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March 2016

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ffmpeg-devel.log.20160325
by burek 26 Mar '16

26 Mar '16

[04:06:55 CET] <cone-532> ffmpeg 03James Almer 07master:21cd0228beca: avcodec/libopenjpegdec: fix mixed declarations and code [04:06:55 CET] <cone-532> ffmpeg 03James Almer 07master:99388eb091d0: avcodec/libopenjpegenc: fix mixed declarations and code [08:27:34 CET] <ubitux> mmh... fftw3... [08:27:40 CET] <ubitux> why not simply improve our asm [08:31:05 CET] <nevcairiel> because everyone can plug in an external library, improving our asm takes brains [08:31:18 CET] <ubitux> maybe it is just for benchmark? [08:32:23 CET] <nevcairiel> that would bbe a silly thing to add to the code [10:03:20 CET] <ubitux> michaelni: do you want time to look at the sws/aarch64 hscale patch later, or i can apply? [10:29:54 CET] <durandal_1707> so ok to apply j2k parser? [11:54:33 CET] <ubitux> so for some reason [11:54:49 CET] <ubitux> with ffmpeg -i dsc00451.jpg -vf scale=1024x1024 -f null - [11:55:05 CET] <ubitux> on x86, it's going through yuv2yuvX_sse3 [11:55:18 CET] <ubitux> while on aarch64, it's going into yuv2planeX_8_c [11:56:24 CET] <ubitux> why isn't it using ff_yuv2planeX_8_sse2 for example? [11:56:31 CET] <ubitux> (on x86) [11:57:35 CET] <ubitux> mmh it's also actually entering yuv2planeX_8_c [11:57:37 CET] <ubitux> now i'm confused [12:21:48 CET] <michaelni> durandal_1707 feel free to apply [12:22:45 CET] <michaelni> ubitux, if you think the code is ok feel free to apply, iam no aarch64 asm guy [12:23:03 CET] <ubitux> you're still sws guy so i was asking [12:23:54 CET] <ubitux> michaelni: while i have you, did you see my sws question about the scale filters the other day? [12:24:08 CET] <ubitux> @ubitux michaelni: double p = -2.196152422706632; (libswscale, spline code), is this 3*(1-sqrt(3)) or just a coincidence? [12:24:10 CET] <ubitux> @ubitux i lack too much knowledge about this code to have meaningful remarks but it looks like it could involve the filter parameter [12:24:12 CET] <ubitux> @ubitux sth like param * (1-sqrt(3)), like in other algorithms [12:24:14 CET] <ubitux> this one ^ [12:30:17 CET] <ubitux> :( [12:34:31 CET] <michaelni> ubitux, didnt see the spline question, must have missed it [12:37:44 CET] <michaelni> ubitux, looks like 3*(1-sqrt(3)) [12:37:44 CET] <kierank> michaelni: did you see h264 segfault? [12:39:17 CET] <michaelni> ubitux, anything could involve the filter param, why this one? does it look better for an image, if so please elaborate / provide the image [12:39:33 CET] <michaelni> & testcase [12:39:49 CET] <michaelni> kierank, hmm, i dont immedeatly remember [12:40:20 CET] <kierank> https://trac.ffmpeg.org/ticket/5371#comment:1 [12:40:46 CET] <ubitux> michaelni: i was merely looking at the code trying to understand; that constant was/is an enigma to me, and when i saw it was apparently 3*(1-sqrt(3)) and that the other p computation in other filters were involving either 3 or param[0], i was wondering if there was any relationship [12:41:13 CET] <ubitux> and if that p computation shouldn't be matching [12:43:57 CET] <michaelni> ubitux, i suspect (but dont really remember) that numerical value was the result of some numerical optimization to result in a spline with continous derivates at the control points and specific properties at the ends [12:44:20 CET] <michaelni> but i could be totally wrong [15:07:14 CET] <atomnuker> how come so many video parsers check for PARSER_FLAG_COMPLETE_FRAMES yet only flac in libavformat enables that flag [15:11:27 CET] <nevcairiel> its set if need_parsing is AVSTREAM_PARSE_HEADERS [15:11:31 CET] <nevcairiel> which is rather commonb [15:11:33 CET] <nevcairiel> -b [15:12:33 CET] <nevcairiel> if a container knows its always split properly already, then it sets this and avoids the overhead [15:48:05 CET] <ubitux> oh, i saved 4 mul and 1 add [15:48:10 CET] <ubitux> let's do better. [15:48:15 CET] <ubitux> zomg speed [15:48:49 CET] <kierank> ubitux: can I ask why you use filters of specific sizes [15:49:26 CET] <ubitux> common size of gl textures, and inputs are of known size [15:51:13 CET] <Daemon404> tep2 next week, nevcairiel? wm4? ubitux? [15:51:30 CET] <wm4> sure [15:51:32 CET] <ubitux> i'm going to take some holidays, so it sounds perfect to me [15:51:40 CET] <Daemon404> cool. [15:51:42 CET] <JEEB> holidays... sounds so good... [15:51:44 CET] <ubitux> i'll probably add a "sub_time_base" to codecpar [15:51:44 CET] <Daemon404> i'd haunting me. [15:51:49 CET] <JEEB> I'll probably get more active in a few weeks [15:51:50 CET] <Daemon404> it's* [15:51:58 CET] <JEEB> been doing long days for almost a year now [15:52:12 CET] <wm4> Daemon404: what does "fiel" mean on the etherpad [15:52:17 CET] <Daemon404> fiel atom [15:52:21 CET] <wm4> oh [15:52:25 CET] <Daemon404> the diff for those files is all a missinfg fiel atom [15:52:29 CET] <Daemon404> related to ->field_order [15:52:35 CET] <Daemon404> probably something i nffmpeg.c [15:54:06 CET] Action: wm4 curses ffmpeg.c with foam coming from the mouth [16:04:30 CET] <ubitux> foam or fiel ? [16:04:37 CET] Action: ubitux goes out [16:16:07 CET] <cone-671> ffmpeg 03Clément BSsch 07master:277408b7f16e: sws/aarch64/yuv2rgb: save a few mul and add [16:38:18 CET] <nevcairiel> Daemon404: the field order problem is simple, ffmpeg.c derives the field order during playback from the frames, and then sets avctx->field_order appropriately, which doesnt seem like its "wrong" to do so in general .. and mov writes all its headers at the end of muxing, but codecpar only gets "generated" once when opening the muxer, and not updated afterwards [16:38:38 CET] <nevcairiel> so in short, the compat code just doesnt cover such a use case [16:38:59 CET] <nevcairiel> the use-case in itself doesnt seem entirely wrong, not sure its ever claimed anywhere you cant modify the avctx after opening the muxer [16:39:22 CET] <nevcairiel> (in the past it was used for encoding even, it would get modified all the time) [16:39:51 CET] <nevcairiel> we could add some hack-ish thing to refresh the value, or in particular only the field_order value, or something [16:43:44 CET] <Daemon404> nevcairiel, hmm [16:44:34 CET] <nevcairiel> it would of course get fixed for ffmpeg.c when its moved to codecpar, but there is a reason its not done in the same step, it allows to test the compat :) [16:45:28 CET] Action: Daemon404 thinks the whole field_order thing is a tad iffy, but eh [17:46:54 CET] <cone-671> ffmpeg 03Thilo Borgmann 07master:4ebf0b109cdb: lavu/dict: Add new flag to allow multiple equal keys. [17:46:55 CET] <cone-671> ffmpeg 03Thilo Borgmann 07master:4d251723c0a1: lavfi: Add coreimage filter for GPU based image filtering on OSX. [17:51:29 CET] <ubitux> -#define LIBAVFILTER_VERSION_MINOR 39 [17:51:31 CET] <ubitux> +#define LIBAVFILTER_VERSION_MINOR 40 [17:51:33 CET] <ubitux> #define LIBAVFILTER_VERSION_MICRO 102 [17:51:35 CET] <ubitux> :( [17:53:28 CET] <Daemon404> context? [17:53:39 CET] <ubitux> 102 100 for micro [17:53:46 CET] <Daemon404> i know [17:54:02 CET] <ubitux> from last commit(s) [17:54:03 CET] <nevcairiel> thilos commit, probably [17:54:05 CET] <Daemon404> o ok [17:55:00 CET] <wm4> I sure like how that dict API change was just pushed without proper discussion [17:55:16 CET] <nevcairiel> wasnt there an discussion with people expressing concerns? [17:55:30 CET] <wm4> just me [18:33:04 CET] <BBB> I feel nowadays most concerns are just noe addressed [18:33:18 CET] <BBB> if you ignore and push quickly, you dont have to address concerns [18:33:24 CET] <BBB> not the first time that happens [18:35:59 CET] <jamrial> if that happens, the person with the concerns should revert the commit [18:36:03 CET] <jamrial> if people actually showed some resistance when this happens, and there was some concequence for those pulling this crap (no more commit rights for example) it would stop happening [18:36:47 CET] <jamrial> but so far apathy and "whatever, fuck it" tends to favor the person pulling this crap [18:41:58 CET] <BBB> I just feel Im too old to get into fights about this stuff [18:42:16 CET] <Daemon404> it's a fundementaimbalance [18:42:33 CET] <Daemon404> the people wanting to push thier (bad) code have 10x more energy to argue their point [18:44:08 CET] <Daemon404> ... as i spot a ganesh patch to add fftw [18:44:09 CET] <Daemon404> sigh. [18:45:21 CET] <kierank> durandal_1707: do we have a moving zoneplate like gstreamer? [18:52:13 CET] <wm4> "I believe I've figured out exactly what caused it: a commit to FFmpeg from Feb 9th that changed the optimization settings on GCC 4.9 or higher to not disable tree vectorization anymore when using [the default opt level of] -O3. If I add -fno-tree-vectorize to FFmpeg's extra-cflags to force the old behavior, then mpv is fine. I was using GCC 5.3.0." [18:52:21 CET] <wm4> does anyone know about such an issue? [18:52:45 CET] <wm4> (this is on 32 bit Windows) [18:53:15 CET] <atomnuker> there was a large discussion around that, most people thought that gcc's vectorization was mature enough [18:54:29 CET] <wm4> "thought" [18:56:44 CET] <atomnuker> that bug says otherwise [18:56:59 CET] <jamrial> it's not quite there yet. try to run fate vsynth tests on gcc 5.3 x86_64 linux with cpuflags set to 0, and watch huffyuv crash because of a miscompilation of diff_bytes_c() [18:57:24 CET] <jamrial> nobody noticed or cared since there are asm versions for that [18:58:31 CET] <jamrial> admitedly, the code in question is weird and was written this way because we disabled tree vectorization [18:59:24 CET] <jamrial> every other c boilerplate dsp function compiles fine [19:00:29 CET] <BBB> if it miscompiles, we should re-disable it [19:00:32 CET] <BBB> and report bug to gcc [19:01:11 CET] <jamrial> i tried to make a standalone test case but couldn't reproduce it that way [19:01:39 CET] <jamrial> and didn't want to report yet another bug with instructions to compile ffmpeg and run a test suite [19:02:20 CET] <nevcairiel> optimization bugs are hard to reproduce in minimal cases, unfortunately [19:02:32 CET] <durandal_1707> kierank: wut? [19:02:41 CET] <kierank> moving zoneplate filter [19:03:08 CET] <jamrial> it's also far from important or critical, seeing we mmx optimized versions, so unless you're still rocking a pentium from 1993 you'll be fine [19:03:45 CET] <wm4> " Atom Z3735F (Silvermont)" [19:03:45 CET] <kierank> what other libs can we get rid of apart from dcadec [19:05:10 CET] <durandal_1707> kierank: what it does? [19:05:16 CET] <jamrial> libwavpak? [19:05:24 CET] <jamrial> we have both decoder and encoders for that [19:05:36 CET] <kierank> durandal_1707: https://www.youtube.com/watch?v=aRqkGs0o6h0 [19:05:37 CET] <durandal_1707> nothing because of Carl [19:05:47 CET] <kierank> and it goes up and down [19:05:47 CET] <durandal_1707> libnut [19:05:58 CET] <atomnuker> whoa, we have a wavpack encoder [19:06:54 CET] <durandal_1707> kierank: why you need it? [19:07:03 CET] <kierank> it's a standard test pattern [19:07:07 CET] <jamrial> someone could take the celt parts of the opus interal decoder and write an actual celt decoder, so we can get rid of libcelt [19:07:21 CET] <nevcairiel> would that be enough to do that? [19:08:32 CET] <durandal_1707> atomnuker: wavpack supports floats, which is awesome [19:09:39 CET] <durandal_1707> kierank: I guess I will write one [19:09:59 CET] <wm4> what do you think about making ffmpeg more modular [19:10:05 CET] <wm4> like having external filters [19:10:35 CET] <durandal_1707> nice idea, but my hands are tied [19:10:58 CET] <llogan> durandal_1707: but then you can make compiz 3d desktop effect!