manually. [01:01] <CIA-41> ffmpeg: Possibly the decoder could accept the header as extradata to simplify [01:01] <CIA-41> (78 lines omitted) [01:09] <Daemon404> michaelni, the Libav folks asked me to add a FATE test for VBLE, so I did. I'm assuming FFmpeg wants it too. [01:10] <Daemon404> Who handle's FFmpeg's FATE rsync server? [01:17] <michaelni> Daemon404, flowers-partial-2MB.avi uploaded to our rsync server, thanks [01:17] <Daemon404> ok, cool. [01:18] <ubitux> AVFilterBufferRef has data/linesize/format which are common with AVFilterBuffer [01:18] <ubitux> and AVFilterBuffer has w and h which are common with AVFilterBufferRefVideoProps [01:19] <ubitux> but AVFilterBufferRefVideoProps also has data/linesize/format [01:19] <ubitux> ._. [01:19] <michaelni> a buffer has its width & height and allocated space [01:20] <michaelni> a reference in that buffer may have different width/height/linesize, it may be just a part of the whole [01:21] <ubitux> this what the "crop" comment is about i guess [01:27] <michaelni> and the ref is common stuff with video/audio in seperate structs [03:06] <ubitux> can we assume av_read_frame() will return a packet containing exactly one subtitle? [03:07] <michaelni> normally the demuxer/parser should split things in sane units (frames) [03:09] <ubitux> according to the doxy we have 1 pkt = 1 frame for video, the audio case is a bit special, but for subtitle nothing is specified [03:09] <michaelni> Zeranoe, btw, i totally forgot, about the pthread/w32thread default thing, if you received no reply from wbs then iam happy to apply a patch with whatever change you want [03:09] <ubitux> since a pkt is associated with a pts, i guess we will have 1 pkt = 1 subtitle [03:11] <michaelni> I think that with the current lack of full subs support the read_frame should be adapted to what makes most sense for a better and more complete implementation [03:12] <ubitux> ok [03:12] <ubitux> i wonder how we will deal with karaoke stuff& [03:15] <ubitux> i also wonder if packetizing the subtitles will make sense with libass [03:16] <ubitux> ah, ass_process_chunk, one event in matroska format [03:17] <ubitux> so if we demux an external ass file, we will have to "mux" it before feeding it to the subtitles chain [03:37] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * rd7fc100694 10ffmpeg/libavcodec/tiffenc.c: Support tiff RGBA encoding. [03:37] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * r40c36196d5 10ffmpeg/libavcodec/libopenjpegenc.c: Support yuva420p encoding via libopenjpeg. [03:37] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * rbba8975a9c 10ffmpeg/libavcodec/h264_ps.c: [03:37] <CIA-41> ffmpeg: Be less verbose when skipping RBSP data in H264 PPS. [03:37] <CIA-41> ffmpeg: Fixes ticket #693. [03:37] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * rfab5973fa5 10ffmpeg/libavcodec/mpeg4videodec.c: [03:37] <CIA-41> ffmpeg: Be less verbose when skipping MPEG-4 ASP frames. [03:37] <CIA-41> ffmpeg: Fixes ticket #560. [04:08] <Zeranoe> michaelni: Thank you, I haven't had much time as of late so I won't be able to get a patch to you for a while but I will work on it in my free time. Was there any objections to my suggested method? [04:12] <michaelni> i got no reply sadly [04:16] <Daemon404> interesting that vble is actually faster when build with clang/llvm [04:16] <Daemon404> built* [04:17] <Daemon404> too bad debian's version of GNU gold is busted as shit [04:17] <Daemon404> so i cant try out LTO [11:43] <vega13> If I make a custom avcodeccontext and my mpeg4 packets that I want to decode have extradata, how do I go about setting my avcodeccontext? [11:44] <vega13> if I decode it with my avcodeccontext I get the warning: first frame is no keyframe for every frame [11:44] <vega13> when I load a non custom avcodecctx with pCodecCtx=pFormatCtx->streams[videoStream]->codec; [11:44] <vega13> I don't get this error and frames are complete and don't look messed up [11:44] <vega13> on inspection of difference in data between the two avcodeccontexts it seems the one that works contains extradata.. [11:47] <vega13> What am i doing wrong, it's not possible to use a custom avcodeccontext on a stream like this? [14:42] <CIA-41> ffmpeg: 03Peter Ross 07master * r1fd6924320 10ffmpeg/libavcodec/pictordec.c: [14:42] <CIA-41> ffmpeg: pictordec: support uncompressed images [14:42] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [19:02] <Node_164> mjpeg: unsupported coding type (ce)? thx.. [19:03] <gnafu> Node_164: You're welcome. [19:31] <Compn> Node_164 : is it all different unsupported coding types ? [19:31] <Compn> like each frame reports a diff coding type ? [19:31] <Compn> if so, its probably not mjpeg [22:27] <ubitux> would it make sense to add a AVSubtitle in the AVFilterBuffer* struct? [22:27] <ubitux> or should i duplicate the "interesting" content, just like it's done with the a/v data? [22:31] <ubitux> it would be equivalent to put a AVFrame in the filterchain (i wonder why it's not the case actually, maybe because it has too much unrelated information in it?) [22:55] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r096701d9b2 10ffmpeg/doc/examples/muxing.c: [22:55] <CIA-41> ffmpeg: muxing example: set encoder defaults [22:55] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:59] <michaelni> ubitux, i think stefano wanted to keep the lavc/lavfi APIs not too entangled [23:01] <ubitux> ok [23:01] <ubitux> you need various things of lavc anyway [23:01] <ubitux> in the source filter at least [23:01] <ubitux> but well ok [23:02] <ubitux> it sounds overkill to duplicate the AVSubtitles though (even if it's a subset of the fields) [23:03] <michaelni> ive no oppinion on this either way, i just remember sastes opposition to passing a AVFrame as param to a lavfi func [23:06] <ubitux> may it be related to ABI issues too? [23:06] <michaelni> could be [23:07] <michaelni> also AVFRame is alot bigger tham AVSubtitles [23:33] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rcf257f30b9 10ffmpeg/doc/examples/muxing.c: [23:33] <CIA-41> ffmpeg: doc/example/muxing: fix video timestamps [23:33] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Sat Dec 3 2011