~!!~11!! [19:11:01 CET] <wm4> tied? [19:11:25 CET] <durandal_1707> libav does mandate API [19:11:42 CET] <kierank> I want to write a lower level lavfi api [19:12:09 CET] <wm4> as in? no buffersrc/sink? [19:12:13 CET] <durandal_1707> llogan: I still have no idea what is that.. [19:12:25 CET] <kierank> wm4: yeah just send things to filters manually [19:12:39 CET] <wm4> durandal_1707: developing in Libav is actually less painful than in ffmpeg (shorter compile times, less insanity, less resistance to remove insanity) [19:13:12 CET] <nevcairiel> if compile time is an argument for you, you are the one with insanity =p [19:13:38 CET] <durandal_1707> less filters, decoders, demuxers, muxers, encoders.. [19:13:43 CET] <jamrial> wm4: --disable-everything --enable-what-i-want :p [19:14:21 CET] <durandal_1707> but yes I get your idea [19:14:29 CET] <llogan> durandal_1707: i guess he wants a VV->V filter? "cube spin" transition/wipe. [19:14:36 CET] <wm4> doesn't help if you change central headers and then want to rerun fate with everything and such [19:14:55 CET] <nevcairiel> our fate also tests substantially more [19:14:57 CET] <nevcairiel> so there is that :p [19:15:20 CET] <jamrial> and overall coverage is still poor [19:15:28 CET] <nevcairiel> thats true [19:15:39 CET] <durandal_1707> llogan: for free? No way [19:15:46 CET] <wm4> and before michaelni got enough of merging, you didn't have to care about the ffmpeg side of things [19:21:14 CET] <atomnuker> less insanity, less resistance to remove insanity << plenty of insanity got removed, the remaining insanity is there because people think there's a resistance to removing the insanity [19:21:47 CET] <kierank> there is [19:22:42 CET] <jamrial> atomnuker: carl put himself as "maintainer" of libutvideo just to have an argument against removing it [19:22:56 CET] <llogan> what do we have yet? ffserver, libutvideo, several prores encs [19:23:30 CET] <atomnuker> I'm sure if someone submits a patch again and properly explains why it needs to be removed it will be removed [19:23:46 CET] <llogan> that would be the third attempt, IIRC. [19:24:02 CET] <BBB> Im willing to fix the prores debacle as part of libav agreeing to merge our two projects back together [19:24:06 CET] <BBB> I think I offered that before [19:24:18 CET] <nevcairiel> thats never going to happen [19:24:20 CET] <atomnuker> third time's always lucky... I think [19:24:26 CET] <kierank> Our prores encoded files don't even work on real hardware [19:24:30 CET] <BBB> then well have multiple prores encdecoders for a very long time [19:24:30 CET] <jamrial> atomnuker: that's what happened last time. patch + arguments in favor of removing it. the result was carl doing what i said above [19:24:34 CET] <nevcairiel> they are happy to live in their ignorant bubble, even if noone uses their project anymore [19:25:00 CET] <BBB> I think they would be happier if we were nicer to them [19:25:05 CET] <BBB> calling them ignorant doesnt help [19:25:18 CET] <nevcairiel> i called their bubble that, not them! [19:25:19 CET] <nevcairiel> :D [19:25:26 CET] <llogan> kierank: with either encoder? (there are two, right?) [19:25:41 CET] <BBB> you called them out for living in it [19:25:50 CET] <kierank> llogan: yes [19:26:21 CET] <nevcairiel> BBB: but thats what they do, may not be pretty but its the truth, ignore everyone else and work away [19:27:39 CET] <kierank> They'll cherry pick so they aren't ignoring per se ;) [19:28:28 CET] <llogan> I think Ronald has a better perspective though, having more experience working with them. also, one of them seem quite sensitive given the emails i got regarding responses on mailing list [19:29:01 CET] <durandal_1707> kierank: are you saying that libav prores is better? [19:29:07 CET] <BBB> I meet them occasionally [19:29:12 CET] <BBB> so I talk with them about this stuff, yes [19:29:29 CET] <BBB> I assume Daemon404 talks about this too, although maybe it got old after a while [19:29:50 CET] <durandal_1707> where you meet them? [19:29:59 CET] <BBB> durandal_1707: we all live in NY [19:30:02 CET] <kierank> durandal_1707: no [19:30:20 CET] <BBB> nevcairiel: blunt question - why are you interested in merging? do you work for some video ingestion company also? [19:31:00 CET] <llogan> kierank: can you give a few examples of this hardware? [19:31:27 CET] <jamrial> from what i've seen, they will work with you if your objective is to do something for the project. they will accept patches and maybe help you writing them [19:31:33 CET] <kierank> Hyperdeck pro [19:31:56 CET] <jamrial> they will however not work with you if your objective is to find some common solution between the two projects even with the argument of doing it for the sake of the end user [19:32:27 CET] <durandal_1707> exactly [19:33:21 CET] <llogan> kierank: oh, BM. i couldnt even get one of their capture devices to work for me... [19:33:35 CET] <wm4> I'm interested in merging the projects because there are still some talented devs in Libav, and the split is causing lots of friction (having to merge their commits etc.) [19:34:52 CET] <jamrial> wm4: they seemingly have no interest in a merge, and we have some really big boulders in the way of progress, like carl runing every attempt at diplomacy [19:35:43 CET] <wm4> can we just fire carl [19:36:10 CET] <jamrial> it would be nice if he at least stopped screaming "thieves and liars" like a broken record every time some remotely related subjects come up [19:36:22 CET] <Compn> wm4 : combine powers with j-b to merge projects [19:36:28 CET] <Daemon404> well he could also stop harass everyone via private email too [19:36:30 CET] <Daemon404> but that wont happen [19:37:54 CET] <kierank> Because certain people (TM) support Carl [19:38:34 CET] <jamrial> support, or tolerate? [19:38:41 CET] <atomnuker> excommunicate? [19:38:59 CET] <kierank> Support and/or tolerate [19:39:01 CET] <durandal_1707> that's possible? [19:39:29 CET] <BBB> Im exactly with wm4, I want a merge because they have some very talented devs and I dont want them to just die out [19:39:35 CET] <kierank> Yes ban him [19:40:50 CET] <durandal_1707> tell him what is going to happen if he continues that [19:40:56 CET] <nevcairiel> BBB: that would be my motivation as well, but i also dont have any delusions of that going well [19:41:24 CET] <kierank> I will but we need people who stay silent like michaelni and ubitux to do what is best for the community [19:42:20 CET] <ubitux> i highly doubt carl is the reason libav doesn't want to merge [19:42:21 CET] <durandal_1707> also when is next meeting going to happen? [19:42:44 CET] <ubitux> just like michael wasn't really the reason of not merging back either [19:42:53 CET] <kierank> ubitux: irrespective of the merge he is the cancer of our community [19:42:57 CET] <jamrial> ubitux: his attitude certainly doesn't help [19:43:18 CET] <kierank> Creating a hostile environment does not attract new devs [19:43:19 CET] <ubitux> i'll stay neutral [19:43:19 CET] <jamrial> carl's i mean [19:43:36 CET] <kierank> ubitux: and so you are part of the problem then [19:43:39 CET] <BBB> carls behaviour at VDD was not exactly appreciated, to say it softly [19:44:04 CET] <durandal_1707> what he did? [19:44:07 CET] <BBB> they dont care as much about michaels [19:44:23 CET] <BBB> as in, they can live with it, its not that they like it but it wont kill them to accept it [19:44:29 CET] <BBB> carls behaviour is a hard blocker [19:44:52 CET] <llogan> i appreciate some of the work he does on trac [19:45:03 CET] <BBB> carl does not take criticisim or constructive suggestions to behavioural changes well, and tends to instead subtly insult the person asking him questions instead [19:45:19 CET] <ubitux> kierank: i'm fine having him around, and i don't consider him an hindrance to the project; that's my own judgement and you're free to disagree, but i won't go into the witch hunting [19:45:21 CET] <BBB> the work he does on trac should not require his destructive behaviour towards libav users or developers [19:45:26 CET] <BBB> I dont see how the two are connected [19:45:41 CET] <kierank> ^this [19:45:44 CET] <Compn> i dont think theres anyone on either project who cause too much trouble [19:45:49 CET] <BBB> I could perfectly well imagine him continuing to write patches and do trac work, but without insulting libav users or developers [19:46:41 CET] <kierank> ubitux: it is the attitude of saying "ah but he works on trac and sends patches which makes his insults ok" which is disgusting [19:46:51 CET] <durandal_1707> Don doesn't want to leave admin of some ffmpeg commits stats web page... [19:47:01 CET] <kierank> I have had this argument with atomnuker repeatedly as well [19:47:23 CET] <ubitux> kierank: i'm not OK with the insults, but i don't want to be involved in social issues [19:47:39 CET] <Daemon404> and thats why this is doomed to fail [19:47:42 CET] <wm4> note that the resistance against cleanup isn't going to attract Libav either [19:48:06 CET] <ubitux> it's not like i'm going to stop you from doing police work [19:48:10 CET] <wm4> (although