1 0

ffmpeg.log.20111202
by burek 03 Dec '11

03 Dec '11

[00:52] <shysoft> Is there a trick to using debugging symbols with ffmpeg? [00:52] <shysoft> I'm having trouble stepping into the libav* libraries. [01:03] <teratorn> shysoft: yeah there is kind of a trick [01:04] <teratorn> shysoft: this is what I use: --enable-debug --disable-stripping --disable-asm --disable-optimizations [01:04] <shysoft> teratorn: thanks. I'll try that. [01:42] <pasteeater> relaxed: you're right. trampst...i mean timestamp has been depreciated for creation_time. where the hell have i been? [02:20] <abstrakt> how do I know if this avi is 1080i or 1080p [02:20] <abstrakt> here's the output of ffmpeg -i [02:20] <abstrakt> Stream #0.0: Video: h264 (Main), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9] [02:51] <SMJ> http://paste.servut.us/plain/5v0z can someone explain what operation is not permitted? [02:55] <SMJ> ah... just when I thought Ubuntu had a decently recent version of ffmpeg, it's not even at the same major version... [03:36] <JackD> Good Night/Day/Morning [03:37] <JackD> How can I get video from my camera and create a jpg every second in a folder? [03:38] <JackD> I had tried ffmpeg -f vfwcap -s 640x480 -r 1 -i 0 output%d.jpg but it make the jpg change every second not be created every second [03:38] <JackD> In true it create a big jpeg sequence equal the framerate of my camera. [03:44] <kanzure> jackd: %03d maybe? [05:24] <Whitewizard> Moar problems! when i seek to a certain point in a video the audio becomes desynced! using -ss 00:10:42 [05:34] <rynop> im using ffmpeg to convert a glc-capture format file to libx264. Every preset i use seems to create a video that "dark" - like not as bright as the original. Total ffmpeg noob here, is this a common problem? [05:36] <rynop> http://pastebin.com/82XgZ0rc is an example of what im doing that is producing a dark video. [05:46] <pasteeater> Whitewizard: option placement is important. did you use it as an input or output option? [05:46] <Whitewizard> before the -i [05:46] <pasteeater> use a pastebin site to show your ffmpeg command and the complete console output [05:47] <pasteeater> rynop: same for you [05:47] <Whitewizard> like its weird.. i got 5 seconds of video, that i want, but an extra 5 seconds of audio before it [05:47] <rynop> pasteeater, i did post my pastebin.. [05:47] <hi117> it still had pipes included [05:48] <rynop> pasteeater, or u want full output [05:48] <Whitewizard> http://pastebin.com/gub5LcZ1 [05:49] <pasteeater> rynop: yes, include command and complete output [05:50] <pasteeater> Whitewizard: where's your command? [05:50] <Whitewizard> lets try this again.. [05:51] <Whitewizard> http://pastebin.com/7RcnQj9W [05:51] <rynop> ah sry. http://pastebin.com/B9QwnVnC and also tried http://pastebin.com/Y2QSyBdL . Same "dark" results. [05:51] <Whitewizard> yes, i spelt converter wrong, but its too late to change [06:00] <pasteeater> rynop: does it look normal with ffplay? ffplay video.mp4 [06:03] <rynop> pasteeater, nope still darker than the original (glc-play 1092-3.glc). This is a recording of a video game (starcraft 2), its a pretty "dark" game to begin with, not sure if that matters when the encoder is compressing. [06:03] <Whitewizard> the effect seems proportionate to how many frames were dropped.. [06:03] <pasteeater> i'm not familiar with glc-play. there might be some colorspace conversion going on. [06:04] <rynop> pasteeater, is there some sort of "dark_mask" option in ffmpeg? reading http://www.mplayerhq.hu/DOCS/tech/encoding-tips.txt seems like my problem is simliar to this menucoder "dark_mask" [06:04] <pasteeater> Whitewizard: did you try moving -ss after -i input? [06:06] <Whitewizard> it jsut forever encodes at 0fps [06:07] <pasteeater> wait for it. that method is slower, but it will help rule out the cause. [06:08] <pasteeater> it appears to be sitting there, but it's acrually decoding until it reaches your -ss value. [06:08] <pasteeater> rynop: i'm not sure [06:08] <rynop> pasteeater, I did just find a -lumi_mask option in ffmpeg, messign w/ that [06:09] <pasteeater> you could try: ffmpeg -i - -vframes 1 output.png [06:10] <pasteeater> to test if rgb looks normal. [06:10] <rynop> frames X - where X is the number of the frame in the video? higher means later in vid? [06:11] <pasteeater> -vframes = number of frames to output [06:11] <pasteeater> 1 for 1 image. [06:11] <rynop> ah ok. early in the vid (the loading screen) the brightness is fine. its just in game. So i realy need to get some frames from the middle of my vid to compare [06:12] <pasteeater> you should also try a more recent ffmpeg: http://ubuntuforums.org/showthread.php?t=786095 [06:12] <rynop> pasteeater, ok good idea. [06:15] <Whitewizard> that seemed to haveworked [06:15] <Whitewizard> odd.. [06:15] <Whitewizard> thnaks [06:15] <Whitewizard> thanks [06:16] <pasteeater> you might be able to speed up decoding with -threads 4 or some appropriate number as an input option. [06:17] <pasteeater> rynop: you could add -ss 23 if you want ffmpeg to skip the first 23 seconds so you have time to get from menu to game [06:19] <pasteeater> (as an output option) [06:25] <rynop> pasteeater, -lumi_mask -dark_mask seem to be helping. Just messing around with values to get teh right setting, not documented well [06:26] <rynop> so i lied its not helping, was comparing different frames. [06:32] <pasteeater> did you try the png test? did you try a newer ffmpeg? [06:34] <rynop> no to both. not yet. didnt feel like going thru teh ffmpeg compile [06:35] <rynop> watned to try some of these other options before getting into compiling. looked for a launchpad ffmpeg repo - found some but they are way back level, not sure whats up with that. [07:48] <alyawn> I'm attempting to write a simple filter and every frame gets dropped. Is there a way to find out why ffmpeg is dropping my frames? [08:22] <relaxed> alyawn: I think the select filter will do that. [08:24] <relaxed> oh, nm [08:25] <relaxed> alyawn: loglevel may help [08:52] <cryptopsy> yesterday i couldn't convert vob to mp3 because a libmp3lame was missing but i'm able to convert avi to mp3 no problems [09:03] <relaxed> cryptopsy: I recall. [09:03] <cryptopsy> yes, so what's libmp3lame? [09:03] <cryptopsy> why do i need it? [09:03] <cryptopsy> exactly [09:04] <cryptopsy> relaxed: actually in some circumstances i can produce an output file (mp3) [09:04] <cryptopsy> that is non-0 size [09:04] <cryptopsy> for example my vob is 1giga, the output file is a 220mega mp3 [09:04] <relaxed> are you copying? [09:04] <cryptopsy> -acodec copy , if that's what you mean [09:05] <cryptopsy> i just want to extract the audio [09:06] <relaxed> ffmpeg -i input.vob output.flac [09:08] <cryptopsy> relaxed: 0 size [09:08] <cryptopsy> -rw-r--r-- 1 root root 1.0G Dec 1 00:56 vts_01_1.vob [09:09] <cryptopsy> -rw-r--r-- 1 root root 0 Dec 2 04:07 test.flac [09:10] <relaxed> can you pastebin the command and output? [09:10] <cryptopsy> can i just paste to the channel? [09:10] <cryptopsy> i'll start pastebining when others are talking [09:11] <relaxed> no [09:11] <cryptopsy> k what pastebin do i use? [09:11] <cryptopsy> oh it's back up [09:11] <cryptopsy> http://pastebin.com/VHi08duX [09:12] <relaxed> Did you compile ffmpeg? [09:13] <cryptopsy> sure [09:13] <relaxed> why is everything disabled? [09:14] <cryptopsy> 0.8.6(04:02:24 12/02/11)(mp3 -3dnow -3dnowext -X -aac -aacplus -alsa -altivec -amr -avx -bindist -bzip2 -celt -cpudetection -debug -dirac -doc -encode -faac -frei0r -gsm -hardcoded-tables -ieee1394 -jack -jpeg2k -mmx -mmxext -network -oss -pic -qt-faststart -rtmp -schroedinger -sdl -speex -ssse3 -static-libs -test -theora -threads -truetype -v4l -vaapi -vdpau -vorbis -vpx -x264 -xvid -zlib) [09:15] <relaxed> I'm 12 and what is this? [09:15] <cryptopsy> the compile options for ffmpeg [09:15] <cryptopsy> so it is compiled with mp3 support, but not 3dnow, etc [09:16] <cryptopsy> + + mp3 : Add support for reading mp3 files [09:16] <relaxed> OH! this is gentoo? [09:16] <cryptopsy> yes [09:16] <cryptopsy> sure [09:17] <relaxed> good because the way you're building it is completely broken [09:17] <cryptopsy> ah okay that would explain it [09:17] <cryptopsy> i updated the repo and ffmpeg upgraded to a new version [09:17] <cryptopsy> everything is fucked and busted in the tree nowadays, i don't know how it happened [09:17] <cryptopsy> how were you able to tell? [09:18] <relaxed> Because on pastebin I could see all the --disable-* in your output. Plus you can't seem to encode simple audio. [09:19] <cryptopsy> unfortunatley i don't have the .avi i keep talking about [09:22] <relaxed> lol, wtf is "-encode"? Do you have encoding disabled? [09:23] <cryptopsy> - - encode : Adds support for encoding of audio or video files [09:23] <cryptopsy> new flag [09:23] <cryptopsy> it was builtin in the old version [09:23] <cryptopsy> gentoo doesn't warn users about these changes [09:23] <cryptopsy> thanks [09:24] <relaxed> most of those use flags are wrong. [09:24] <cryptopsy> i'll show you the list [09:25] <relaxed> pastebin [09:25] <cryptopsy> i'm getting sick of pastebin, now it's requiring people to break captchas [09:27] <cryptopsy> http://pastebin.com/zFSt6SnW [09:29] <cryptopsy> ompldr.org/vYmpzaA [09:30] <cryptopsy> the second link works, lists all the use flags and their descriptions [09:31] <relaxed> add support for all of these http://pastebin.com/WXMzEc4h [09:32] <cryptopsy> we could go to mp3 [09:32] <cryptopsy> pm* [09:33] <cryptopsy> pastebin are not loading for me [09:34] <cryptopsy> but with encode seems to work [09:34] <cryptopsy> slowly [09:34] <cryptopsy> it was much faster for avi [10:01] <Guest26588> hi there anyone alive? [10:06] <giany> hum? [10:06] <giany> any idea what this means : "[mov,mp4,m4a,3gp,3g2,mj2 @ 0x9e71340] moov atom not found" [10:06] <giany> when checking a converted movie? [13:58] <vega13> what does a videoparser do exactly? [14:15] <vega13> never mind [14:22] <zmitya> hi Gents [14:22] <zmitya> yesterday I have posted this: https://lists.ffmpeg.org/pipermail/ffmpeg-user/2011-December/003445.html [14:22] <zmitya> do you guys have any comments on this ? or did I post to a wrong place ? [14:22] <aacffmpeg> hi, question: i have an ios application that grabs audio and converts it to AAC. What I want is to feed FFMPEG with these AAC packages and send them via ffmpeg to my MPEG-TS stream. How can I go about this? Any references? [15:18] <id10t> 'lo all [15:20] <id10t> anyone know if there is a way to save a movie from the visualizations that ffplay produces for a mp3 file? IOW, play mp3 file, dump visualizations to a movie file w/ mp3 file as audio track [15:36] <simon3z> hi all I have a mpeg-ts stream (from dvb-t) that has some damaged parts and some non-monotonic dts... is there any way to let ffmpeg fix this problems copying the streams to a mpeg-ps file? (using "-vcodec copy -acodec copy -f vob" the output has the a/v streams not synchronized) [16:58] <cryptopsy> i convert .vob to mp3 with ffmpeg -i my.vob -ac 2 -ab 320k out.mp3 , and i can see it encode with bitrate=320k, but 'file out.mp3' shows [16:58] <cryptopsy> Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 56 kbps, 48 kHz, Stereo [16:58] <cryptopsy> is that a bitrate of 56kbps i see there? [17:07] <teratorn> I have a C/ffmpeg program that does a basic transcoding job... but I'm having trouble with the resulting file. input and output are both webm. ffprobe shows "1000k" fps for the input, and "1kfps" for the output file - and the output file plays as fast as the cpu can render it... any clue what could be wrong? [18:14] <beandog> hey guys which ML would be the best one to send a feature request to [18:16] <cryptopsy> ? [18:16] <beandog> ! [18:17] <beandog> specifically, wanted to post the idea of getting qt-faststart in ffmpeg, along with an option to fix the metadata automatically at end of encoding with an -http option or something to make it stream-happy [18:34] <twid> when i try to convert file using ffmpeg i get "mjpeg: unsupported coding type (cf)"... what does it mean by that? thx.... [18:55] <cryptopsy> how can i confirm the bitrate of an mp3 ffmpeg produced? [18:58] <twid> cryptopsy : You can use ffprobe to check bitrate of stream.... [19:00] <simon3z> hi all I have a mpeg-ts stream (from dvb-t) that has some damaged parts and some non-monotonic dts... is there any way to let ffmpeg fix this problems copying the streams to a mpeg-ps file? (using "-vcodec copy -acodec copy -f vob" the output has the a/v streams not synchronized) [19:03] <cryptopsy> twid: thanks [19:04] <twid> NO prb.. [19:06] <cryptopsy> it's telling me 320k bit rate but file seems to say 56k [19:06] <cryptopsy> file 2.mp3 [19:06] <cryptopsy> 2.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 56 kbps, 48 kHz, Stereo [19:06] <cryptopsy> indeed ffmpeg is reporting correctly "bitrate: 320 kb/s" [19:06] <cryptopsy> what is that file is telling me? [19:20] <JackD> Hello [19:23] <JackD> Can someone help me with a simple task with ffmpeg cli? [19:24] <pasteeater> beandog: you can add a feature request to the bug tracker [19:24] <pasteeater> https://ffmpeg.org/trac/ffmpeg/ [19:24] <beandog> pasteeater: kk thanks [19:25] <pasteeater> type: enhancement, priority: wish [19:28] <JackD> weird nicks ... [19:32] <JackD> I think I mdo not need to past it on pastebin - is is small - but if I really need... Well I want to captire a frame every second and put the image in a jpg file. I tried "ffmpeg -f vfwcap -s 640x480 -r 1 -i 0 output%d.jpg" but it only refresh each second it do not create the jpg each second. [19:34] <JackD> How can I do it, or where I can find info about it? I read some docs but I do not went so far. [19:35] <beandog> pasteeater: okay [19:39] <teratorn> anyone have a clue why ffmpeg might produce a file which it is then unable to process? http://codepad.org/ijvbPYMc [19:42] <JackD> teratorn: have an error about your framerate in the output, have you figured it? [19:44] <teratorn> JackD: I don't know. I think it's a bug that ffmpeg produces a file that it chokes on [19:50] <pasteeater> teratorn: i can duplicate the issue if I add -r 0.08 [19:51] <pasteeater> also, sameq does not mean same quality, and you shouldn't be using it in this case as vp8 and h264 probably don't use the same quantizer scale [19:52] <pasteeater> ffmpeg -h | grep sameq: -sameq use same quantizer as source (implies VBR) [19:52] <teratorn> I see [19:53] <pasteeater> so either use -b:v or -crf when encoding from h264 to vp8 [19:54] <teratorn> OK - well I still have the same issue without -sameq [19:54] <pasteeater> can you upload foo.mp4 somewhere? [19:54] <teratorn> sure, one sec [19:55] <teratorn> http://teratorn.org/code/foo.mp4 [19:58] <teratorn> pasteeater: foo.mp4 might be really screwy - I dunno [20:01] <pasteeater> it has some weird frame rate issues [20:03] <pasteeater> you can add -r 26.28 and it should encode [20:03] <pasteeater> what did foo.mp4 come from? [20:04] <teratorn> a rather complex capturing/compositing/encoding pipeline [20:04] <teratorn> which, apparently, has problems [20:07] <teratorn> pasteeater: thanks, that helps. do you know what specifically is wrong with foo.mp4? [20:27] <Mista_D> Can I utilize all 16 cores to encode a 2 pass 720p file from 1080p source? `-threads 0` only uses 200% instead of 1600% according to top. [20:31] <Mavrik> Mista_D, for which pass_ [20:31] <Mavrik> pass 1 will probably use only 1 or 2 cores [20:31] <Mavrik> pass 2 should use at least half [20:36] <JackD> pasteeater: [20:36] <JackD> can you give me a hand ? [20:40] <cbsrobot> JackD: I guess your card runs at a certain speed [20:40] <cbsrobot> eh framerate [20:40] <cbsrobot> why not do it with a for loop ? [20:41] <cbsrobot> or while [20:41] <JackD> but it could be slow, no? [20:42] <cbsrobot> well you want it to grab one fram per second - no? [20:43] <JackD> yes, but have no way to make ffmpeg drop some frames? [20:44] <cbsrobot> i=0; for ((;;)); do i=`expr $i + 1`; ffmpeg -f vfwcap -s 640x480 -vrames 1 -i 0 output-$i.jpg; done; [20:44] <cbsrobot> untested [20:45] <cbsrobot> ah shit - add: sleep 1; [20:46] <JackD> ok ok ok you convinced me, I will use a script or compile a c source, [20:47] <cbsrobot> I never used vfwcap btw [20:47] <JackD> I see [20:47] <JackD> I hnever use too [20:47] <JackD> thankyou [20:48] <cbsrobot> or maybe [20:48] <cbsrobot> use filters [20:48] <cbsrobot> like -vf select='not(mod(n\,15))' [20:48] <cbsrobot> for 15fps input [20:49] <cbsrobot> fmpeg -f vfwcap -s 640x480 -vrames 1 -i 0 -vf select='not(mod(n\,15))' output%d.jpg [20:49] Action: cbsrobot copied your command line ... [20:49] <cbsrobot> and drop the vframes - lol [20:50] <cbsrobot> I should stop writing wrong stuff [20:57] <JackD> cbsrobot: I tried the filtes but with no much sucess, the for ideia is not too hungry then I will chose it by now. Thanks [20:58] <cbsrobot> JackD: I guess the filters version is saner [20:59] <cbsrobot> if you use this, do you get an output ? [20:59] <cbsrobot> ffmpeg -f vfwcap -s 640x480 -i 0 test.mpg [20:59] <JackD> yes [21:00] <cbsrobot> whats the framerate of the output file ? [21:00] <JackD> - 0 is a surveillance camera with 30fps [21:01] <JackD> -i 0 * [21:01] <JackD> 29fps [21:01] <cbsrobot> ffmpeg -f vfwcap -s 640x480 -i 0 -vf select='not(mod(n\,29))' output%d.jpg [21:01] <cbsrobot> do you get jpeg files when running ^ [21:02] Action: cbsrobot thinks you can even use output-%05d.jpg [21:02] <JackD> I think I will get error with this, trying... [21:03] <JackD> too many arguments in mod [21:03] <cbsrobot> ah [21:04] <cbsrobot> ffmpeg -f vfwcap -s 640x480 -i 0 -vf "select=not(mod(n\,29))" output%d.jpg [21:05] <JackD> 217 jpg files in 10s [21:06] <cbsrobot> try [21:07] <cbsrobot> hmm [21:08] <cbsrobot> still thinking [21:08] <JackD> I still trying some idiot things here [21:08] <cbsrobot> ah [21:09] <cbsrobot> ffmpeg -f vfwcap -r 30 -s 640x480 -i 0 -vf "select=not(mod(n\,29))" output%d.jpg [21:10] <JackD> 217 files for 10s [21:10] <cbsrobot> shit [21:10] <JackD> a lot of [21:10] <cbsrobot> ffmpeg -f vfwcap -r 30 -s 640x480 -i 0 -vf "select=not(mod(n\,100))" output%d.jpg [21:11] <cbsrobot> do you get less jpegs ? [21:12] <JackD> i tried with 300 and i get 2 jpgs in 10 seconds [21:12] <JackD> what n\ means? [21:12] <cbsrobot> n = input frame number [21:12] <cbsrobot> \ = escapecharacter for , [21:13] <cbsrobot> , is used to separate filters aswll [21:13] <JackD> not is the bitewise NOt right? [21:13] <cbsrobot> not is just not [21:13] <JackD> ahh ahahahahaha [21:14] <cbsrobot> mod(n,100) is only 0, if you hit the 10th frame [21:14] <cbsrobot> now inverse that [21:14] <cbsrobot> you can quote me on that ! [21:14] <cbsrobot> lol [21:14] <relaxed> JackD: for highest quality jpgs add -qscale 1 [21:14] <cbsrobot> *100th [21:16] <JackD> inverse? [21:16] <JackD> aaaaa [21:17] <JackD> relaxed: if I use -qscale 6 i get less quality right? [21:17] <cbsrobot> JackD: yes [21:18] <relaxed> JackD: correct, I think the scale is 1 - 31 (or is it 51)? [21:18] <relaxed> it's 31 [21:18] <cbsrobot> imho 31 [21:18] <cbsrobot> or 42 ? [21:19] <cbsrobot> it's 42 [21:19] <relaxed> that's the answer to everything! :) [21:19] <cbsrobot> I heard it on TV ! [21:23] <JackD> if I use 100 or above I get a frame every 5s [21:23] <JackD> bellow it the result is the same 217 files [21:26] <cbsrobot> JackD: pastebin the lines for ffmpeg -f vfwcap -r 30 -s 640x480 -i 0 -t 10 -vf showinfo -f null -y /dev/null [21:36] <JackD> what info do you want from it? [21:36] <JackD> it repeated the same line [21:38] <cbsrobot> all to pastebin.com [21:38] <JackD> http://codepad.org/bU7Bq4xW [21:42] <JackD> have ffmpeg a silent option? [21:42] <cbsrobot> aaaah codepad wraps the line ... that sucks [21:43] <cbsrobot> JackD: pastebin please [21:43] Action: cbsrobot too lazy [21:43] <JackD> no no it was DOS problem [21:43] <JackD> not codepad [21:44] <JackD> I will try to enlarge [21:44] <cbsrobot> ah [21:44] <cbsrobot> np [21:50] <JackD> http://srb.no-ip.org:9073/pastebin/ffmpeg.html ? [21:51] <cbsrobot> JackD: cant connect [21:53] <JackD> I will ask someone to put it on pastebin ... [21:53] <cbsrobot> JackD: np [21:53] <cbsrobot> JackD: nevermind [21:54] <cbsrobot> I see the value of n [21:54] <cbsrobot> actually it should work as is [21:55] <cbsrobot> try to tune the mod value to your needs [21:55] <cbsrobot> ffmpeg -f vfwcap -r 30 -s 640x480 -i 0 -vf "select=not(mod(n\,50))" output-%05d.jpg [21:57] <JackD> anyway http://pastebin.com/CUmTS6Za [21:58] <cbsrobot> JackD: also wrapped [21:59] <cbsrobot> nevermind [21:59] <JackD> as I said is is because of the DOS not because of the pastebin or codepad [21:59] <JackD> I get 8 or 80 [22:00] <JackD> I can not tune it or it generates 217 or 1frame [22:04] <cbsrobot> JackD: lol [22:28] <davidOmatic> hi! can ffmpeg extract tracks from audio cds? [22:50] <osearth> "Dying is winning, the longer you embrace life and joy the more you won." - ose [23:01] <Mista_D> What is the basic difference between slice and frame thread modes? Does/can they affect quality of encoding at all? [23:04] <cbsrobot> Mista_D: affect quality - hopefully not [23:04] <cbsrobot> slice is like a frame split in to smaller chunks [23:05] <cbsrobot> frame threading is just one frame per thread [23:07] <Mista_D> cbsrobot: thanks. I thought when only partial frame inforamtion is available to the encoder it can not really achieve good compression as its can't reuse the parts of video that are the same... [23:11] <Mista_D> Is there a way to extract stdout & errout with multiple outputs?? eg: `./avconv -i 1080.ts -s 1280x720 720p.mp4 2>&1 >720.log -s 320x176 -r 12 preview.mp4 2>&1 >preview.log` -- doesn't work, logs are the same. [23:16] <cbsrobot> use ....mp4 > preview.log 2>&1 error.log [23:16] <cbsrobot> ah - multiple ... no idea [23:22] <_stev3> hey - can anyone tell me how i can get ffmpeg to return the resolution of a file? [23:23] <pasteeater> _stev3: ffmpeg -i input 2>&1 | awk '/Video/{print $7}' [23:25] <_stev3> that returns '[PAR' [23:26] <pasteeater> probably depends on your format [23:26] <cbsrobot> hehe [23:27] <pasteeater> or maybe ffmpeg -i input 2>&1 | grep -i video | grep -o "[0-9]*x[0-9]*" [23:27] <pasteeater> not very efficient [23:28] <_stev3> nice, thanks :D [23:28] <cbsrobot> ffprobe -show_streams input | egrep "width|height" [00:00] --- Sat Dec 3 2011