they'll probably remain stupidly stubborn on this anyway) [19:48:11 CET] <kierank> ubitux: then by extension you approve Carl's behaviour [19:48:18 CET] <kierank> Whether you like it or not [19:48:21 CET] <Daemon404> ^ [19:48:26 CET] <jamrial> BBB: he wont change until he sees some real consequences for his toxic behavior. so far in his mind he seems to think the criticism is just a vocal minority and that the fact nothing bad happens ever is because of an implicit overall approval from the project [19:48:45 CET] <ubitux> kierank: and i also approuve you doing the witch hunting which i do not want to be part of [19:48:58 CET] <durandal_1707> insults can't be tolerated [19:49:11 CET] <ubitux> something happened with him again btw? [19:49:13 CET] <atomnuker> kierank: I did neversay it's okay [19:49:19 CET] <BBB> jamrial: probably true, yes& are you suggesting I get back into policing? [19:49:21 CET] <ubitux> or it's just the usual circle jerk? :) [19:49:32 CET] <kierank> ubitux: so it's fine to have racists, sexists, homophobes in FFmpeg just as long as they write patches? [19:50:11 CET] <ubitux> could be ok yes [19:50:21 CET] <kierank> Then your views are abhorrent [19:50:53 CET] <ubitux> tsss. [19:50:59 CET] <llogan> i think i would have remembered racist stuff... [19:51:08 CET] <llogan> etc. [19:51:10 CET] <ubitux> we was just doing an analogy [19:51:18 CET] <ubitux> about tolerating the intolerable [19:51:22 CET] <durandal_1707> ubitux: he was dick on irc, remmember? [19:51:39 CET] <Compn> the thing is, if it was just get rid of carl to merge the projects we would have done that already [19:51:44 CET] <ubitux> yes? not like everyone has been without his flaws here [19:52:05 CET] <ubitux> but again, why are we suddenly talking about carl? [19:52:36 CET] <jamrial> because we started talking about libav and a potential merge [19:52:52 CET] <jamrial> and the reasons it hasn't happened yet surfaced. hence carl [19:52:56 CET] <ubitux> and so we needed a culprit about the situation? [19:53:03 CET] <ubitux> which led to this stupid drama [19:53:04 CET] <Compn> ubitux : there must be a scapegoat [19:53:06 CET] <ubitux> while nothing happened? [19:53:09 CET] <Compn> right [19:53:22 CET] <Compn> nothing happened, no one talked about merging [19:53:33 CET] <Compn> just kierank and daemon dont like carl [19:53:36 CET] <ubitux> it's hard to believe there is so much emotional shit still hanging around [19:55:08 CET] <llogan> so, how about that sports team? [19:55:18 CET] <durandal_1707> no, everything happened, who can summon irc meeting? [19:56:39 CET] <durandal_1707> ubitux: I wouldnt call it emotional shit [19:57:04 CET] <ubitux> then what's with this sudden witch hunting and rage quits? [19:57:43 CET] <durandal_1707> you said it all [19:57:58 CET] <ubitux> then i got accused of not helping and being the reason of the drama [19:58:08 CET] <llogan> burn him! [19:58:14 CET] <ubitux> i can't call this being rational [19:58:44 CET] <durandal_1707> writting patches and spreading shit is bad [19:59:04 CET] <jamrial> kierank kinda overreacted, i agree with you in that regard [19:59:54 CET] <ubitux> anyway, sure for an irc meeting [20:00:00 CET] <ubitux> what do you want to discuss? [20:00:09 CET] <ubitux> aside from kicking out carl [20:00:12 CET] <ubitux> for no reason [20:00:14 CET] <ubitux> :D [20:00:21 CET] <durandal_1707> you don't want to have someone extremist to be part of project [20:00:33 CET] <Compn> are you calling carl extremist? [20:00:57 CET] <ubitux> i really think it doesn't matter to have a despisable person in the projet [20:01:05 CET] <ubitux> BUT it shouldn't be expressed on the project ground [20:01:15 CET] <durandal_1707> no, but he is certainly problematic personality [20:01:21 CET] <ubitux> like, it's pretty ok if carl has his hate twitter stream [20:01:28 CET] <ubitux> but doesn't express his view on the trac and mailing list [20:02:01 CET] <durandal_1707> hmm, that is questionable [20:02:14 CET] <michaelni> "so it's fine to have racists, sexists, homophobes in FFmpeg just as long as they write patches?" <-- if theres any racism, sexism, or gender/sexual preferance based attacks, please step in and do something about it, talk with the people involved and make it clear to them that this is not acceptable, also if that fails inform one of the maillist admins or irc ops about it [20:02:23 CET] <ubitux> anyway, i'm obviously ok with punishment as part of actions within ffmpeg realm [20:02:54 CET] <ubitux> (i won't do it personally because i don't feel like being too much involved in this, but feel free to) [20:03:23 CET] <phh> v bv [20:03:28 CET] <phh> fail [20:04:10 CET] <llogan> phh: VBV can be achieved with -maxrate and -bufsize [20:10:36 CET] <llogan> due to spam the cvslog entries for web commits may be delayed because messages "from" ffmpeg-cvslog are not set to go to queue until i make an appropriate filter [20:10:50 CET] <llogan> *are set to go to queue [20:15:44 CET] <Compn> i mean [20:15:51 CET] <Compn> kierank and daemon should just say it [20:15:54 CET] <Compn> they go or carl goes [20:16:35 CET] <durandal_1707> why coreimage filter have m extension? [20:16:52 CET] <nevcairiel> because its objective c [20:19:20 CET] <BBB> Compn: I dont think they want carl to go [20:19:41 CET] <BBB> Compn: I think they would like carl to be quiet when it comes to libav matters or responding to libav users or devs [20:23:03 CET] <Compn> maybe i misread then [20:54:28 CET] <BBB> Compn: we all agree carl is very valuable and does lots of good stuff, like trac work and patches to address a whole bunch of bugs that usually come from user reports [20:54:50 CET] <BBB> Compn: all of that is very valuable and helpful and we dont want to lose that, just like we dont want to lose the libav contributors by killing libav [20:56:20 CET] <Compn> i would love to find a solution [20:56:31 CET] <Compn> maybe people just dont want to work with each other [20:56:34 CET] <Compn> cant change that BBB [20:57:16 CET] <BBB> we can try :) Im not that pessimistic [20:57:31 CET] <BBB> lots of people tell me you cant write a vp9 decoder that is faster than the fastest vp8 decoder [20:57:41 CET] <BBB> lots of things cant be done according to naysayers [20:57:43 CET] <BBB> Im a yaysayer [20:58:04 CET] <BBB> its easy to say cant be done, but heroes are born when you do these thigns anyway [20:58:09 CET] <BBB> We Can Do This [20:58:22 CET] <wm4> no we can't! [20:59:03 CET] Action: llogan signs up for BBB's motivational speech series, "How to Collaborate Powerfully" [20:59:15 CET] <BBB> \o/ [20:59:19 CET] <BBB> my first subscriber ever [20:59:29 CET] Action: BBB awards llogan a plaque for being the first BBB follower ever [21:11:07 CET] <llogan> does anyone here know anyone what actually uses ffserver? [21:11:56 CET] <JEEB> quite a bit of people on #ffmpeg are trying to use it, which most community members seem to try and not have happen as it has been noticed that almost nobody uses it from the core group [21:12:35 CET] <JEEB> and in most cases you have better alternatives, such as nginx-rtmp (which takes in rtmp and delivers HLS/DASH with its NIH'd muxer to the clients) [21:13:20 CET] <wm4> "trying to use it" [21:13:22 CET] <wm4> I love that [21:13:30 CET] <JEEB> yes [21:13:34 CET] <wm4> it's like ffserver has masochistic devs behind it [21:13:36 CET] <wm4> err [21:13:38 CET] <wm4> sadistic [21:13:42 CET] <wm4> S&M devs [21:14:19 CET] <llogan> i'm considering a survey on the ML asking if anyone actually uses it (successfully). [21:14:25 CET] <wm4> not like ffmpeg CLI is much nicer (just look what at the result if you naively try to transcode a file, if you can even guess the syntax) [21:14:51 CET] <J_Darnley> At least people can tell you how to use ffmpeg. [21:15:04 CET] <JEEB> basically <join #ffmpeg> -> "hi I just found ffserver but I can't wrap myself around it" -> "uhh, hi. the thing is barely if at all maintained. unless you think you can start maintaining it, please just try using ffmpeg and maybe an actual separate streaming server such as *cast or nginx-rtmp instead" [21:15:07 CET] <TD-Linux> the problem is with ffserver, no one can tell you how to use the alternatives either [21:15:20 CET] <J_Darnley> People who know how to use ffserver don't exist or want to keep the secret sauce to themselves. [21:15:41 CET] <JEEB> yeah, I've seen people note that they use ffserver in production, but it's definitely some sort of black arts [21:15:47 CET] <wm4> why don't the ffserver defenders try to write a good ffmpeg-based streaming server? [21:15:57 CET] <wm4> to replace it [21:15:58 CET] <nevcairiel> because thats effort [21:16:04 CET] <JEEB> probably because ffserver already exists and might be doing what they need :D [21:16:05 CET] <J_Darnley> Because the competition might use it! [21:16:23 CET] <llogan> or if i submit a patch to prune it, will any actual users defend it? [21:16:43 CET] <JEEB> probably not, just people who have heard of people using it [21:16:51 CET] <TD-Linux> llogan, it would be nice to make a mapping of all of ffserver's features to other open source projects [21:16:54 CET] <wm4> oh fuck ganesh going on about intrinsics and inline asm again [21:17:08 CET] <JEEB> or if you're lucky, that one guy who posted patch or so a few months ago [21:17:24 CET] <llogan> TD-Linux: is such a thing even possiburu? does anyone actually know ffserver's features? [21:17:44 CET] <TD-Linux> llogan, that's the point of the exercise in making the mapping :) [21:17:51 CET] <JEEB> wm4: if he wants to optimize stuff just tell him to start conversion of the openhevc crap into proper asm :V [21:17:56 CET] <JEEB> will keep him busy for a while [21:18:57 CET] <wm4> he doesn't want to write yasm [21:19:04 CET] <JEEB> unsurprising [21:19:10 CET] <wm4> for some reason he thinks inline asm is better [21:19:35 CET] <JEEB> intrinsics I can kind of understand, those are shared by a lot of compilers and are supported for both 32bit and 64bit [21:19:47 CET] <JEEB> inline asm for archs that don't need to have that... eww [21:20:03 CET] <ethe> is yasm much better than intrinsics? [21:20:23 CET] <J_Darnley> I think so [21:20:24 CET] <TD-Linux> in general yes, because you can do sane register allocation [21:20:36 CET] <J_Darnley> 1 instruction for movd rather than a dozen functions [21:21:01 CET] <ethe> ah right. yeah, that would make sense. [21:21:31 