1 0

ffmpeg-devel.log.20111201
by burek 02 Dec '11

02 Dec '11

[00:29] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * r9dcd8b86f9 10ffmpeg/libavcodec/pictordec.c: [00:29] <CIA-41> ffmpeg: Use correct context for av_log. [00:29] <CIA-41> ffmpeg: Fixes ticket #695. [01:14] <CIA-41> ffmpeg: 03Michael Bradshaw 07master * r0275b75a7e 10ffmpeg/libavcodec/libopenjpegenc.c: BPP fix, YUVP9, 10, 16 support, make RGB24 default [03:02] <CIA-41> ffmpeg: 03Anton Khirnov 07master * r91b412e786 10ffmpeg/avconv.c: [03:02] <CIA-41> ffmpeg: avconv: update InputStream.pts in the streamcopy case. [03:02] <CIA-41> ffmpeg: This was broken in 2a651b719c309c5e2fc663a5a9d6ca36153ab98f. [03:02] <CIA-41> ffmpeg: 03Anton Khirnov 07master * r2092232581 10ffmpeg/libavformat/utils.c: [03:02] <CIA-41> ffmpeg: lavf: estimate frame duration from r_frame_rate. [03:02] <CIA-41> ffmpeg: If r_frame_rate is set, it should be more reliable for this than either [03:02] <CIA-41> ffmpeg: codec or stream timebase. [03:02] <CIA-41> ffmpeg: 03Anton Khirnov 07master * r741a05a293 10ffmpeg/avconv.c: avconv: compute next_pts from input packet duration when possible. [03:02] <CIA-41> ffmpeg: 03Anton Khirnov 07master * rd3e3ffa501 10ffmpeg/libavformat/rmdec.c: [03:02] <CIA-41> ffmpeg: rmdec: don't set codec timebase. [03:02] <CIA-41> ffmpeg: It's not supposed to be set outside of lavc. [03:02] <CIA-41> ffmpeg: Set r_frame_rate instead. [03:02] <CIA-41> ffmpeg: 03Anton Khirnov 07master * rc98c1f434e 10ffmpeg/libavformat/matroskadec.c: [03:02] <CIA-41> ffmpeg: matroskadec: don't set codec timebase. [03:02] <CIA-41> ffmpeg: It's not supposed to be set outside of lavc. [03:02] <CIA-41> ffmpeg: 03Anton Khirnov 07master * r7bb3e62594 10ffmpeg/avconv.c: [03:02] <CIA-41> ffmpeg: avconv: make copy_tb on by default. [03:02] <CIA-41> ffmpeg: I.e. on streamcopy set output codec timebase from input stream timebase [03:02] <CIA-41> ffmpeg: (as opposed to input codec timebase). This should be more sane, because [03:02] <CIA-41> ffmpeg: since the stream is not decoded, the input codec tb has no relation to [03:02] <CIA-41> ffmpeg: the timestamps of the copied packets. [03:02] <CIA-41> ffmpeg: 03Martin Storsjö 07master * r2583660664 10ffmpeg/libavformat/ (rtpdec.h rtsp.c): [03:02] <CIA-41> ffmpeg: rtpdec: Add an init function that can do custom codec context initialization [03:02] <CIA-41> ffmpeg: Signed-off-by: Martin Storsjö <martin(a)martin.st> [03:02] <CIA-41> ffmpeg: 03Miroslav SlugeH 07master * r06d7325ab1 10ffmpeg/libavformat/ (Makefile rtpdec.c rtpdec_formats.h rtpdec_g726.c): (log message trimmed) [03:03] <CIA-41> (18 lines omitted) [03:54] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r54e40b9c14 10ffmpeg/ffmpeg.c: [03:54] <CIA-41> ffmpeg: ffmpeg: factorize duration calculation in transcode_video() [03:54] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:54] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r6e96ad8215 10ffmpeg/ffmpeg.c: [03:54] <CIA-41> ffmpeg: ffmpeg: make copy_tb tri state so the user can select either timebase or [03:54] <CIA-41> ffmpeg: leave it at auto [03:54] <CIA-41> ffmpeg: Inspired-by a change from Anton Khirnov [03:54] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:26] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rec20fc1581 10ffmpeg/libavformat/ (avformat.h options.c utils.c version.h): [04:26] <CIA-41> ffmpeg: lavf: allow grouping packets in chunks of a user specified size and duration. [04:26] <CIA-41> ffmpeg: This is similar to MP4Boxs -inter [04:26] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:26] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r31f9032b78 10ffmpeg/libavformat/ (avformat.h options.c utils.c version.h): [04:26] <CIA-41> ffmpeg: lavf: add audio_preload option, this allows interleaving audio earlier [04:26] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:26] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r03b60509cb 10ffmpeg/ffmpeg.c: [04:26] <CIA-41> ffmpeg: ffmpeg: fix changing sample rate handling [04:26] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:52] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r67eb95ce15 10ffmpeg/ffmpeg.c: [04:52] <CIA-41> ffmpeg: ffmpeg: fix segfault with ffmpeg -h [04:52] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [09:59] <Bankin> I have a question about your task in Google code in for FFmpeg:create a logo, an icon or slogan - for icon or logo what shoul be the size of the logo/icon and for the slogan should it be just text or it shoulb be an image again ? [10:44] <Bankin> I have a question about your task in Google code in for FFmpeg:create a logo, an icon or a slogan - for icon or logo what shoul be the size of the logo/icon and for the slogan should it be just text or it should be an image again ? [11:44] Last message repeated 1 time(s). [12:30] <Compn> Bankin : make it a vector image, so it can be scaled small or large ;) [12:31] <Compn> it can be either, but i think a logo is what we are looking for, not the ffmpeg text [12:45] <Bankin> Thank you : ) [15:12] <ubitux> we really need to add various cmd line checks like ffmpeg/ffprobe/... -h [15:12] <ubitux> it seems i'm not the only one to make the OptionDef mistakes :p [15:36] <CIA-41> ffmpeg: 03Peter Ross 07master * rf3f488423a 10ffmpeg/libavcodec/pictordec.c: [15:36] <CIA-41> ffmpeg: pictordec: prevent segfault when reading corrupted files [15:36] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:37] <CIA-41> ffmpeg: 03Peter Ross 07master * r43a2a8b4a3 10ffmpeg/libavcodec/pictordec.c: [15:37] <CIA-41> ffmpeg: pictordec: use correct context for another av_log [15:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:35] <Tjoppen> hm [16:35] <Tjoppen> georg's single_eubc patch changes the seek test's result. I'll have to investigate why [17:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r3e1951e8f4 10ffmpeg/libavcodec/h264_ps.c: [17:37] <CIA-41> ffmpeg: h264: More restrictive more_rbsp_data_in_pps() [17:37] <CIA-41> ffmpeg: Fixes Ticket697 [17:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:02] <CIA-41> ffmpeg: 03Carl Eugen Hoyos 07master * rd9cc6bb432 10ffmpeg/ffmpeg.c: [18:02] <CIA-41> ffmpeg: Support ffmpeg -dn to disable copying data streams. [18:02] <CIA-41> ffmpeg: Fixes ticket #647. [18:47] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rb50be4e38d 10ffmpeg/libavcodec/vc1.c: [18:47] <CIA-41> ffmpeg: vc1_parser: fix NULL pointer dereference [18:47] <CIA-41> ffmpeg: Fixes Ticket456 [18:47] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:57] <cbsrobot> weird - converting qtrle to h264 gives me wrong colors and a unplayable file in quicktime, while qtrle to prores seems ok [18:58] <Daemon404> that could merely be because quicktime's h264 decoder is a piel of crap [18:58] <Daemon404> and can't handle a lot of stuff [18:58] <Daemon404> it needs a specific subset of h264 iirc [18:59] <cbsrobot> its just the prores to h264 is fine afterwards - i do not think its quicktime - at least not this time [19:00] <cbsrobot> well I need to investigate later [19:00] <Daemon404> oh [19:00] <Daemon404> that sounds strange. [19:00] <Compn> sounds like [19:01] <Compn> wrong colorspace conversion [19:01] <Compn> qtrle is paletted isnt it ? [19:01] <cbsrobot> it's rgb24 [19:02] <cbsrobot> It autoinserts following scale: [19:02] <cbsrobot> w:720 h:576 fmt:rgb24 -> w:720 h:576 fmt:yuv420p flags:0x4 [19:02] <cbsrobot> w:720 h:576 fmt:yuv420p -> w:720 h:576 fmt:rgb24 flags:0x4 [19:02] <cbsrobot> fo the qtrle to h264 conversion [19:02] <cbsrobot> *for [19:03] <cbsrobot> while qtrle to prores uses: [19:03] <cbsrobot> w:720 h:576 fmt:rgb24 -> w:720 h:576 fmt:yuv420p flags:0x4 [19:03] <cbsrobot> w:720 h:576 fmt:yuv420p -> w:720 h:576 fmt:yuv422p10le flags:0x4 [19:03] <cbsrobot> hmm - I need to run [19:04] <cbsrobot> I try to add a bugreport [19:13] <cbsrobot> hmm - ffplay can play it, mplayer says: unsupportet pixel fmt: 83, quicktime shows a black frame, vlc a purple one ... nice [20:47] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4a8e3324fb 10ffmpeg/libavcodec/libx264.c: [20:47] <CIA-41> ffmpeg: libx264: fix generic boolean support [20:47] <CIA-41> ffmpeg: Fixes Ticket660 [20:47] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:57] <cbsrobot> ^ nice [20:58] <cbsrobot> a nice gci task would be to add a "-target pal-bluray" support [20:58] <cbsrobot> and ntsc-bluray aswell [20:58] <nevcairiel> oh please no, people should just forget about pal and ntsc with bluray [20:58] <nevcairiel> make it all the same [20:59] <cbsrobot> well sometime sourcematerial does not allow you to - i guess [20:59] <nevcairiel> besides, bluray doesnt really have ntsc or pal modes [20:59] <Compn> nevcairiel : you think converting framerates (from pal to ntsc) is a better idea ? [20:59] <Compn> or from pal to (whatever bluray default you want) is a good idea ? [21:00] <nevcairiel> Blu-ray doesnt support 25fps progressive, so you have to do something badly with