CET] <TD-Linux> though for some things I've been able to make intrinsics that compile identically to hand written assembly [21:21:34 CET] <JEEB> of course, as can be seen by the openhevc stuff, you can still gain nicely by doing intrinsics [21:22:03 CET] <JEEB> you will get some extra from making it proper asm, but at that point it becomes a case of "is it worth it time-wise" [21:23:11 CET] <JEEB> nev seems to be maintaining the openhevc patches from when they last tried to kind of upstream things in his fork so they could be built with latest FFmpeg [21:24:09 CET] <wm4> and everyone uses them [21:24:41 CET] <JEEB> yeah, since they enable much faster playback - even though they are intrinsics [21:38:28 CET] <jamrial> it's not so much "getting a bit more performance than intrinsics" as it's "making sure we're not dependant on the compiler doing a good job" [21:39:03 CET] <JEEB> true that [22:02:51 CET] <nevcairiel> this thilo guy reads like an argument with carl, just less bad english [22:09:01 CET] <nevcairiel> and of course noone else will ever say anything to the topic, adn the guy never ever see that his behavior wasnt quite perfect [22:14:35 CET] <cone-057> ffmpeg 03Paul B Mahol 07master:e2298b3fccde: avcodec/jpeg2000dec: account two last bytes from end of bytestream as EOC marker [22:14:35 CET] <cone-057> ffmpeg 03Paul B Mahol 07master:585cfabb799d: avcodec/jpeg2000dec: add slice threading support [22:16:27 CET] <wm4> nevcairiel: there's an ease of contributing vs. code quality trade-off [22:17:07 CET] <wm4> I'm not sure where it should be, and also the requirements seem to be quite inconsistent in ffmpeg [22:17:50 CET] <ethe> Timothy_Gu: could I get a link to that gist again (the one with the asm explained), I'd like to convert the rest of the inline to yasm? (but I lost the patch because I had to reimage) [22:18:01 CET] <nevcairiel> getting changes to a public API reviewed and OKed isnt an outrageous requirement [22:18:07 CET] <ubitux> llogan: pretty sure the ffserver users will not answer to the survey [22:18:30 CET] <ethe> remove it and see how long it takes people to notice [22:19:10 CET] <ubitux> pretty sure you can apply this logic to about 70% of the code [22:19:20 CET] <ubitux> and no one will notice until about 1 month to a year [22:19:33 CET] <ubitux> that doesn't mean we should do that [22:19:49 CET] <nevcairiel> how obnoxious ffserver has been, i bet it causes complaints within a week [22:20:01 CET] <Daemon404> nobogy using ffserver SHOULD be using it [22:20:07 CET] <Daemon404> it's the best solution to 0 problems [22:20:17 CET] <ubitux> the thing is, ffserver users aren't using ffmpeg/master [22:20:24 CET] <ubitux> they're with their centos etc [22:20:37 CET] <nevcairiel> so, ffmpeg 0.8? [22:20:38 CET] <ubitux> so unless you get phoronixed, no one will tell [22:20:42 CET] <jamrial> so they will not care if ffserver is gone in 3.1? :p [22:21:06 CET] <Daemon404> why dont you ask the "maintainer" to make it kosher with tep2, or rm it [22:21:09 CET] <Daemon404> end of story [22:21:15 CET] <Daemon404> if it cant be done, rm it. [22:21:21 CET] <ubitux> is it blocking tep2? [22:21:27 CET] <nevcairiel> who knows [22:21:31 CET] <ubitux> if it's just vomitting warning it doesn't matter [22:21:32 CET] <nevcairiel> i have no idea how it works [22:21:39 CET] <Daemon404> ubitux, it will break when stuff deprecated in tep2 is removed [22:21:44 CET] <Daemon404> it uses ffm to serialize an avctx or some shit [22:21:45 CET] <nevcairiel> thats 2 years [22:21:51 CET] <ubitux> that ^ [22:21:51 CET] <Daemon404> and uses it to "communicate" [22:22:06 CET] <ubitux> so it will automatically be deleted in 2 years [22:22:11 CET] <ubitux> unless the maintainer does his job [22:22:11 CET] <nevcairiel> but i dont know if it starts breaking to some degree if ffmpeg.c uses codecpar instead of codec [22:22:15 CET] <Daemon404> ubitux, sure ok [22:22:19 CET] <Daemon404> thats the same deal :P [22:22:29 CET] <nevcairiel> i have no idea how ffserver interacts with avformat [22:22:33 CET] <ubitux> less aggressive than dropping it in a merge [22:22:35 CET] <ubitux> ;) [22:22:37 CET] <nevcairiel> is it self-contained, or does it use ffmpeg.c? [22:22:43 CET] <wm4> I don't even understand what ffserver does [22:23:04 CET] <nevcairiel> or do you use ffmpeg.c on the clientside of a ffserver? [22:23:05 CET] <JEEB> it's weird, you have ffm files yet often you see people also using ffmpeg with it [22:23:19 CET] <ubitux> wm4: it serves feed which you push to it with ffmpeg only, because it's abusing a weird cache file [22:23:20 CET] <Daemon404> it uses ffm to configure the other side [22:23:27 CET] <TD-Linux> on my TODO list for a while has been to set up a webcam using the various open source live streamers so I can actually recommend one to people [22:23:42 CET] <Daemon404> TD-Linux, probably OBS is your best bet [22:23:44 CET] <TD-Linux> (other than icecast) [22:23:45 CET] <nevcairiel> if it uses all these crazy properties in the ffm, then those will no longer be communicated to the decoder after tep2 if it uses ffmpeg.c [22:23:46 CET] <Daemon404> lots of people on twitch use it [22:23:49 CET] <nevcairiel> so who knows what happens [22:24:06 CET] <TD-Linux> Daemon404, but OBS is only a source client right? [22:24:13 CET] <Daemon404> yeah [22:24:23 CET] <Daemon404> i bet vlc can view it though :P [22:24:28 CET] <Daemon404> not that that is ag reat soluton [22:24:31 CET] <Daemon404> solution* [22:24:41 CET] <JEEB> OBS can push to something like nginx-rtmp tho [22:24:43 CET] <TD-Linux> I was thinking of things more like Icecast but for HLS or DASH [22:24:52 CET] <nevcairiel> i used obs with nginx-rtmp last year [22:24:53 CET] <nevcairiel> worked ok [22:24:59 CET] <Daemon404> TD-Linux, then you want to do what jeeb said [22:25:06 CET] <Daemon404> nginx has modules for dash and hls and rtmp [22:25:11 CET] <Daemon404> iirc obs can feed this [22:25:15 CET] <nevcairiel> it can [22:25:23 CET] <TD-Linux> ok I'll try out nginx-rtmp and see how it goes [22:25:32 CET] <nevcairiel> dash didnt really work for me with nginx, but i didnt spend too much time on it [22:25:39 CET] <nevcairiel> just used rtmp since i had only a small client base [22:25:39 CET] <JEEB> it has some NIH'd muxer [22:25:48 CET] <JEEB> for both HLS and DASH output [22:25:58 CET] <Daemon404> the hls muxer was pretty decent iirc [22:26:01 CET] <Daemon404> minimal mpegts overhead [22:26:09 CET] <Daemon404> a certain large cdn uses it [22:26:12 CET] <Daemon404> (not akamai) [22:27:47 CET] <TD-Linux> looks like it lacks a webm muxer as well [22:28:31 CET] <Daemon404> because nobody cares about streaming webm live [22:28:32 CET] <kierank> what do we need faac for? [22:28:33 CET] <Daemon404> except youtube [22:28:38 CET] <Daemon404> kierank, we dont [22:28:45 CET] <ubitux> durandal_1707: zoneplate demo somewhere? [22:28:46 CET] <kierank> ok that's getting choppe [22:28:53 CET] <J_Darnley> heaac, no? [22:29:02 CET] <JEEB> faac has no he-aac [22:29:15 CET] <J_Darnley> oh, is that fdk then? [22:29:17 CET] <JEEB> fdk-aac and already removed libaacplus were the only he-aac encoders [22:29:22 CET] <JEEB> ye [22:29:30 CET] <Daemon404> fdk isnt gonna be removed any time soon [22:29:44 CET] <JEEB> yeah, and it shouldn't be until we get a good HE-AAC encoder [22:29:58 CET] <nevcairiel> its also still better at LC [22:30:10 CET] <nevcairiel> although claudio is working on more improvements [22:30:11 CET] <wm4> Daemon404: that's sad, webm should be relatively nice for streaming? [22:30:23 CET] <wm4> although no reason not to use mpegts I guess [22:30:27 CET] <durandal_1707> ubitux: its moving circles around [22:30:41 CET] <ubitux> what's the use case? [22:31:00 CET] <Daemon404> wm4, nice in what way [22:31:14 CET] <Daemon404> it's webm dash. [22:31:17 CET] <durandal_1707> kierank needs it, look at yooutube video's [22:31:27 CET] <ubitux> url? [22:31:29 CET] <wm4> Daemon404: I mean as opposed to fragmented crap [22:31:37 CET] <Daemon404> wm4, webm *dash* [22:31:40 CET] <Daemon404> it IS fragmented [22:31:48 CET] <wm4> yeah, I didn't get that before [22:33:35 CET] <durandal_1707> ubitux: on phone now, for usecase search 4k zoneplate on YouTube [22:34:57 CET] <ubitux> ok thx [22:35:06 CET] <jamrial> ubitux: as kierank posted above https://www.youtube.com/watch?v=aRqkGs0o6h0 [22:35:18 CET] <ubitux> i missed that, thx [22:35:34 CET] <ubitux> is this a stress test for wavelet codecs? [22:36:03 CET] <TD-Linux> Daemon404, webm fragements aren't really 1:1 comparable to mp4 fragmenting though [22:36:09 CET] <ubitux> i wonder what kind of artifacts it raises [22:36:59 CET] <TD-Linux> as all the stuff it uses was already there (cue points) [23:25:11 CET] <llogan> kierank: i don't see very many users using libfaac anymore [23:25:37 CET] <kierank> I see [23:35:22 CET] <Compn> kierank / Daemon404 : so did you want carl to leave or carl to stop harassing libav and users? i wasnt clear on what you were saying. [23:35:41 CET] <llogan> here we go again... [23:42:56 CET] <BtbN> Would it be worthwile merging vf_hwupload and vf_hwdownload from libav? Or is that about to happen anyway, and the merges just haven't got that far yet? [23:43:40 CET] <BtbN> https://git.libav.org/?p=libav.git;a=commit;h=07a844f32ebb78503981df017fa3e… it seems to be rather new [23:44:18 CET] <Daemon404> BtbN, all new merges are waiting on the tep2 work [23:44:29 CET] <BtbN> ah [23:44:29 CET] <JEEB> so feel free to contribute to the TEP2 effort [23:44:32 CET] <JEEB> :) [23:44:35 CET] <BtbN> What even is that? [23:45:39 CET] <BtbN> I'm thinking about writing a NVCUVID based h264 decoder, and that automated CUDA hwframe up/downloading would come very handy there. [23:54:16 CET] <BtbN> Uhm, isn't that check incomplete? [23:54:17 CET] <BtbN> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=configure;h=feb0bc287bc8c7e6… [23:54:36 CET] <BtbN> Or does check_lib automaticaly fail/disable cuda somehow? [23:57:08 CET] <BtbN> Yeah, it is. --enable-cuda works even on systems without CUDA. [23:57:26 CET] <nevcairiel> should fail to compile i guess [23:57:47 CET] <BtbN> Yeah, it fails to build, but shouldn't configure fail already? What else is the point of that check_lib call? [23:59:41 CET] <BtbN> https://gist.github.com/ccf56f8a31fbe004d1b2 like this [00:00:00 CET] --- Sat Mar 26 2016

1 0

ffmpeg.log.20160325
by burek 26 Mar '16

26 Mar '16

[00:00:09 CET] <J_Darnley> Do editors have to call every operation "merge"? [00:00:30 CET] <J_Darnley> Granted I can't think of a better verb for this. [00:00:34 CET] <whoever> hi all, I am tring to use ffmpeg to convert .mkv to mp4 and i am getting driver fail [00:01:11 CET] <J_Darnley> "getting driver fail" is not an error message [00:01:29 CET] <J_Darnley> Please put your whole command and the output on pastebin. [00:03:06 CET] <Wader8> I never liked 16:9 and I probably never will [00:03:22 CET] <J_Darnley> Well, I agree with you there [00:03:34 CET] <Wader8> they just took the top and bottom off and put it on the sides, what a joke [00:05:37 CET] <Wader8> 16:10 I heard is actually also physically more correct as it's closer to the golden ratio, but I'm not sure why aren't the gurus more outspoken, the TV industry kinda has too much influence on the PC market, the only thing left they can do is to force HDMI to displace DP [00:05:49 CET] <whoever> https://bpaste.net/show/d62348a90d57 [00:06:12 CET] Action: J_Darnley thinks you should be stoned [00:06:25 CET] <whoever> the video play in its native format (befor attempting to convert) [00:07:06 CET] <whoever> whys that [00:07:13 CET] <J_Darnley> > mencoder [00:07:31 CET] <J_Darnley> It's ${CURRENT_YEAR}! [00:07:39 CET] <drv> year of mencoder on the desktop [00:07:43 CET] <whoever> well isnt mencoder part of ffmgeg [00:08:07 CET] <J_Darnley> huehuehue [00:08:14 CET] <J_Darnley> this guy [00:08:31 CET] <DHE> I thought mencoder was part of mplayer [00:09:01 CET] <whoever> with ffmeg i get unknown encoder libfaac [00:09:55 CET] <llogan> "ffmpeg version svn-57" [00:10:05 CET] <llogan> copyright 1974 [00:16:59 CET] <whoever> http://pastebin.com/itRumRGT and it doesn' seem to matter what I change my lib to, ie: libaac. i seem to always get unknown encoder [00:17:30 CET] <J_Darnley> No doubt because it wasn't compiled with it. [00:17:44 CET] <J_Darnley> Which we would know if you would follow instructions. [00:18:06 CET] <J_Darnley> and "libaac" doesn't exist [00:18:32 CET] <J_Darnley> just use "aac" and you should be fine. [00:18:34 CET] <whoever> yes that is why i changed it [00:20:48 CET] <llogan> whoever: the complete console output is missing from your pastebin [00:27:38 CET] <whoever> llogan: thx [00:27:44 CET] <whoever> http://pastebin.com/guSsyaq6 [00:28:13 CET] Action: J_Darnley sighs [00:28:33 CET] <J_Darnley> It wasn't compiled with that encoder either. [00:28:42 CET] <J_Darnley> And its name would be libfdc_aac [00:29:35 CET] <J_Darnley> You could also update the version you're running [00:29:50 CET] <J_Darnley> Then you would get a file aac encoder built in [00:29:57 CET] <J_Darnley> *a fine aac encoder [00:30:18 CET] <whoever> J_Darnley: i got it from my distros repos [00:30:33 CET] <whoever> so i assumed it came with support for it [00:31:23 CET] <TD-Linux> ffmpeg isn't redistributable with libfdk_aac so it had better not come with it [00:32:56 CET] <whoever> J_Darnley: so do i have any option or do i need to compile from source [00:33:04 CET] <llogan> whoever: download a binary from here: http://johnvansickle.com/ffmpeg/ [00:33:26 CET] <llogan> then do: ffmpeg -i input -c:v copy -c:a aac output.mp4 [00:33:40 CET] <llogan> also, why do you want mp4? [00:33:58 CET] <whoever> so it will play on a dvd player [00:36:18 CET] <llogan> if the device doesn't like 5.1 AAC add "-ac 2" [00:37:19 CET] <llogan> you add -t 60 to encode a 60 second test segment. [00:37:22 CET] <llogan> *can add [00:38:25 CET] <AlexQ> Hi. I was having a problem yesterday when remuxing Matroska. I have encountered that problem again today: it seems to be related with copying SubRip streams ?!? : it causes the audio to be inaudible, even though the stream is there and can be extracted and played successfully by itself. Happens like that on multiple players [00:39:55 CET] <AlexQ> Any insight? [00:40:25 CET] <AlexQ> BTW. Can I somehow extract the .srt file while remuxing MKV? To have only one pass? [00:45:51 CET] <AlexQ> Okay, maybe I'll report it as a bug in the future or sth. For now extracting .srt before remuxing and using the subs from that separate file will do [00:46:38 CET] <J_Darnley> ffmpeg -i INPUT output.mkv output.srt [00:47:06 CET] <J_Darnley> As for your problems, how about an error message [00:53:38 CET] <AlexQ> Well, there are no error messages while encoding or null-decoding using ffmpeg - haven't tried with verbose or debug loglevels. And my quesion was if I could extract SubRip stream(s) to different files while remuxing MKV, thus effectively having 2 or more output files. [01:03:21 CET] <AlexQ> Is that even possible? [01:06:14 CET] <AlexQ> Okay,just multiple outputs then. And I will have a single demuxing pass, so very useful [01:06:40 CET] <AlexQ> Will try to come back with a sample and log tomorrow maybe; maybe I'll recompile ffmpeg with the newest git snapshot first [01:06:49 CET] <AlexQ> to check if that problem still appears [01:06:54 CET] <AlexQ> cheers and bye [02:51:42 CET] <explodes_> my call to sws_scale keeps crashing, can't figure out why. the source pixels is 32*1280*720 bytes long [02:52:37 CET] <explodes_> http://pastebin.com/dM1Jph5c [02:54:21 CET] <J_Darnley> I though data should be an array of 4 pointers [02:54:40 CET] <J_Darnley> perhaps the swscale api is different [02:55:11 CET] <explodes_> the pixels argument is a void* [02:55:42 CET] <explodes_> er, in avpicture_fill [02:55:47 CET] <explodes_> and, no it isnt [02:57:14 CET] <explodes_> compiling... [02:58:36 CET] <explodes_> still fails [02:58:55 CET] <explodes_> Rather, it still SIGSEGV's [03:08:05 CET] <explodes_> When I call sws_scale with a fresh calloc'd (32*1280*720, uint8_t), it doesnt crash at all.. Makes me think the bitmap I am pulling pixels from is not ready or something [03:08:23 CET] <explodes_> But the pixel pointer is not NULL, so I know it must have some pixels [03:11:22 CET] <explodes_> avpicture_fill succeeds with the bitmap's pixels, too. so I'm not sure what is happening.. [04:11:51 CET] <explodes_> Yea, I still cannot figure this out: http://pastebin.com/XVSM2d2h [04:40:18 CET] <whoever> hi all, i can't semm to convert a mkv to mp3 via command line, but i can do it with DEEVEEDEE [04:40:34 CET] <whoever> does anyone know what command deeveedee is excuting [04:44:51 CET] <whoever> $ ffmpeg -i The.Hunger.Games.Mockingjay.Part.2.2015.1080p.BluRay.x264.DTS-JYK.mkv -c:v copy -c:a aac output.mp3 [04:44:54 CET] <whoever> ffmpeg version N-69032-g3aaff80 Copyright (c) 2000-2015 the FFmpeg developers [04:44:57 CET] <whoever> built on Jan 12 2015 12:27:33 with gcc 4.9.2 (GCC) 20141224 (prerelease) [04:44:59 CET] <whoever> configuration: [04:45:02 CET] <whoever> libavutil 54. 16.100 / 54. 16.100 [04:45:04 CET] <whoever> libavcodec 56. 20.100 / 56. 20.100 [04:45:07 CET] <whoever> libavformat 56. 18.101 / 56. 18.101 [04:45:09 CET] <whoever> libavdevice 56. 4.100 / 56. 4.100 [04:45:12 CET] <whoever> libavfilter 5. 7.100 / 5. 7.100 [04:45:14 CET] <whoever> libswscale 3. 1.101 / 3. 1.101 [04:45:17 CET] <whoever> libswresample 1. 1.100 / 1. 1.100 [04:45:19 CET] <whoever> Input #0, matroska,webm, from 'The.Hunger.Games.Mockingjay.Part.2.2015.1080p.BluRay.x264.DTS-JYK.mkv': [04:45:22 CET] <whoever> Metadata: [04:45:22 CET] <c_14> did you even read anything that was written there? [04:45:25 CET] <whoever> encoder : libebml v1.3.3 + libmatroska v1.4.4 [04:45:27 CET] <whoever> creation_time : 2016-03-11 22:24:04 [04:45:30 CET] <whoever> Duration: 02:16:57.26, start: 0.000000, bitrate: 3607 kb/s [04:45:30 CET] <furq> this could go on for a while [04:45:32 CET] <whoever> Chapter #0:0: start 0.000000, end 459.041000 [04:45:35 CET] <whoever> Metadata: [04:45:37 CET] <whoever> title : Chapter 01 [04:45:40 CET] <whoever> Chapter #0:1: start 459.041000, end 1009.216000 [04:45:42 CET] <whoever> Metadata: [04:45:45 CET] <whoever> title : Chapter 02 [04:45:47 CET] <whoever> Chapter #0:2: start 1009.216000, end 1486.985000 [04:45:50 CET] <whoever> Metadata: [04:45:52 CET] <whoever> title : Chapter 03 [04:45:55 CET] <whoever> Chapter #0:3: start 1486.985000, end 1919.959000 [04:45:57 CET] <whoever> Metadata: [04:46:00 CET] <whoever> title : Chapter 04 [04:46:02 CET] <whoever> Chapter #0:4: start 1919.959000, end 2231.687000 [04:46:05 CET] <whoever> Metadata: [04:46:07 CET] <whoever> title : Chapter 05 [04:46:10 CET] <whoever> Chapter #0:5: start 2231.687000, end 2802.716000 [04:46:12 CET] <whoever> Metadata: [04:46:15 CET] <whoever> title : Chapter 06 [04:46:17 CET] <whoever> Chapter #0:6: start 2802.716000, end 3300.880000 [04:46:20 CET] <whoever> Metadata: [04:46:22 CET] <whoever> title : Chapter 07 [04:46:23 CET] <relaxed> no [04:46:25 CET] <whoever> Chapter #0:7: start 3300.880000, end 3766.429000 [04:46:27 CET] <whoever> Metadata: [04:46:30 CET] <whoever> title : Chapter 08 [04:46:31 CET] <furq> oh christ look how many chapters there are [04:46:32 CET] <whoever> Chapter #0:8: start 3766.429000, end 4167.496000 [04:46:35 CET] <whoever> Metadata: [04:46:37 CET] <whoever> title : Chapter 09 [04:46:40 CET] <whoever> Chapter #0:9: start 4167.496000, end 4722.968000 [04:46:42 CET] <furq> we're going to be here all week [04:46:42 CET] <whoever> Metadata: [04:46:45 CET] <whoever> title : Chapter 10 [04:46:47 CET] <whoever> Chapter #0:10: start 4722.968000, end 5391.135000 [04:46:50 CET] <whoever> Metadata: [04:46:52 CET] <whoever> title : Chapter 11 [04:46:55 CET] <whoever> Chapter #0:11: start 5391.135000, end 5828.239000 [04:46:57 CET] <whoever> Metadata: [04:47:00 CET] <whoever> title : Chapter 12 [04:47:02 CET] <whoever> Chapter #0:12: start 5828.239000, end 6541.785000 [04:47:05 CET] <whoever> Metadata: [04:47:07 CET] <whoever> title : Chapter 13 [04:47:10 CET] <whoever> Chapter #0:13: start 6541.785000, end 7015.758000 [04:47:12 CET] <whoever> Metadata: [04:47:15 CET] <whoever> title : Chapter 14 [04:47:17 CET] <whoever> Chapter #0:14: start 7015.758000, end 7413.239000 [04:47:20 CET] <whoever> Metadata: [04:47:22 CET] <whoever> title : Chapter 15 [04:47:25 CET] <whoever> Chapter #0:15: start 7413.239000, end 8217.259000 [04:47:27 CET] <whoever> Metadata: [04:47:30 CET] <whoever> title : Chapter 16 [04:47:32 CET] <whoever> Stream #0:0(eng): Video: h264 (High), yuv420p, 1920x800 [SAR 1:1 DAR 12:5], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) [04:47:35 CET] <whoever> Metadata: [04:47:38 CET] <whoever> BPS : 2850290 [04:47:40 CET] <whoever> BPS-eng : 2850290 [04:47:43 CET] <whoever> DURATION : 02:16:57.251000000 [04:47:45 CET] <whoever> DURATION-eng : 02:16:57.251000000 [04:47:48 CET] <whoever> NUMBER_OF_FRAMES: 197017 [04:47:50 CET] <whoever> NUMBER_OF_FRAMES-eng: 197017 [04:47:53 CET] <whoever> NUMBER_OF_BYTES : 2927694006 [04:47:55 CET] <whoever> NUMBER_OF_BYTES-eng: 2927694006 [04:47:58 CET] <whoever> _STATISTICS_WRITING_APP: mkvmerge v8.5.2 ('Crosses') 32bit [04:48:00 CET] <whoever> _STATISTICS_WRITING_APP-eng: mkvmerge v8.5.2 ('Crosses') 32bit [04:48:03 CET] <whoever> _STATISTICS_WRITING_DATE_UTC: 2016-03-11 22:24:04 [04:48:05 CET] <whoever> _STATISTICS_WRITING_DATE_UTC-eng: 2016-03-11 22:24:04 [04:48:08 CET] <whoever> _STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES [04:48:10 CET] <whoever> _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES [04:48:13 CET] <whoever> Stream #0:1(eng): Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 768 kb/s (default) [04:48:16 CET] <whoever> Metadata: [04:48:18 CET] <whoever> BPS : 754499 [04:48:21 CET] <whoever> BPS-eng : 754499 [04:48:23 CET] <whoever> DURATION : 02:16:57.259000000 [04:48:26 CET] <whoever> DURATION-eng : 02:16:57.259000000 [04:48:28 CET] <whoever> NUMBER_OF_FRAMES: 770368 [04:48:31 CET] <whoever> NUMBER_OF_FRAMES-eng: 770368 [04:48:33 CET] <whoever> NUMBER_OF_BYTES : 774990208 [04:48:36 CET] <whoever> NUMBER_OF_BYTES-eng: 774990208 [04:48:38 CET] <whoever> _STATISTICS_WRITING_APP: mkvmerge v8.5.2 ('Crosses') 32bit [04:48:41 CET] <whoever> _STATISTICS_WRITING_APP-eng: mkvmerge v8.5.2 ('Crosses') 32bit [04:48:43 CET] <whoever> _STATISTICS_WRITING_DATE_UTC: 2016-03-11 22:24:04 [04:48:46 CET] <whoever> _STATISTICS_WRITING_DATE_UTC-eng: 2016-03-11 22:24:04 [04:48:48 CET] <whoever> _STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES [04:48:51 CET] <whoever> _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES [04:48:53 CET] <whoever> File 'output.mp3' already exists. Overwrite ? [y/N] y [04:48:56 CET] <whoever> [aac @ 0x2736e80] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it. [04:48:59 CET] <whoever> damnit [04:49:01 CET] <whoever> that was supose to be contrl k not c [04:49:04 CET] <whoever> sorry [04:49:06 CET] <whoever> it says aac is not supported but the command i used did not specify an audio codec [04:49:09 CET] <whoever> ffmpeg -i The.Hunger.Games.Mockingjay.Part.2.2015.1080p.BluRay.x264.DTS-JYK.mkv -c:v copy -c:a aac output.mp3 [04:49:12 CET] <whoever> oops and so i do [04:49:15 CET] <whoever> i might want to learn how to see straight [04:49:45 CET] <c_14> update your version of ffmpeg [04:49:46 CET] <furq> well this has been special [04:49:52 CET] <c_14> The internal encoder isn't experimental anymore [04:50:01 CET] <furq> yeah but he doesn't actually want to reencode so it doesn't matter [04:50:30 CET] <c_14> and next time, just quit your client (sigkill if needs be) and then rejoin [04:50:35 CET] <furq> or rather he doesn't want to reencode to aac [04:50:38 CET] <c_14> It's better than just letting it send out line after line for hours [04:51:13 CET] <furq> some clients will let you /part in the middle of a big paste [04:51:37 CET] <furq> those clients also tend to ask you to confirm a big paste though, so i doubt the one you're using does [05:33:40 CET] <whoever> http://pastebin.com/LVtyTg8A [05:33:56 CET] <whoever> still get same error as befoe [05:35:34 CET] <furq> that's probably because you're still running the same command as before [05:35:45 CET] <relaxed> whoever: ffmpeg -i input -map 0:a -c:a libmp3lame -q:a 0 out.mp3 [05:36:14 CET] <relaxed> add -ac 2 [05:36:21 CET] <furq> -i input -vn -q:a 0 -ac 2 out.mp3 [05:36:39 CET] <furq> you might not even need -vn, i forget [05:39:01 CET] <whoever> relaxed: why ac-2 [05:39:40 CET] <relaxed> because libmp3lame doesn't have 5.1 support, only stereo [05:42:26 CET] <whoever> running it without ac 2 for now , thx for the heads up [05:48:07 CET] <whoever> damnit, i picked the wrong ouput format, mcv,i am tring to convert to a format that will play on a dvd player. so this time i am tring avi [06:21:51 CET] <zosky_> hi yall. trying very unsuccessfully to concat a bunch of vids. its catching them as input (stream 0.0 to 0.9) ... but it only copies stream 0.0 & 0.1... what am i missing ? http://pastebin.com/zdxE8RPg [06:24:19 CET] <zosky_> sorry, should ask on #libav (was not super clear on the difference) [06:24:20 CET] <relaxed> zosky_: you're using avconv (#libav), not ffmpeg. Also, -i concat: will not work with mp4. [06:24:44 CET] <relaxed> zosky_: MP4Box -cat 1.mp4 -cat 2.mp4 -new combined.mp4 [06:25:07 CET] <zosky_> relaxed: thanks, will try [06:33:18 CET] <zosky_> cheers mate. all i needed was a nudge in the right direction. [09:04:04 CET] <_Raiz> wanna convert mp4 to ogg... [09:04:34 CET] <_Raiz> $ ffmpeg -i in.mp4 -codecs libvorbis out.ogg [09:04:49 CET] <_Raiz> doesn't work [09:05:15 CET] <ln-> "doesn't work" is not a problem description [09:05:38 CET] <_Raiz> ln-: my question is, what's the wrong with that command [09:05:40 CET] <_Raiz> I know it's wrong [09:06:04 CET] <_Raiz> there's no problem but I just have no idea how the command should be [09:06:14 CET] <ln-> i wish i knew [09:11:32 CET] <parrot1> I tried to link against ffmpeg libraries but I got error undefined reference to symbol 'avfilter_graph_alloc@@LIBAVFILTER_6' . Any idea? [09:44:24 CET] <lukesan> morning all [13:02:05 CET] <styler2go> Can i somehow get a stream of my desktop without overlaying it? [13:38:16 CET] <whoever> what format do i need to convert .mkv to so that it will play in dvd player , tried avi with /usr/bin/ffmpeg -i The.Hunger.Games.Mockingjay.Part.2.2015.1080p.BluRay.x264.DTS-JYK.mkv -map 0:a -c:a libmp3lame -q:a 0 out.avi [13:39:00 CET] <whoever> /usr/bin/ffmpeg -i The.Hunger.Games.Mockingjay.Part.2.2015.1080p.BluRay.x264.DTS-JYK.mkv -map 0:a -c:a libmp3lame -q:a 0 out.avi [13:39:13 CET] Last message repeated 2 time(s). [13:39:13 CET] <J_Darnley> Depends on what the DVD player supports other than a DVD [13:39:14 CET] <whoever> /usr/bin/ffmpeg -i The.Hunger.Games.Mockingjay.Part.2.2015.1080p.BluRay.x264.DTS-JYK.mkv -map 0:a -c:a libmp3lame -q:a 0 out.avi [13:39:47 CET] <yongyung> I've been using libx264 -preset ultrafast -qp 15 to encode lossless footage to something more manageable for editing, but sony vegas still has problems playing it back without lagging, especially when things like slow motion/speed ups come into play. Surprisingly (for me) this is actually CPU and not I/O bound (I have a 3570k(a)4.2GH, so not the worst CPU), do you guys have any advise as to which codec to use that sony vegas can handle better? [13:42:17 CET] <gnome1> whoever: worst case, DVD-Video [13:42:55 CET] <gnome1> whoever: first you really need to figure out which formats does it support [13:45:06 CET] <furq> whoever: if you want the video then get rid of -map 0:a [13:45:11 CET] <furq> and also ideally read the manual [15:04:57 CET] <whoever> if my video seems a little pixelated or cartoon looking after conversion would i increase the sample rate { [15:06:16 CET] <whoever> gnome1: i am tring avi , since even when you try to build a dvd most often it is converted to an avi, ten the menues are made and theh the whole thing is built [15:07:51 CET] <whoever> and furq i would , if i still had it from 15 years ago [15:15:59 CET] <cosmo1t> Is there a decent reference/tutorial for changing code that uses AVPicture? [15:16:32 CET] <cosmo1t> i see ont he main docs it has replacment functions it says to cxall (mostly avframe stuff) but the alloc and free have no referenc.e I'm assuming you are supposed to handle the buffers on your own or use another AVFrame? [15:23:07 CET] <JEEB> cosmo1t: I recommend looking at the examples under the docs [15:23:24 CET] <JEEB> https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples [15:23:37 CET] <JEEB> most of the common scenarios are there [15:23:48 CET] <JEEB> demuxing, decoding, encoding, filtering [15:24:14 CET] <JEEB> you can then see what applies to your use case to move from AVPicture to AVFrame (I guess?) [15:35:31 CET] <_Timon> lolwat [15:35:37 CET] <_Timon> taskkill cant kill ffmpeg.exe [15:35:51 CET] <_Timon> "received > 3 system signals, hard exiting" [15:35:56 CET] <_Timon> still running [15:39:10 CET] <_Timon> litterally nothing is working [15:39:16 CET] <_Timon> never had a process being this stuck on windows before [15:39:46 CET] <J_Darnley> Wow really? [15:39:58 CET] <J_Darnley> I'm always encountering shit I can't kill on Windows. [15:40:24 CET] <_Timon> well, not with the normal methods ofcourse [15:40:27 CET] <_Timon> that's all too common [15:48:07 CET] <martin5> hi all. I use ffmpeg by code in asp.net to conver audio files to mp3. [15:48:18 CET] <martin5> Dim strfichero As String [15:48:18 CET] <martin5> Dim mh As New MediaHandler [15:48:18 CET] <martin5> strfichero = Mid(fichero, InStrRev(fichero, "\") + 1) [15:48:18 CET] <martin5> mh.FFMPEGPath = Server.MapPath("Comun/Ffmpeg/ffmpeg.exe") [15:48:18 CET] <martin5> mh.InputPath = Left(fichero, Len(fichero) - Len(strfichero) - 1) [15:48:18 CET] <martin5> mh.OutputPath = Server.MapPath("Temporal") [15:48:18 CET] <martin5> mh.BackgroundProcessing = True [15:48:19 CET] <martin5> mh.FileName = strfichero [15:48:19 CET] <martin5> mh.OutputFileName = Left(strfichero, Len(strfichero) - 4) [15:48:20 CET] <martin5> mh.OutputExtension = ".mp3" [15:48:20 CET] <martin5> mh.Audio_Bitrate = 128 [15:48:21 CET] <martin5> mh.Process() [15:48:32 CET] <J_Darnley> ignored [15:49:46 CET] <JEEB> martin5: pasting code onto IRC is the best way to get on people's ignore list. congratulation. for future reference, use pastebin-like sites and link to your code rather than paste it [15:51:01 CET] <martin5> it converts to mp3, I can listen the song with a double click, but I can't listen to the song on my web (IE11, firefox, opera, safari, chrome). can u help me, please? [15:53:28 CET] <martin5> oks, thanks. I take note [15:53:32 CET] <martin5> sorry [16:14:31 CET] <gnome1> martin5: maybe it's a browser problem? how is it supposed to play on the browser? [16:19:27 CET] <cosmo1t> thanks jeeb wil do [16:24:30 CET] <cosmo1t> Jeeb thatnks, scaling_video.c has the exact deal. converting YUV420PO to RGB24 and has it's own data/linesize buffers. [16:24:50 CET] <cosmo1t> thanks for