it. .) [21:00] <cbsrobot> see http://www.x264bluray.com/home/1080p [21:00] <Compn> lol [21:01] Action: Compn has no clue about bray [21:01] Action: Compn calling it bray from now on [21:01] <Daemon404> [15:00] < nevcairiel> Blu-ray doesnt support 25fps progressive, so you have to do something badly with it. .) [21:01] <Daemon404> you can cheat teh system [21:01] <Daemon404> you can flag it as 50i [21:01] <Daemon404> whe nit is actually 25p [21:01] <Compn> ugh [21:01] <Daemon404> i have a bunch of blurays that do this [21:01] <Daemon404> liek sherlock [21:01] Action: Compn puts his fingers in his ears and runs away [21:01] <nevcairiel> its a ugly hack [21:01] <nevcairiel> if a bad player trys to deinterlace it, quality degrades =( [21:02] <Daemon404> nevcairiel, id rather have that ugly hack than no 25p [21:02] <Daemon404> :P [21:02] <Daemon404> (i only use a computer to play) [21:02] <Daemon404> and i dont really understand why they didnt include a 25p spec... [21:02] <cbsrobot> isnt 25pfs using the same hack ? [21:02] <Daemon404> no [21:03] <cbsrobot> eh 25psf [21:03] Action: pasteeater still waiting on x264 --device [21:04] <cbsrobot> Progressive segmented Frame (PsF, sF, SF) is a scheme designed to acquire, store, modify, and distribute progressive-scan video using interlaced equipment and media. [21:04] <cbsrobot> ^ wiki [21:04] <nevcairiel> its really the same hack, yea [21:04] <nevcairiel> assuming the processing software understands 2:2 pulldown [21:06] <brontosaurusrex> hi, is there something special to do at compile time to enable multithreaded prores decoder/encoder? [21:07] <cbsrobot> brontosaurusrex: no [21:08] <brontosaurusrex> hrm, i'am getting zero speed boost [21:09] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rdd974c1bc1 10ffmpeg/libavcodec/ (allcodecs.c libx264.c): [21:09] <CIA-41> ffmpeg: libx264: Implement rgb24 support through a seperate AVCodec. [21:09] <CIA-41> ffmpeg: This avoids people mistakely encoding in a way that many players dont support. [21:09] <CIA-41> ffmpeg: Fixes Ticket658 [21:09] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:10] <Daemon404> cbsrobot, in ffmpeg you might [21:11] <Daemon404> since there are 2 prores decoders [21:11] <Daemon404> it might matter which is used. [21:11] <cbsrobot> ^ michaelni : thanks [21:13] <cbsrobot> Daemon404: both have CODEC_CAP_SLICE_THREADS [21:14] <Daemon404> i do seem to recall one being faster though [21:14] <Daemon404> but i see your point [21:18] <brontosaurusrex> ffmpeg -threads 0 and the rest of cli - sounds good or...? [21:18] <brontosaurusrex> getting really small speed boost [21:20] <brontosaurusrex> for example ffmpeg -threads 8 -i "$files" -vcodec prores -profile 2 -acodec pcm_s16le "$out.mov" [21:21] <cbsrobot> brontosaurusrex: what format is your infile ? [21:21] <brontosaurusrex> prores [21:21] <cbsrobot> ah prores -> prores ? [21:22] <brontosaurusrex> cbsrobot, yes, just testing [21:22] <Daemon404> is teh encoder multithreaded? [21:22] Action: Daemon404 checks [21:23] <Daemon404> (that may be your bottleneck) [21:23] <brontosaurusrex> Daemon404, yeah, iam not sure [21:23] <Daemon404> try outputting to /dev/null [21:23] <Daemon404> ? [21:23] <brontosaurusrex> or getting some other input than prores to test? [21:24] <brontosaurusrex> or output [21:24] <Daemon404> i thought your goal was to test prores decoding speed multithreaded [21:24] <brontosaurusrex> Daemon404, both [21:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r59a58884b9 10ffmpeg/ffplay.c: [21:37] <CIA-41> ffmpeg: ffplay: Copy audio side data too. This fixes handling of some rare nellymoser files that change the sample rate mid stream (sample file at: http://trac.videolan.org/vlc/ticket/5586) [21:37] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:37] <CIA-41> ffmpeg: Signed-off-by: Marton Balint <cus(a)passwd.hu> [21:37] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r8c15f9d79c 10ffmpeg/: [21:37] <CIA-41> ffmpeg: Merge remote-tracking branch 'cus/stable' [21:37] <CIA-41> ffmpeg: * cus/stable: [21:37] <CIA-41> ffmpeg: ffplay: Copy audio side data too. This fixes handling of some rare nellymoser files that change the sample rate mid stream (sample file at: http://trac.videolan.org/vlc/ticket/5586) [21:37] <CIA-41> ffmpeg: Merged-by: Michael Niedermayer <michaelni(a)gmx.at> [21:39] <ubitux> michaelni: 5a15602a4e99c730036c33b467f60248889219e1 any reason this wasn't merge? [21:46] <ubitux> i was thinking about merging fate{,2}.mak [21:57] <michaelni> ubitux, i dont remember, feel free to remove it [21:58] <ubitux> ok [21:59] <cbsrobot> this aint my day [21:59] <cbsrobot> now with latest x264 and ffmpeg all h264 files look dark ... [22:00] <michaelni> do they look better with old ffmpeg ? [22:00] <michaelni> old x264 ? [22:00] <cbsrobot> i'm testing [22:02] <cbsrobot> michaelni: a nice gci task would be to add a "-target pal-bluray" support [22:03] <michaelni> AFAIK tasks added now wil only show up in 2 weeks [22:07] <cbsrobot> seems older ffmpeg [22:10] <michaelni> older works ? [22:10] <michaelni> can you bisect ? [22:13] <cbsrobot> I wont have time but my last revision was quite new ... [22:16] <michaelni> which revission and how can i reproduce it ? [22:20] <michaelni> cbsrobot, yuv or rgb x264 ? [22:20] <cbsrobot> well ok [22:21] <cbsrobot> so it seems quicktime player shows the file really dark [22:21] <cbsrobot> vlc - no problem [22:22] <michaelni> is it >8bit or rgb and does qt support that ? [22:22] <michaelni> some change in the default encoding options maybe ... [22:23] <cbsrobot> I need to check [22:24] <brontosaurusrex> cbsrobot, lion? [22:24] <cbsrobot> no snow leo [22:25] <brontosaurusrex> same here, if you have a small sample i can test vlc vs qt on this system [22:25] <cbsrobot> wait i'll get back to master [22:26] <michaelni> we dont have a OSX fate client yet, anyone of you want to change that ? [22:26] <brontosaurusrex> what would be a fate client? [22:27] <cbsrobot> it's also way too dark with ffmpeg version N-34970-g6f326b5 [22:27] <cbsrobot> which i used before [22:29] <cbsrobot> hmmm - how to show in git what revision i'm using ? [22:31] <michaelni> git show [22:31] <cbsrobot> ok [22:31] <michaelni> brontosaurusrex, a fate client is something that tests ffmpeg regularly so we know early when we messed up [22:32] <brontosaurusrex> michaelni, the git getting, compiling, test running platform? [22:32] <michaelni> yes [22:32] <brontosaurusrex> michaelni, is that something automagic or mainly manual thing? [22:32] <michaelni> automatic [22:33] <michaelni> it just needs a kick once a month when it got mysteriously stuck [22:34] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r18abf46b6f 10ffmpeg/configure: [22:34] <CIA-41> ffmpeg: configure: add forgotten libx264rgb_encoder_deps [22:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:34] <brontosaurusrex> so the software/script is allready written? [22:37] <michaelni> yes, you just need a few lines that calls it in regular intervals can be a cronjob, manual or a loop with sleep [22:37] <michaelni> brontosaurusrex, i can forward you a mail that describes precissely how to set it up [22:38] <michaelni> and theres doc/fate.txt in git [22:38] <brontosaurusrex> michaelni, right, so fate.sh, is it written in a way to autoupdate itself? [22:38] <brontosaurusrex> ok, ill check the git, unless you have more info in that mail? [22:38] <michaelni> i think the mail says the same [22:41] <michaelni> mail forwarded [22:42] <brontosaurusrex> michaelni, thanks [22:43] <cbsrobot> michaelni: all fine [22:43] <cbsrobot> its quicktime that sucks [22:44] <cbsrobot> it has problems with high profile - as usual [22:45] <michaelni> that should be documented better, maybe a warning when one encodes with high profile [22:48] <cbsrobot> gci ... [23:34] <CIA-41> ffmpeg: 03Adrian Drzewiecki 07master * rdd7453a24e 10ffmpeg/libavformat/id3v2.c: [23:34] <CIA-41> ffmpeg: Fix id3v2 extended header handling. [23:34] <CIA-41> ffmpeg: When skipping over the extended header, take into account [23:34] <CIA-41> ffmpeg: that the size field has already been read. The extended header [23:34] <CIA-41> ffmpeg: also takes up space, so adjust total header length accordingly. [23:34] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:42] <CIA-41> ffmpeg: 03Derek Buitenhuis 07master * r69035f3ec7 10ffmpeg/libavcodec/libutvideo.cpp: [23:42] <CIA-41> ffmpeg: Revert "libutvideodec: Make buf_size local" [23:42] <CIA-41> ffmpeg: This change was dependent on a different patch that [23:42] <CIA-41> ffmpeg: never actually made it into FFmpeg, and it actually [23:42] <CIA-41> ffmpeg: ended up breaking builds. [23:42] <CIA-41> ffmpeg: This reverts commit 70cf7bb958899c369f778e29b8bd4203723bed04. [23:42] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:54] Action: Daemon404 sees michaelni got in a remark about git trees ;) [23:56] <cbsrobot> bcoudurier: will you submit vf_sub to ffmpeg ? [00:00] --- Fri Dec 2 2011

1 0

ffmpeg.log.20111201
by burek 02 Dec '11

02 Dec '11

[00:00] <Toux1> some one has the prob, when type "ffmpeg -filters" show an empty list? [00:19] <pasteeater> Toux1: are they using ubuntu? [00:20] <pasteeater> giany: you might get an answer if you give some more useful information [00:20] <pasteeater> giany: use a pastebin site to show your ffmpeg command and the complete console output [00:20] <Toux1> Hi pasteeater [00:20] <Toux1> thank you for your reply [00:20] <Toux1> no i'm using centos 5.6 with cpanel [00:21] <pasteeater> pastebin the complete console output from the ffmpeg -filters command [00:21] <pasteeater> i'm guessing your ffmpeg is too old [00:22] <Toux1> http://pastebin.com/ccXAQRDq [00:23] <giany> pasteeater: : http://pastebin.com/pj8GDBpi [00:25] <pasteeater> Toux1: i believe your ffmpeg is too old [00:26] <Toux1> jeje yes, but i don't know I can't install it from source compilling and installing, this is the yum install ffmpef and ffmpeg-devel .. i can't install a newest ffmpeg version.... [00:27] <pasteeater> giany: can you also show the output of "ffmpeg -i 'kb1_0.mp4' -f null /dev/null" [00:27] <pasteeater> Toux1: why can't you compile it? [00:28] <Toux1> it gives so much errors when I try to install [00:28] <pasteeater> what kind of errors? [00:29] <Toux1> let me do a pastebin for show you, thank you pasteeater :-) [00:30] <giany> pasteeater: http://pastebin.com/u95BYKL5 [00:31] <pasteeater> giany: does your first command actually create an output? [00:31] <giany> yes, it creates [00:32] <pasteeater> if it sounds ok you can probably ignore those errors [00:32] <giany> how can ignore those errors? things is that i run it through a php app [00:32] <giany> and it stops due to that [00:32] <pasteeater> but i recommend you pay attention to the other warning: WARNING: The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s [00:33] <pasteeater> so change -b 250 to -b 250k and same for -ab [00:33] <giany> let me test [00:33] <pasteeater> it won't stop your aac decoing errors though [00:34] <pasteeater> do you have a sample you can provide? [00:34] <pasteeater> of an input that gives the decoding errors? [00:34] <giany> you mean kb1_0? [00:35] <pasteeater> yes [00:36] <giany> it has aprox 300mb.. [00:37] <pasteeater> maybe you can isolate the audio stream to make a sample since it's the trouble maker. [00:37] <giany> if i use a smaller mp4 those errors go away [00:39] <giany> pasteeater: do drop=27671 mean those frames are dropped? [00:39] <sacarasc> Yes. [00:40] <giany> well..that sounds bad [00:41] <Toux1> now no errors pasteeater, but when I type ffmpeg not output on my server [00:41] <Toux1> http://pastebin.com/7Q1i7p6T [00:42] <Toux1> I did, ./configure after make and after make install this is the make install output, and server says command not found... [00:43] <Toux1> I don't know... [00:43] <giany> Toux1: /usr/local/bin/ffmpeg [00:43] <giany> probably its there if you did not specify --prefix in the ./configure [00:43] <Toux1> directory not exists... [00:44] <giany> updatedb && locate "bin/ffmpeg" [00:44] <Toux1> and exists [00:44] <Toux1> root@ns388770 [/usr/local/bin]# updatedb && locate "bin/ffmpeg" [00:44] <Toux1> /usr/local/bin/ffmpeg [00:45] <Toux1> root@ns388770 [/usr/local/bin]# ls [00:45] <Toux1> ./ ffmpeg* ffprobe* flvtool2@ nasm@ peardev* perl@ php-config* qt-faststart@ yasm@ [00:45] <Toux1> ../ ffplay* ffserver* lame* pear* pecl* php* phpize* re2c@ [00:45] <Toux1> exists.. but [00:45] <giany> but? [00:45] <Toux1> Says is not a directory [00:45] <Toux1> I can't enter [00:46] <giany> just run : /usr/local/bin/ffmpeg [00:46] <Toux1> ah yes [00:46] <Toux1> I have output with 0.8.7 installed [00:47] <Toux1> but when i try run my php script, says command no exists [00:47] <giany> put the proper path in the script [00:47] <giany> e.g : /usr/local/bin/ffmpeg [00:47] <Toux1> how can I do it [00:47] <giany> go #php :) [00:48] <Toux1> i have not defined that, I only use exec() and ffmpeg -i and rest of command [00:48] <giany> there its your problem [00:48] <giany> instead of ffmpeg -i [00:48] <giany> put /usr/local/bin/ffmpeg -i ... [00:49] <Toux1> WOWOWOW, THANK YOU!!!!!!!!!!!! [00:49] <Toux1> I stay so much hours thinking is not installed [00:49] <giany> read about linux relative and absolute path [00:50] <Toux1> Ok, thank you very much both, pasteeater, and you giany :-) [00:50] <giany> yw [00:50] <Toux1> I'll do ;-) [01:01] <giany> pasteeater: any idea what this means : "[libvpx @ 0x8843aa0] Failed to initialize encoder: ABI version mismatch" ? [01:03] <Toux1> Giany is obligatory have installed ffmpeg-devel? [01:03] <giany> no, not really [01:03] <Toux1> mm now says Unknown encoder 'libx264 [01:03] <giany> you need to install it too [01:04] <Toux1> with yum? [01:05] <giany> i'm not sure that is in the repos [01:12] <Toux1> /usr/local/bin/x264: error while loading shared libraries: libavformat.so.52: cannot open shared object file: No such file or directory [01:14] <giany> ldd /usr/local/bin/x264 [01:14] <giany> you might need to add /usr/local/lib to your ld.conf path and then run ldconfig [01:14] <Toux1> http://pastebin.com/C17A4gQU [01:16] <Toux1> where is ld.conf [01:16] <giany> /etc/ld.so.conf [01:17] <Toux1> so include /usr/local/lib [01:17] <Toux1> and run ldconfig another time no? [01:18] <giany> my ld.so.conf contains this : [01:18] <giany> /usr/local/lib [01:18] <giany> after you add that run :ldconfig [01:19] <Toux1> yes with include not run [01:21] <Toux1> when I run my ffmpeg command says the same error [01:21] <Toux1> not found x264 [01:23] <Toux1> must I change the way to call -vcodec libx264 ? [01:26] <relaxed> Toux1: yes [01:28] <Toux1> Hi, relaxed [01:28] <Toux1> actually i'm trying to call this lib with "-vcodec libx264" [01:29] <Toux1> but says not found, do you know other way? sorry my english, If I type /usr/local/bin/x264 --fullhelp [01:29] <Toux1> runs [01:35] <relaxed> does `ffmpeg -codecs 2>&1| grep libx264` return anything? [01:37] <Toux1> Nop, nothing [01:37] <relaxed> then ffmpeg wasn't compiled with libx264 support [01:38] <relaxed> Did you compile it? Distro and version? [01:38] <Toux1> centos 5.6 , and ffmpeg 0.8.7 [01:39] <Toux1> when i try ./configure --enable-libvpx --enable-libx264 --enable-gpl [01:39] <Toux1> says ERROR: libvpx decoder version must be >=0.9.1 [01:40] <relaxed> the version installed is too old [01:40] <Toux1> ERROR: libx264 version must be >= 0.115. now [01:40] <Toux1> pffff [01:40] <relaxed> Toux1: uname -a [01:41] <Toux1> Linux ovhserver 2.6.38.2-grsec-xxxx-grs-ipv6-64 #2 SMP Thu Aug 25 16:40:22 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux [01:42] <relaxed> Toux1: https://sites.google.com/site/linuxencoding/builds [01:43] <Toux1> and for install how can I install them [01:43] <Toux1> I download both? [01:44] <JodaZ_> so i am compiling an old ffmeg (2.4.5) and i get the error "undefined reference to `x264_encoder_encode'" i tried compiling libx264 fresh and installed the debianmultimedia ones for squeeze, and i still get that error [01:44] <relaxed> Toux1: download the ffmpeg link, unpack and run ./ffmpeg from the directory [01:44] <Toux1> thank you relaxed for your time and help :-) [01:44] <Toux1> I'll do now [01:45] <JodaZ_> i asume an api breakage [01:45] <JodaZ_> but when was it, and how do i get a version old enough ? [01:46] <relaxed> JodaZ_: figure out when 2.4.5 was released and check out the source with git based on the date. [01:46] <relaxed> for both projects [01:47] <JodaZ_> how do i find out when 2.4.5 was released ? [01:48] <relaxed> google.com [01:48] <Toux1> uploading relaxed, by wget get me error [01:49] <relaxed> what? [01:50] <Toux1> i 'm uploading the packpage ffmpeg from the google code site you said 2 mins left :-) [01:50] <relaxed> downloading? [01:50] <relaxed> :-P [01:50] <Toux1> hehe :-) [01:53] <Toux1> mmm ./ffmpeg not runs [02:07] <ZacS1234> Hi [02:07] <ZacS1234> is there a list any where of the metadata keys that can be used with mp4 containers? [02:17] <pasteeater> relaxed: is that your site? [02:18] <relaxed> pasteeater: yes [02:29] <pasteeater> relaxed: has anyone actually donated? [02:29] <relaxed> yes [02:29] <pasteeater> good [02:30] <pasteeater> No more Old Milwaukee for you. [02:31] <relaxed> Nothing's finer on a hot day :) [02:32] <ZacS1234> how does one set the creation_time metadata value, I have built a string "2011-12-01 01:29:53Z" and I try to set the creation_time key in the Output Format Context Metadata struct but I still get the default timestamp [02:35] <pasteeater> ZacS1234: i've only used '-timestamp now' to set creation_time [02:42] <ZacS1234> isnt that deprecated? [03:30] <kookykookie> is there any where i can get a 0.8.6 source? [03:35] <Mista_D> How do I get json output from ffprobe? With `ffporbe -print_format json -i test.mp4`I get stdout + empty brackets ): [03:45] <relaxed> Mista_D: ffprobe -print_format json -show_format -show_streams [03:46] <Mista_D> relaxed: much obliged. Thank you. [03:51] <kookykookie> im having trouble compiling a version of mplayer that uses ffmpeg source [03:52] <kookykookie> it says there is something wrong with libmpdemux/mp_taglists.c [03:53] <kookykookie> error: 'CODEC_ID_G723_1' undeclared here (not in a function) [03:53] <kookykookie> error: 'CODEC_ID_G729' undeclared here (not in a function) [03:59] <kcm1700> http://www.ffmpeg.org/releases/ffmpeg-0.8.6.tar.gz <- 0.8.6 source code? [04:00] <kookykookie> Thanks =) [05:01] <kookykookie> i just realize my question isnt really related to ffmpeg [05:49] <jackyyll> hi i'm trying to configure ffmpeg with --enable-rtmp and it's saying ERROR: librtmp not found [05:49] <jackyyll> --enable-librtmp* [06:02] <teratorn> jackyyll: installed rtmpdump/librtmp? [06:02] <jackyyll> yup [06:11] <ZacS123> Hi [06:12] <ZacS123> I can't seem to set the creation_time metadata value. my command is as follows: ffmpeg -i test.mov -vcodec libx264 -metadata creation_time="2011-12-01 16:12:05" output.mp4 [06:13] <ZacS123> the creation_time always comes out as 1970-01-01 00:00:00 [06:18] <relaxed> try date [06:28] <relaxed> ZacS123: It works using libav [06:28] <ZacS123> relaxed: does creation_time work for you in libav? [06:29] <ZacS123> I am using Zeraone's ffmpeg builds [06:29] <ZacS123> relaxed: also data & creation_time are meant to be used for different things. date does work [06:30] <relaxed> Yes it does. Here are some builds to test with. http://win32.libav.org/ [06:31] <relaxed> creation_time : 2011-12-01 16:12:05 [06:32] <ZacS123> im just downloading it now [06:33] <ZacS123> is one generally better off using ffmpeg/libavcodec or libav [06:33] <relaxed> jackyyll: make sure you install librtmp-dev too [06:37] <relaxed> ZacS123: I use libav (avconv) and Debian + Ubuntu have moved to libav as well. [06:59] <ZacS123> relaxed: I get a warning strptime() unavailable on this system, cannot convert the date string. [06:59] <ZacS123> and the creation_time with avprobe is still 1970 [07:01] <relaxed> ZacS123: Sorry to hear that. It works in linux. [07:52] <h725> hi, how can i see what codecs are available for the -vcodec option? [07:57] <kode54> ffmpeg -codecs [07:57] <h725> ffmpeg: missing argument for option '-codecs' [07:57] <kode54> huh [07:57] <kode54> must be a new option [07:57] <kode54> ffmpeg version 0.8.5.git-38a444e, Copyright (c) 2000-2011 the FFmpeg developers [07:58] <h725> http://pastebin.com/jvdcDYMR [07:58] <h725> hm I have 0.5.1 [07:58] <kode54> check the full help page, then [08:00] <h725> I know it's off topic but how do i get a newer ffmpeg on ubuntu? [08:01] <kode54> I got this version by installing a devel package instead of the regular flavor of ffmpeg [08:01] <kode54> but there's probably no such package in ubuntu [08:02] <relaxed> h725: http://bit.ly/3xSE5 [08:06] <h725> relaxed: thx I'll try that [08:06] <h725> will take ages on that atom processro [08:08] <relaxed> h725: Maybe it's time to install a more recent version of ubuntu. Might be faster :) [08:08] <h725> hm I'm tempted [08:08] <h725> and I fear that the whole thing does not work anyways :-/ [08:09] <relaxed> the netbook? [08:10] <h725> no it's an eeepc [08:28] <ZacS123> what swsScale flag should one use? [08:32] <relaxed> ZacS123: lanczos or spline [09:05] <ZacS123> relaxed: in what case would one use one over the ovther? [09:05] <ZacS123> *other [09:52] <phpmvc> is there a convenient way to get a newer ffmpeg for my debian distro [10:00] <cryptopsy> how to convert vob to mp3 ? [10:00] <cryptopsy> ffmpeg -i my.vob -ab 320 -ac 2 -f mp3 test.mp3 , doesn't work, file is 0 bytes [10:06] <sacarasc> cryptopsy: Pastebin the complete output. [10:18] <cryptopsy> sacarasc: http://pastebin.com/ssG2pzfv [10:19] <sacarasc> WARNING: The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s [10:19] <sacarasc> Add a k after 320. [10:19] <sacarasc> So it's -ab 320k [10:21] <cryptopsy> not necessary [10:21] <cryptopsy> also doesn't work [10:21] <cryptopsy> same error [10:21] <cryptopsy> warnings aren't errors [10:21] <cryptopsy> i also tried ithout the entire statement [10:21] <cryptopsy> taken from a command that worked for avi to mp3 [10:21] <cryptopsy> the difference is that theres many vobs here, but there was only one avi [10:21] <sacarasc> Might need -sn too. [10:22] <cryptopsy> it doesn't matter [10:22] <cryptopsy> im extracting audio, [10:22] <cryptopsy> subtitles don't mean shit [10:22] <relaxed> cryptopsy: your build lacks libmp3lame support [10:22] <cryptopsy> what's that supposed to mean? [10:22] <cryptopsy> that it can't produce mp3's ? [10:22] <cryptopsy> because it can [10:22] <sacarasc> Wait a minute... There is no audio stream in that file... [10:22] <cryptopsy> file ancient-skin.mp3 [10:22] <cryptopsy> ancient-skin.