responding, i hadn't found those files evn though i have th source. appreciate it. [16:25:15 CET] <JEEB> I recommend looking at the zscale filter or even the base zimg library for better quality colorspace conversions [16:25:24 CET] <JEEB> https://github.com/sekrit-twc/zimg [16:26:00 CET] <JEEB> depends on your requirements of course [16:26:57 CET] <cosmo1t> yeah for sure. i'm just tryingot wack the deprecated warning from this ios app. but i will look @ that. [16:27:06 CET] <JEEB> oh [16:27:08 CET] <JEEB> iOS [16:27:24 CET] <cosmo1t> yippie.. [16:27:36 CET] <JEEB> yeah, I don't think zimg has any NEON asm [16:28:00 CET] <cosmo1t> i was just reading htat, it's targetd @ posix .. but can't hurt ot look [16:28:16 CET] <JEEB> well it builds for both windows and *nix-based things [16:28:33 CET] <cosmo1t> right... we'll see if it cna handle a x-compile. [16:28:36 CET] <JEEB> at least those are tested :p but it's rather not tested on ARM [16:28:45 CET] <cosmo1t> hehe [16:28:53 CET] <cosmo1t> i'll createa nother branch for that task. [16:29:05 CET] <JEEB> and even if it builds on ARM, it might be quite a bit slower because it has a lot of IA32/x86_64 asm [16:29:21 CET] <JEEB> same as with swscale, but I think swscale at least has some ARM asm >_> [16:29:49 CET] <cosmo1t> swscale has some arm .. just no hardware accell for yuv420->rgb.. only ppc and i386 for that [16:30:10 CET] <JEEB> hwaccel usually means dedicated hardware [16:30:18 CET] <JEEB> not SIMD [16:30:18 CET] <cosmo1t> well using the gpu right? [16:30:21 CET] <JEEB> no [16:30:25 CET] <JEEB> swscale uses no GPU [16:30:26 CET] <cosmo1t> ah n/mind then [16:30:32 CET] <cosmo1t> i didn't read enough into that [16:30:34 CET] <JEEB> swscale is fully CPU only [16:31:12 CET] <JEEB> usually player software has opengl shaders for the YCbCr->RGB conversion and scaling [16:32:56 CET] <cosmo1t> okay. thatnks for the feedback and such. just like antyhing, it's WAY more complicated after you peel back the first sheet [16:33:23 CET] <JEEB> also do remember the software license that FFmpeg is under [16:33:38 CET] <JEEB> if you compile by default, it's LGPL. if you hit the --enable-gpl switch it's GPL [16:34:02 CET] <JEEB> former requires you to make it possible to switch the library if the user so wants, and latter is the usual viral thing :P [16:35:44 CET] <cosmo1t> yeah we didnt' enable gpl. [16:35:55 CET] <JEEB> IANAL, but for shared libraries for LGPL people usually publish the exact source code they used for the thing, and with static linking object files for the non-open part and source code for FFmpeg. This way the library can be switched if the user wants that (of course you don't have to support such users, just making it possible is enough) [16:36:16 CET] <JEEB> the thing in the shared part being in this case FFmpeg [16:36:20 CET] <cosmo1t> right [16:36:37 CET] <cosmo1t> that would assume the preson that designed it separted those functions from main app. [16:36:46 CET] <cosmo1t> so youc ould swap a library. [16:37:12 CET] <JEEB> no idea, just make sure you're following the license and you should be OK [16:37:30 CET] <cosmo1t> yep they already did all that, this is like the 4th or 5th release. [16:37:42 CET] <cosmo1t> i did add a ticketot have it reviewed though [16:37:44 CET] <cosmo1t> thanks [16:38:13 CET] <JEEB> there's an unfortunate amount of people who just copypasta FFmpeg configuration without thinking about it [16:38:31 CET] <cosmo1t> yeah it can bite you .. [16:38:36 CET] <JEEB> > trying to publish a closed source application > having --enable-gpl in the configuration [16:40:58 CET] <cosmo1t> yeah [16:41:00 CET] <cosmo1t> oops. [16:41:33 CET] <JEEB> anyways, I'm trying to look into all this mess of cargo cult that is building FFmpeg for ARMv7 [16:41:48 CET] <JEEB> seems like everyone has their dozens of cflags they're pushing [16:42:04 CET] <JEEB> and no-one has actually put out any general benchmarks regarding any of that [16:42:16 CET] <JEEB> or if they've run FATE with such (which is the regression test suite) [16:42:17 CET] <cosmo1t> yeah it's not very concise. and yeah no benchmarks either. [16:43:06 CET] <JEEB> enable-thumb seems to be an official options so I enabled that, although I'm not fully sure if that gets enabled by default if your cpu is armv7 [16:43:11 CET] <JEEB> *option [16:43:11 CET] <gnome1> aah, compilation for ARM, plus features some ARMs don't really support (IIRC valgrind can't be compiled for some instruction sets), ... [16:43:35 CET] <JEEB> then you've got the usual march etc stuff [16:44:05 CET] <gnome1> was thumb for speed or space? I think one of the results was smaller binaries [16:44:18 CET] <cosmo1t> i didn't enable that. [16:44:39 CET] <JEEB> --enable-thumb compile for Thumb instruction set [16:44:43 CET] <JEEB> is all that the help says [16:44:56 CET] <gnome1> but I don't think you'll need a train of options for compiling (as far as you already have the cross-compiler ready) [16:45:06 CET] <JEEB> yeah [16:45:11 CET] <JEEB> it Just Worked in general [16:45:26 CET] <JEEB> just looking through all these random pastebins by random people building for iOS/Android [16:45:40 CET] <Mavrik> gnome1 it's both [16:45:45 CET] <Mavrik> ** sometimes [16:45:47 CET] <Mavrik> ;) [16:46:08 CET] <Mavrik> Thumb makes for significantly smaller binaries and it tends to be faster in some cases (and much slower in others). [16:46:21 CET] <JEEB> I'll be updating to NDK r11b and I guess I'll be trying out 3.8 clang while at it [16:47:48 CET] <JEEB> but yeah, I bet more perf could be gotten from learning NEON and making actual optimizations rather than squeeling around configure parameters ;) [16:47:50 CET] <cosmo1t> jeeb i used this: http://pastebin.ca/3411141 [16:48:19 CET] <JEEB> yeah, that's still sane. no random cargo cult rice :) [16:48:33 CET] <J_Darnley> Too bad only silly people use arm. If sensible people used it might get some of that attention. [16:48:35 CET] <JEEB> although --enable-static is still the default [16:48:53 CET] <JEEB> J_Darnley: I kind of like being able to have libmpv's opengl rendering on my android :D [16:48:53 CET] <cosmo1t> so you only need disable-shared [16:49:05 CET] <JEEB> cosmo1t: shared is disabled by default as well [16:49:17 CET] <JEEB> so you only have to poke those when you want to build only shared [16:49:21 CET] <cosmo1t> well damn i swear i read ./configure --help | less [16:49:22 CET] <JEEB> --disable-static --enable-shared [16:49:38 CET] <JEEB> but if you only need static, you just don't do anything with those two :) [16:49:48 CET] <cosmo1t> yeah i just pulled them from the .sh [16:50:17 CET] <cosmo1t> well thats scary, app compiled [16:50:22 CET] <cosmo1t> now to see if the pciture is purple or solid black [16:51:05 CET] <cosmo1t> damn it's purple. [16:51:15 CET] <cosmo1t> why isnt' it purple wheni use avpicture. sigh. [16:52:06 CET] <JEEB> well at least it seems like your pointers are kind of pointing at some of the video data :P [16:52:19 CET] <JEEB> now you just have to find out where you're doing things wrong [16:52:28 CET] <cosmo1t> story of my oife [16:52:31 CET] <cosmo1t> life [16:55:21 CET] <cosmo1t> one would think this wouldn't makit it go purple http://pastebin.ca/3411148 [16:59:48 CET] <cosmo1t> well thats a relief. the develop branch is purple as well. wtf. time to go back to the last release branch. /me bangs head [17:00:00 CET] <JEEB> :D [17:00:11 CET] <cosmo1t> thanks again. jeeb if u get anywhere with arm configure options i'd liek to see the data [17:00:28 CET] <cosmo1t> of course i dont' have another camera. [17:00:33 CET] <JEEB> probably not since I'm just doing it for the lulz [17:00:41 CET] <cosmo1t> k [17:00:47 CET] <cosmo1t> well thanks for the help [17:24:50 CET] <wooster> does RTSP ANNOUNCE not work anymore? https://github.com/revmischa/rtsp-server/issues/6 [17:25:03 CET] <wooster> apparently returns 405? [17:37:54 CET] <jones___> hi! my goal is to create a video live stream from bitmaps i copy from a hooked process. i compiled x264 and ffmpeg (for swscale) and started to implement. now during my research i found out ffmpeg practically has the feature i was starting to implement already. now i would like to know how i can feed the bmp frame data from my process to the ffmpeg process? i am running windows 10 x64. any advice is apreciated. [17:38:34 CET] <J_Darnley> If you have files on disk use the image2 demuxer [17:39:14 CET] <jones___> i dont have them on disk and i wondered if there would be a way to feed them directly from memory. I get 25 frames per second. [17:39:28 CET] <J_Darnley> Use the API [17:40:00 CET] <J_Darnley> Set up an AVFrame or AVPicture with the right properties [17:40:26 CET] <jones___> ok [17:40:35 CET] <J_Darnley> If that's not possible copy the pixels into one. [17:41:10 CET] <jones___> avframe and avpictures are streamable structures for ffmpeg i assume? [17:42:22 CET] <jones___> thanks for the hint. ill read up on avframe and avpicture and the api a little. any functions you recommend starting with? [17:42:29 CET] <J_Darnley> It is a strucure for passing around an uncompressed video frame. [17:43:02 CET] <J_Darnley> Where to start? One of the malloc functions? [17:43:07 CET] <J_Darnley> I don't know. [17:43:10 CET] <jones___> heh. [17:43:18 CET] <J_Darnley> You're one one who knows what he has already. [17:43:44 CET] <J_Darnley> Perhaps the encoding examples. [17:45:54 CET] <jones___> well currently i am transmitting raw 24bpp bmp data over tcp to a server. this works well as long as it is on my local machine. i only have 2 mbps upstream. so my approach was to use x264 to encode it to h264 and use swscale to convert the RGB to YUV. On my test programm i am currently at encoding the bitmap frames and i wanted to start to work on the transfer. while googling i found out ffmpeg already has a feature like that (frame [17:46:04 CET] <jones___> and i wouldnt want to implement what probably is better implemented in ffmpeg already [18:13:57 CET] <tyngdekraften> Is DSD encoding supported by ffmpeg? [18:17:51 CET] <BluesKaj> what would be the best audio string to insert in this command? card 0/hw:0 is the default in kubuntu system settings multimedia: ffmpeg -r 30 -f v4l2 -video_size 720x480 -i /dev/video0 output.avi ...I'm trying to copy some old vhs tapes with a capture card and eventually burn them to dvd [18:33:53 CET] <BluesKaj> BBL [18:55:30 