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 48 kbps, 44.1 kHz, Stereo [10:22] <sacarasc> Just video and 2 subs it looks like. [10:23] <cryptopsy> there's audio in it because i can listen to it with mplayer [10:23] <cryptopsy> whatever a 'stream' means [10:26] <cryptopsy> so what now ... [10:26] <cryptopsy> mplayer pipe to ffmpeg? [10:32] <relaxed> cryptopsy: grab the ffmpeg build here. https://sites.google.com/site/linuxencoding/builds , unpack and run ./ffmpeg -i input.vob -sn -vn -acodec libmp3lame -aq 0 output.mp3 [10:35] <relaxed> add -ac 2 [11:12] <cryptopsy> You can use FFmpeg to convert in a terminal window. According to the FFmpeg documentation, you can convert VOB to AVI with "ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi" where "snatch_1.vob" is your VOB file and "snatch.avi" will be your AVI file. You can install FFmpeg by typing "sudo apt-get install ffmpeg" in the terminal. [11:13] <cryptopsy> by that, the mp3 command will be ffmpeg -i my.vob -f mp3 acodec copy -test.mp3 [11:15] <relaxed> cryptopsy: did you download my build? [11:15] <cryptopsy> relaxed: no [11:15] <cryptopsy> didn't see your reply [11:16] <cryptopsy> what does your build do different? [11:16] <relaxed> it's build with libmp3lame support which you need to encode the vob's audio to mp3. [11:17] <relaxed> built* [11:17] <cryptopsy> i'll just install it with my distro [11:18] <cryptopsy> relaxed: ffmpeg is built with mp3 support [11:18] <cryptopsy> + - mp3 : Add support for reading mp3 files [11:18] <cryptopsy> thanks [11:18] <cryptopsy> i'll check it out when i get desperate [11:21] <relaxed> 1) your ffmpeg only has decoding support. 2) my build is a static binary, you don't have to install anything. [11:21] <relaxed> good luck [11:26] <cryptopsy> it is compiled with mp3 encoding support [11:26] <cryptopsy> it works on avi files [13:08] <kim0> Hi folks .. I have a seemingly corrupted mp4 video file. VLC and mplayer are not able to play it. Is there any magic ffmpeg can do to recover it please ? [13:09] <relaxed> kim0: try, ffmpeg -i input.mp4 -vcodec copy -acodec copy output.mp4 [13:10] <kim0> relaxed: the main error seems to be: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1879540] stream 0, missing mandatory atoms, broken header [13:12] <JEEB> well, more than corrupt it looks like you're missing whole chunks of the file then [13:12] <JEEB> although it doesn't say that you're specifically missing the index part [13:12] <kim0> umm [13:13] <JEEB> automatical recovery might be hard [13:13] <relaxed> maybe add -er 0 in there? [13:14] <relaxed> ^^ if that works it would be magical [13:14] <kim0> :) ok trying [13:15] <kim0> hmm no luck yet [13:15] <kim0> JEEB: relaxed .. I have similar files recorded by the same cell phone .. [13:15] <kim0> can I just dd the header from good file to bad file ? [13:15] <JEEB> not copypasta style, no [13:15] <kim0> if I know its offset that is [13:16] <JEEB> you might want to check with l-smash's boxdumper or a similar tool the parse'able parts of the file [13:16] <JEEB> and compare [13:16] <JEEB> to a non-broken file [13:16] <JEEB> the boxdumper pretty much outputs all the container-wise parse'able spots it can find. Positions etc. [13:17] <kim0> JEEB: does it output it to a "repaired" file ? or am I expected to manually utilize the output data [13:17] <JEEB> nothing magical [13:17] <JEEB> it's just a parser [13:17] <JEEB> it outputs the parse'able information about the file into stdout [13:17] <kim0> how's that helpful afterwards [13:17] <JEEB> then you use that information to learn how much if anything is salvage'able [13:18] <JEEB> kim0, if it can't parse for shit you're pretty much out of luck [13:18] <JEEB> :V [13:18] <JEEB> and you can compare the parsing output to a non-broken file from the same encoder/muxer [13:18] <JEEB> which in this case is your phone (?) [13:19] <kim0> JEEB: and then .. how do I fix the header [13:19] <kim0> hexedit ? [13:19] <JEEB> something along those lines [13:21] <kim0> JEEB: I just built boxdumper .. ran it .. here's the output: http://paste.ubuntu.com/755945/ [13:21] <kim0> looks good ? [13:22] <JEEB> the [] []s at the end look pretty non-convincing, but it does seem to have a nice amount of data [13:22] <JEEB> what was the name of the index box? [13:22] <kim0> JEEB: where do I see that ? [13:23] <JEEB> uhh, I'm just generally asking because I don't remember its name [13:23] <JEEB> having it relatively safe would be a GoodThing [13:24] <JEEB> you do seem to have the moov and trak boxes tho [13:24] <JEEB> you should compare to a non-broken file >_> [13:24] <kim0> any written guide into which bytes to copy .. what data is interesting is much appreciated [13:25] <kim0> I'm a techie .. but this is all foreign to me [13:25] <JEEB> you've not even compared the output of that file to a full file >_> [13:25] <JEEB> also, it all depends on what exactly is broken and if it's lethal [13:26] <JEEB> I mean, if the index box is borked you're just SoOutOfLuck [13:27] <JEEB> tl;dr in any case it means effort, and unless it's a really needed file I'd say you just try to make the file not go corrupt :3 [13:27] <JEEB> (and most probably poking the file format's specs as well, in no way this stuff is automated, unless the remuxer app in L-SMASH magically can remux your file) [13:29] <JEEB> since output from a similar device should be similar (in theory), that is why checking the parse'able output of both the file and a non-borked counterpart is a relatively good idea [13:30] <JEEB> unless you want to learn parsing the format yourself first completely -.- [13:30] <kim0> remuxer segfaults :) [13:30] <JEEB> sounds good [13:30] <kim0> damn! never thought it'd be THAT hard to recover a video [13:31] <kim0> I wouldn't mind loosing a few seconds of corruption [13:31] <kim0> but why loose everything [13:31] <kim0> it's a 200Mb file [13:31] <JEEB> depends on what exactly went awry [13:32] <JEEB> if it was the actual video stream part that got corrupted -- sure that would be something the parser doesn't even care about since it just feeds stuff to the decoder (put very simply, it's not always like that) [13:34] <kim0> JEEB: are you familiar with: http://vcg.isti.cnr.it/~ponchio/untrunc.php [13:34] <kim0> does that do what I'm trying to achieve ? [13:34] <kim0> I built the software .. but it seems to want two files, a bad one and a good one [13:35] <kim0> no idea where am I supposed to get the good one from [13:35] <JEEB> which is not surprising [13:35] <JEEB> you can't just draw conclusions off of a bad file and fix it :P [13:35] <JEEB> <JEEB> since output from a similar device should be similar (in theory), that is why checking the parse'able output of both the file and a non-borked counterpart is a relatively good idea [13:36] <JEEB> also, that thing seems to decode the video and audio tracks? [13:36] <kim0> JEEB: so would you expect that second file it needs, to be just another mp4 from the same device ? [13:36] <JEEB> -.- [13:36] <kim0> sorry .. just checking :) [13:37] <kim0> they're different sizes ..etc [13:37] <kim0> JEEB: ok will try with that tool, passing it another good file [13:37] <kim0> JEEB: thanks a lot for the guidance [13:37] <JEEB> depends on how robust that app is [13:38] <JEEB> I mean, you can't possibly fix everything there [13:38] <JEEB> also, you at least had a partial moov [13:39] <kim0> moov is what contains the actual video (+audio) right ? [13:39] <JEEB> no, I think it describes how streams etc. are layed out through the file [15:45] <Abhay1> y question is specific. When I use FFmpeg-5.2 for encoding multichannel audio ac3 it doesnot give me channel layout error, but when i encode ac3 using FFmpeg-8.7 it gives me channel layout error, why? [17:39] <chrisshattuck> Hi folks, any tips on adding subtitles with ffmpeg? I have SRT files for the subtitles and MP4 video files. [17:46] <GomGom> when using -preset or -profile for x264 encoding, ffmpeg or avconv complains that the option is invalid, how can i specify that this option really is for x264 and not any other avcodec class with such an option ? [17:53] <teratorn> hi I have a little program that is just supposed to transcode an input video file to a webm output... but it's not working exactly.. the output plays waaaay too fast, and I was hoping someone could point me in the right direction by looking at the ffprobe output: http://codepad.org/Wtnhh21w [18:27] <oschonrock> Hi. having trouble with ffmpeg & x11grab after upgrading to ubuntu 11.10 and the new nvidia 280.13 driver. All works brilliantly, except that when capturing a browser session the drop on <select> widgets is captured "erractically" without the backgrond layer. In the resulting video the drop down is not readable. ffmpeg version is latest git, command line is [18:27] <oschonrock> ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i "$screen_no" -acodec pcm_s16le -vcodec libx264 -preset ultrafast -threads 0 "$filename".mkv [18:27] <oschonrock> anyone seen this before? [18:28] <konr> Can you recommend me some good libraries to write video manipulation programs? [18:29] <teratorn> konr: ffmpeg? [18:30] <konr> teratorn: interesting... [18:30] <konr> haha [18:30] <oschonrock> I am seening this symptom using both firefox 8 and chrome 14 [18:32] <oschonrock> have googled around like mad, and can't find anything on it myself....feels like some re-paint event/ sync issue somewhere between the nvidia-driver/X11 and ffmpeg [18:33] <konr> teratorn: can I change the pixels and do things like creating a a box around the subtitle characters? [18:34] <teratorn> you can certainly change pixels... I'm unsure about subtitle effects [18:34] <teratorn> can anyone see what I'm doing wrong that the output webm plays too fast? http://codepad.org/StC97578 [18:42] <SMJ> how do I capture video at hd1080 and scale it down to hd720? [18:43] <SMJ> okay, I think I found it [19:26] <pasteeater> SMJ: 'ffmpeg -i input -vf scale=-1:720 output' if you didnt figure it out [20:04] <teratorn> do players get a "starting" pts value from headers or just from the first packet? [21:20] <ralinux> why it does not works "ffmpeg -i x.mp4 y.mp3"?? [21:20] <ralinux> hello, [21:21] <ralinux> is this really ffmpeg channel? [21:22] <relaxed> no, I created this channel and filled it with bots [21:22] <relaxed> paste your command and output to pastebin.com and maybe on of the bots will help you. [21:23] <relaxed> one* [21:25] <ralinux> http://pastebin.com/pKKK58ch [21:25] <ralinux> come on bots come on, plz [21:26] <pasteeater> hello. this is bot #15. [21:26] <sacarasc> Beep. No MP3 encoder. Beep. [21:26] <pasteeater> bot #27 is a faster typer than bot #15. [21:27] <hi117> oh wow, fancy [21:27] <pasteeater> ralinux: if you're too lazy to fix your ffmpeg, then pipe to lame: ffmpeg -i input -f wav - | lame - output.mp3 [21:28] <ralinux> Beep Beep [21:28] <ralinux> I'd like to fix my ffmpeg but I don't know how. [21:28] Action: relaxed answers in r2d2 nosies [21:28] <pasteeater> did you compile ffmpeg, or is it from your distro repo? [21:29] <ralinux> from my distro repo tgz slackware [21:29] <sacarasc> Slackware, easy to compile for! [21:29] <pasteeater> and they don't --enable-libmp3lame? [21:30] <sacarasc> ralinux: Do you have lame installed? [21:30] <hi117> or maybe he didnt have mp3lame installed at compiletime [21:30] <pasteeater> oh...--enable-memalign-hack. that tells me the maintainer doesn't know what he's doing because that's an option for windows, IIRC. [21:30] <ralinux> yes in /usr/bin/ [21:31] <ralinux> curiosly under puppylinux this works perfectly [21:31] <pasteeater> go find andrew.46 in slack irc and ask for a better slackbuild script. [21:32] <relaxed> Is andrew slacking again? [21:32] <relaxed> har har :) [21:33] <pasteeater> or you could be lazy and use relaxed static build [21:34] <relaxed> I'm going to write a bash script that asks a few questions and compiles ffmpeg git. Then collect millions... [21:34] <sacarasc> Will it use something like checkinstall to make packages for various distros? [21:35] <relaxed> No. It will be distro agnostic and install to the users home dir with a wrapper script for the libs. [21:36] <ralinux> finally the ffmpeg keeps ko [21:38] <relaxed> Hmm, I'll also have it ask if you want ffmpeg or libav (or both ;-) [21:39] <ralinux> libav?? [21:39] <pasteeater> that would just confuse users. and i don't see much of an advantage of doing that anyway [21:42] <relaxed> sacarasc pasteeater will you guys test it and give feedback? [21:42] <pasteeater> sure [21:42] <sacarasc> I could, sure. [21:43] <relaxed> I'll start writing then. [21:44] <pasteeater> and i'll finally get off my ass and do a centos compile guide and add it to the ffmpeg compile wiki section [21:44] <pasteeater> https://ffmpeg.org/trac/ffmpeg/wiki/CompilationGuide [21:44] <pasteeater> also the filtering guide is new. you guys should contribute to that if you like [21:44] <pasteeater> https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide [21:44] <pasteeater> it's a bit disorganized, but it's better than nothing [21:53] <brontosaurusrex> pasteeater, very nice, does it matter where scaling filter is in the chain? (like before input or before output) [21:55] <pasteeater> i think it is ignored if you apply it as an input option [21:55] <brontosaurusrex> ok [21:58] <brontosaurusrex> uf, and volume is no longer 0-265 ? [22:11] <j_off> does ffmpeg support joining video files [22:17] <sacarasc> Yes, but it's not very good at it. [22:19] <j_off> how is it not good [22:20] <sacarasc> It doesn't work with all file types. It is basically cat file1 file2 | ffmpeg -i - [22:28] <j_off> cat ? [22:28] <j_off> what does that have to do with ffmpeg [22:28] <sacarasc> That's basically how ffmpeg joins files. [22:29] <j_off> what is cat [22:29] <sacarasc> cat - concatenate files and print on the standard output [22:29] <j_off> does ffmpeg support cutting out commercials [22:30] <brontosaurusrex> sacarasc, so i guess ffmpeg is there only to rewrite headers [22:30] <sacarasc> You can cut out parts of a file with ffmpeg, but there are better tools for the job. [22:30] <j_off> what tool [22:52] <j_off> what is best tool to cut out commercials [23:02] <beandog> probably avidemux [23:02] <beandog> j_off [00:00] --- Fri Dec 2 2011

1 0

ffmpeg-devel.log.20111130
by burek 01 Dec '11

01 Dec '11

[00:23] <CIA-41> ffmpeg: 03Jean First 07master * r23b4f355bb 10ffmpeg/ffmpeg.c: [00:23] <CIA-41> ffmpeg: ffmpeg.c use b:v for video bitrates using a target [00:23] <CIA-41> ffmpeg: it's more consistent to use -b:v instead of just -b [00:23] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:08] <ubitux> michaelni: no bump needed with the audio preload? [01:25] <michaelni> ubitux, thx ill fix it [02:40] <CIA-41> ffmpeg: 03Aaron Colwell 07master * r39a7a5b8ab 10ffmpeg/libavcodec/pthread.c: [02:40] <CIA-41> ffmpeg: pthread: don't increment index on zero-sized packets. [02:40] <CIA-41> ffmpeg: The next call to decode() will update from an invalid index, which will [02:40] <CIA-41> ffmpeg: either lead to a memcpy() where dest==src (2 threads), or lead to a [02:40] <CIA-41> ffmpeg: crash (>2 threads). [02:40] <CIA-41> ffmpeg: Signed-off-by: Ronald S. Bultje <rsbultje(a)gmail.com> [02:40] <CIA-41> ffmpeg: 03Alex Converse 07master * rfdab793a46 10ffmpeg/avconv.c: [02:40] <CIA-41> ffmpeg: avconv: Handle audio sync for non-S16 sample formats. [02:40] <CIA-41> ffmpeg: Also fix reporting of the number of samples added/dropped. [02:40] <CIA-41> ffmpeg: 03Aneesh Dogra 07master * r464ccb0144 10ffmpeg/libavcodec/indeo3.c: [02:40] <CIA-41> ffmpeg: indeo3: check per-plane data buffer against input buffer bounds. [02:40] <CIA-41> ffmpeg: Fixes : http://bugzilla.libav.org/show_bug.cgi?id=102 [02:40] <CIA-41> ffmpeg: Signed-off-by: Alex Converse <alex.converse(a)gmail.com> [02:40] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rc40a35f8a7 10ffmpeg/: (log message trimmed) [02:40] <CIA-41> ffmpeg: Merge remote-tracking branch 'qatar/master' [02:40] <CIA-41> ffmpeg: * qatar/master: [02:40] <CIA-41> ffmpeg: indeo3: check per-plane data buffer against input buffer bounds. [02:40] <CIA-41> ffmpeg: avconv: Handle audio sync for non-S16 sample formats. [02:40] <CIA-41> ffmpeg: pthread: don't increment index on zero-sized packets. [02:40] <CIA-41> ffmpeg: Conflicts: [03:54] <durandal_1707> i found lost patch for escape130 video codec [04:01] <michaelni> durandal_1707, where ? :) [04:01] <durandal_1707> on web [04:01] <durandal_1707> i downloaded warzone rar [04:01] <durandal_1707> and i found dec130.dll with debugging symbols [04:02] <durandal_1707> from quick look in ht, i noticed dec300.cpp [04:03] <durandal_1707> patch, original work by Eli Friedman just appears to have some color issue, but reading wiki entry i got that apparently fixed [04:06] <durandal_1707> the wiki entry on multimedia.cx for that codec is also write by Eli [04:10] <michaelni> durandal_1707, so you have improved elis patch ? do i understand that correctly [04:13] <durandal_1707> https://github.com/richardpl/ffmpeg/tree/escape130 [04:14] <durandal_1707> it seems duration is incorrectly calculated [04:18] <michaelni> you mean the duration printed by ffmpeg ? [04:18] <durandal_1707> yea [04:18] <durandal_1707> and mplayer [04:19] <michaelni> likely some issue in rpl.c [04:19] <michaelni> is that happening with escape124 too ? [04:21] <durandal_1707> apparently does [04:22] <durandal_1707> PYRAMID sample ~9 vs 15sec [04:22] <michaelni> its quite simply because rpl.c does not set duration [04:23] <michaelni> so libavformat guesses from bitrates probably and they are maybe wrong [04:24] <michaelni> number_of_chunks*frames_per_chunk_fps or something like this could be used for duration [04:24] <michaelni> number_of_chunks*frames_per_chunk * fps [04:25] <michaelni> note AVStream.duration is in the timebase set by av_set_pts_info [04:27] <michaelni> also you shoudl set the author of the initial commit to eli if its his code [04:37] <durandal_1707> do i need to change date too? [04:39] <michaelni> i dont care about the date, you should improve the commit messages a bit though too when you are already at it [04:39] <michaelni> "escape124:" prefixes at the least [04:40] <michaelni> escape130 is what i meant :) [04:52] <durandal_1707> those tables should probably be generated at decoder initialization [05:00] <michaelni> could be done but they are not very large, sign_table is just 256 bytes [05:32] <CIA-41> ffmpeg: 03Joseph Wecker 07master * rd31c68a7ed 10ffmpeg/libavformat/flvenc.c: [05:32] <CIA-41> ffmpeg: removes spurious warning [05:32] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [05:32] <CIA-41> ffmpeg: 03Joseph Wecker 07master * rb55dd10d62 10ffmpeg/libavformat/flvdec.c: [05:32] <CIA-41> ffmpeg: Fix typeo in some rarely used memoization [05:32] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:02] <nathan0n5ire> could some people give me some feedback for this ffmpeg icon I created? http://f.imgtmp.com/Btx89.png [06:13] <michaelni> nathan0n5ire, its a bad time, i think most devels are sleeping :) i need to go to bed too now [06:13] <michaelni> but it looks nice [06:19] <michaelni> nathan0n5ire, one thing i was thinking of was logos for different occasions like google does show on their main page [06:19] <michaelni> but if this is for google code in, then talk with burek first, he is mentor of these logo tasks [06:22] <nathan0n5ire> I realized I actually like ffmpeg' [06:23] <nathan0n5ire> s current icon better then the one I made... [06:28] <michaelni> nathan0n5ire, you could for example try to modify the current one by adding snow to give it a winterly feel ... [06:28] <michaelni> but i really need to go to bed now, good luck with your logo task ! :) [06:29] <michaelni> i mean the current one on ffmpeg.org [06:29] <buzz_> late night eh michael.. sleep well :) [06:29] Action: buzz_ just got up [06:29] <michaelni> :) [09:55] <ubitux> michaelni: "AV_PERM_PRESERVE is missing" as rej_perms right? so if the input pad requires the buffer to be preserved, the buffer will be copied? [09:57] <ubitux> the images are not modified by the filter (but filtered), does it really apply here? [09:57] <ubitux> and last question: what use case does it affect? [09:58] <ubitux> mmh the last N frame feed to the filter will actually output a different frame, so i guess this is what it means by modifying [14:52] <michaelni> ubitux, it seems preserve is not implemented the way i remembered it, the current implementation would rather need rej_perm=AV_PERM_REUSE2 [15:25] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rdae7ff0416 10ffmpeg/libavcodec/wmalosslessdec.c: [15:25] <CIA-41> ffmpeg: wmall: remove tabs that leaked in from a merge and seem to have [15:25] <CIA-41> ffmpeg: managed to silently bypass the push checks. [15:25] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:25] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r4710993800 10ffmpeg/libavformat/movenc.c: [15:25] <CIA-41> ffmpeg: movenc: memset(0) new cluster elements after realloc() [15:25] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [15:25] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rdd32a305bd 10ffmpeg/libavcodec/wmalosslessdec.c: [15:25] <CIA-41> ffmpeg: wmall: fix return type of dump_int_buffer() [15:25] <CIA-41> ffmpeg: Found-by: Dominique Leuenberger [15:25] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [16:38] <vega13> hi [16:39] <vega13> is AVPacket a POD or non-POD ? [16:39] <vega13> : ) [16:41] <vega13> I'm trying to copy it with memcpy but keeps segmentation fault.. [16:42] <vega13> I'm trying to copy it into a custom packet, before i would just memcpy packet.data into the char* buffer [16:43] <vega13> but since avcodec_decode_video2 now only takes full avpackets i have to copy the whole avpacket [16:46] <michaelni> vega13, you probably searchj for av_dup_packet() and i dunno what POD is supposed to mean [16:47] <vega13> okay thanks [16:48] <vega13> so with av_dup_packet i can copy an avpacket into an char buffer? [16:49] <michaelni> no [16:50] <michaelni> but with av_dup_packet() you can keep it while demuxing more packets [16:50] <michaelni> if you want to put it in a char buffer then [16:51] <michaelni> you have to store its data&size at minimum [16:51] <michaelni> you also need dts/pts one way or another [16:52] <michaelni> the decoders wont care but you need them to display at the right time [16:52] <michaelni> and flags may be needed by 1 decoder IIRC [16:54] <michaelni> if you provide avpackets with pts/dts to decoders though then they can reorder the timestamps for you and provide some guess at what timestamp would be correct for the decoded frame [16:57] <vega13> They are already inside my packet, I get my packet from av_read_frame then I want to do memcpy(triPkg->encDataBuffer(),&packet, packet.size+1); [16:58] <vega13> the old code worked: memcpy(triPkg->encDataBuffer(), packet.data, packet.size) [17:01] <michaelni> av_dup_packet(); memcpy &packet, sizeof(AVPacket); and NO freeing of the packet before you are done should work [17:01] <michaelni> or you can just keep using your old code [17:02] <michaelni> (old will fail with 1 decoder but it did not work before either) [17:03] <vega13> thanks ! [17:05] <vega13> gh then they can reorder the timestamps for you and provide some guess at what timestamp would be correct for the decoded frame [17:07] <vega13> sorry [17:07] <vega13> accidently pasted inside here [17:20] <ubitux> michaelni: any idea why this mp3 stream is somewhat misdetected?: http://mp3.live.tv-radio.com/2783/radio-pirate-iledefrance/radio-pirate-ile… [17:21] <ubitux> it looks like it "forgets" to send headers [17:21] <ubitux> i'm not sure the server is at fault [17:22] <ubitux> i get various "Format mp3 detected only with low score of 25, misdetection possible!" and header missing messages [17:33] <michaelni> ubitux, i see header missing but no low scores [17:33] <michaelni> do you have a dumped file with low score to reproduce ? [17:33] <ubitux> sure just a sec [17:33] <michaelni> header missing is probably due to mp3 frames being cut at random byte offset [17:35] <ubitux> wget 'http://blackhole.ubitux.fr/rp.mp3' [17:35] <michaelni> confirmed ill look at it [17:36] <ubitux> thank you [17:46] <michaelni> ubitux, do you have a actually misdetected one or just low score ? [17:46] <michaelni> it has 479 mp3 frames and 500 is the threshold for returning a higher score [17:47] <ubitux> no just low scores + warn/errors; decode and playback are fine [17:47] <ubitux> oh, so a bigger sample should do the trick? [17:47] <michaelni> yes [17:48] <michaelni> the problem with detecting mp3 quickly is that everything contains mp3 and so a truncated mpeg-ps file could be misdetected as mp3 if the threshokd is too low [17:50] <ubitux> how do you make a distinction between them? [17:50] <ubitux> if 500 audio frame oly mp3? [17:50] <ubitux> only* [17:50] <michaelni> comparing scores frorm the probe functions [18:20] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rb51eaf3b8c 10ffmpeg/libavformat/mp3dec.c: [18:20] <CIA-41> ffmpeg: mp3probe: Detect mp3 stronger with just 200 frames, this should speed up detection [18:20] <CIA-41> ffmpeg: on mp3 streams. [18:20] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:20] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r8d5078c10b 10ffmpeg/libavformat/ac3dec.c: [18:20] <CIA-41> ffmpeg: ac3probe: Change threshold from 500 to 200 to keep in sync with mp3. [18:20] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:22] <ubitux> heh, thanks [18:54] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r8f025f2f60 10ffmpeg/libavfilter/defaults.c: [18:54] <CIA-41> ffmpeg: avfilter: align allocated images to 32 for AVX and IFF [18:54] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [18:54] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rfd3e75d2d4 10ffmpeg/libavcodec/utils.c: [18:54] <CIA-41> ffmpeg: lavc: horizontal align for IFF [18:54] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r957a593cd9 10ffmpeg/libavformat/flvdec.c: [21:30] <CIA-41> ffmpeg: flvdemux: export flags for nellymoser through side data. [21:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r11828b8885 10ffmpeg/libavcodec/nellymoserdec.c: [21:30] <CIA-41> ffmpeg: nellymoserdec: Use sidedata to support mid stream sample rate changes. [21:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:30] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * r77af2672b4 10ffmpeg/libavcodec/utils.c: [21:30] <CIA-41> ffmpeg: lavc: Fix side data handling for audio decoders. [21:30] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:22] <CIA-41> ffmpeg: 03Michael Niedermayer 07master * rc863d3751f 10ffmpeg/libavformat/movenc.c: [23:22] <CIA-41> ffmpeg: movenc: replace cluster memset by zeroing only the needed field. [23:22] <CIA-41> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [00:00] --- Thu Dec 1 2011

1 0

ffmpeg.log.20111130
by burek 01 Dec '11

01 Dec '11

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

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