CET] <fatpelt> afternoon all! i've got a command that will take 4 input udp streams and then stack them up into a 2x2 grid and display it. i'm seeing that if i set all 4 incoming streams to the same dup input stream (by just specifying 4 copies of the one -i option) i see about 5 second delay between each image of 2x2 grid. is there any way to get those synchronized? [19:00:55 CET] <fatpelt> also, i'm seeing this a *ton* in my output. i've tried to figure out what it means, but haven't found anything really conclusive: "dts < pcr, TS is invalid" [19:10:54 CET] <andrey_utkin> fatpelt, apparently ffmpeg command line util opens inputs sequentially and analyzes each stream for 5 sec. You'd need custom application using libavformat/libavcodec API which implements multithreading as you wish [19:11:23 CET] <andrey_utkin> then you'd need to solve many questions, like what to do if one channel doesn't have data for presentation in time [19:13:04 CET] <BluesKaj> andrey_utkin, any advice on my post above ? [19:13:19 CET] <andrey_utkin> lemme look [19:14:04 CET] <andrey_utkin> BluesKaj, don't quite get your question [19:15:42 CET] <BluesKaj> andrey_utkin, I need to add audio capture to this string , what and where should it be placed "ffmpeg -r 30 -f v4l2 -video_size 720x480 -i /dev/video0 output.avi" [19:16:40 CET] <BluesKaj> the video is captured just fine , but I also need the audio portion as well [19:17:38 CET] <andrey_utkin> between "video0" and "output" [19:17:50 CET] <andrey_utkin> insert "-f alsa -i default" [19:17:56 CET] <andrey_utkin> or "-f pulse -i default" [19:18:22 CET] <andrey_utkin> maybe you'd need also to add "-map 0 -map 1" if ffmpeg doesn't put both streams to output file [19:19:18 CET] <BluesKaj> ok and where in the command should I insert them [19:19:38 CET] <furq> 18:17:38 ( andrey_utkin) between "video0" and "output" [19:19:42 CET] <BluesKaj> does the sequence matter? [19:19:48 CET] <llogan> you should probably use -framerate instead of -r with v4l2. i can't remember if it makes a difference or not, but that's what is listed in the AVoptinos for that input device [19:19:59 CET] <BluesKaj> ok thanks . andrey_utkin [19:21:19 CET] <BluesKaj> llogan, ok thanks, will do [19:23:02 CET] <fatpelt> andrey_utkin: kk. that's a bummer but good information to have [19:24:52 CET] <fatpelt> any idea on the dts < pcr error? [19:30:00 CET] <andrey_utkin> fatpelt, let's start with looking at full log together with command line [19:44:11 CET] <whoever> looks as though my only option is divx for burned movie to play in a dvd player http://www.manualslib.com/manual/118178/Panasonic-Dvd-Rv22.html?page=5#manu… [19:44:17 CET] <whoever> or did i miss somethinag [19:44:20 CET] <fatpelt> andrey_utkin: let me get a pastebin up [19:49:52 CET] <fatpelt> andrey_utkin: http://pastebin.com/DG3TMAP1 [19:58:41 CET] <andrey_utkin> well, "TS is invalid" is a problem at mpegts "file writer" (at last not reader). More than that, I can't help now. I don't even remember what PCR is and why this could happen [19:58:54 CET] <andrey_utkin> in any uncrear situation, decrease complexity [19:59:41 CET] <andrey_utkin> try "ffmpeg -f lavfi -i testsrc " and all your output flags (starting with "-c:v") [20:00:21 CET] <andrey_utkin> then if thing still happens, drop some output flags and see how it works [20:00:32 CET] <andrey_utkin> i don't know much about these flags and why you use them [20:00:46 CET] <fatpelt> honestly i don't either. most of them came from a coworker [20:00:55 CET] <andrey_utkin> start with nothing [20:01:18 CET] <andrey_utkin> ffmpeg -f lavfi -i testsrc -c libx264 out.mpegts [20:01:27 CET] <andrey_utkin> then [20:02:00 CET] <andrey_utkin> ffmpeg (inputs as you have them) -c libx264 out.mpegts [20:08:43 CET] <fatpelt> ok. i see where you're going. if i get the errors with that then there are issues with input. if not, add back options till i see it [20:15:23 CET] <jones___> is it possible that the sdp created my ffmpeg is not complete? playing it with ffplay gives me a "test.sdp: Invalid data found when processing input". although if i connect with ffplay rtp://127.0.0.1:1234 it works [20:18:33 CET] <andrey_utkin> fatpelt: inputs seem ok [20:18:53 CET] <andrey_utkin> Ah except for desync [20:20:52 CET] <fatpelt> desync? [20:21:12 CET] <fatpelt> i'm getting a different error now. Past duration 0.999992 too large [20:21:21 CET] <fatpelt> the original error is gone [20:21:51 CET] <llogan> that's a warning, not an error [20:23:34 CET] <fatpelt> may have been audio related. i added -an and it seems to have subsided [20:24:21 CET] <fatpelt> adn that's a significantly simpler command. my output looks right (with the delay,, but right) [20:28:33 CET] <fatpelt> so yesterday, someone pointed me at *stack instead of overlay. i don't suspect, but would there be any problem with overlaying on top of the stack [20:28:49 CET] <fatpelt> (they indicated that *stack was more efficient than overlay) [20:29:02 CET] <BluesKaj> dts is digital theater systems audio , similar to dolby digital but less lossy [20:29:20 CET] <fatpelt> and how does that relate to pcr? [20:29:29 CET] <BluesKaj> dunno [20:29:37 CET] <BluesKaj> :-) [20:29:53 CET] <BluesKaj> my 2 cents [20:33:57 CET] <jafa> MPEG2 data going through video decode where only the key frame has a PTS value... the video decode isn't assigning a PTS value so on the output I need to put in a PTS value based on the framerate. What is the best way to get the most up to date framerate (ie the framerate for the current group-of-pictures)? [20:35:42 CET] <sagax> hi all! [20:35:52 CET] <jones___> the command "ffmpeg -f rawvideo -pixel_format bgr24 -video_size 800x600 -framerate 25 -i tcp://localhost:8888 -c:v libx264 -pix_fmt yuv420p -f rtp -vf vflip -an rtp://localhost:81?listen" gives me the following sdp: [20:35:53 CET] <jones___> v=0 o=- 0 0 IN IP4 127.0.0.1 s=No Name c=IN IP4 127.0.0.1 t=0 0 a=tool:libavformat 57.29.100 m=video 81 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1 [20:36:07 CET] <jones___> but playing it with ffplay doesnt work. Protocol not on whitelist 'file,crypto'! test.sdp: Invalid data found when processing input [20:36:09 CET] <jones___> any hints? [20:36:46 CET] <jones___> same thing with a audio only rtp stream. but connecting to it with ffplay rtp://127.0.0.1:81 works [20:37:03 CET] <sagax> how do i know what Format working with AudioCodec? [20:37:17 CET] <sagax> for ffserver [20:37:18 CET] <andrey_utkin> BluesKaj, dts is decoding timestamp [20:38:56 CET] <jafa> ie how do I get the frmaerate associated with the group of pictures the frame is a member of? (rather than the initial detected framerate) [20:40:25 CET] <BluesKaj> andrey_utkin, too many acronyms ! :-) [20:49:55 CET] <sagax> what different format, codec, encode, decode [20:50:38 CET] <J_Darnley> What? I know all those words yet I don't understand. [20:50:45 CET] <sagax> it's crazy, i writing configuration for ffserver and not understand what it all [20:50:58 CET] <sagax> codec, format, decode [20:51:00 CET] <sagax> example [20:51:15 CET] <sagax> format MP3 and codec libmp3lame - working [20:51:39 CET] <sagax> but if i wish other format? how do i change codec or encoder or decoder? [20:52:05 CET] <llogan> sagax: i don't think anyone uses ffserver here. it is basically unmaintained. you're probably on your own. [20:52:20 CET] <J_Darnley> I believe you do that by changing what's written in the config file. [20:52:28 CET] <J_Darnley> also llogan has a good point. [20:53:36 CET] <llogan> only thing i can suggest is 3 links: http://ffmpeg.org/ffserver.html http://ffmpeg.org/sample.html https://trac.ffmpeg.org/wiki/ffserver [20:54:34 CET] <llogan> it is unfortunate that 99% of ffserver questions go unanswered. maybe i should submit a patch to remove it. [20:54:38 CET] <sagax> ok, i trying read more, [20:54:50 CET] <sagax> thanks [20:56:16 CET] <llogan> or a user servey to see who uses it (successfully), what they use it for, and how they use it would be interesting [20:56:30 CET] <J_Darnley> The users would respond. [20:56:34 CET] <J_Darnley> *would not [20:56:35 CET] <furq> if nothing else it should probably be removed from the wiki [20:56:42 CET] <J_Darnley> ffserver users are silent [20:56:50 CET] <J_Darnley> streaming users are silent [20:57:07 CET] <furq> i wish they were silent [20:57:07 CET] <J_Darnley> they all need to protect their secret sauce less the competition take it. [20:57:17 CET] <furq> then i wouldn't have to listen to their worst possible human voices while they play old video games [20:57:20 CET] <J_Darnley> The wannabe users are not silent [20:57:35 CET] <J_Darnley> oh [20:58:33 CET] <gnome1> or maybe they're happy and they don't ever need support of any kind, so they don't even know people are talking about ffserver [20:58:47 CET] <furq> i find it unlikely that ffserver users never need support [20:59:03 CET] <furq> but who knows [20:59:12 CET] <furq> they are a mysterious bunch [21:56:08 CET] <martin55> hi. I want to make a question and put code in http://ffmpeg.pastebin.com/ but i get an error [21:56:25 CET] <martin55> can u help me? [23:28:46 CET] <llogan> martin5: it you're also martin55 then you should provide the actual url [00:00:00 CET] --- Sat Mar 26 2016

1 0

ffmpeg-devel.log.20160324
by burek 25 Mar '16

25 Mar '16

[00:00:31 CET] <atomnuker> not sure if they know only one can be accepted, I assumed they do [00:03:42 CET] <cehoyos> I believe it is appreciated if you make sure they know. (At least this is what I think makes sense.) [00:05:27 CET] <atomnuker> just sent them a message [00:06:58 CET] <cehoyos> Thank you! [00:08:07 CET] <cehoyos> If you believe both could be suitable students, we could try to find alternative projects for one. [00:55:49 CET] <kierank> qiubit: I like the way you exposed the exact issue that my GSoC project wants to solve [00:56:14 CET] <kierank> We have no way of tracking fuzz regressions [01:02:20 CET] <qiubit> kierank: thanks, and BTW I didn't do anything fancy. It turns out that if you give afl enough time, it will find something interesting. And who knows how much more can it find, if fuzzing will be done somewhat more intelligently... [01:05:28 CET] <wm4> lol [01:05:37 CET] <wm4> why do these people want to use ffmpeg at all [01:12:13 CET] <TD-Linux> wm4, because it's apple quality!! [01:12:22 CET] <TD-Linux> (see imgix) [01:28:01 CET] <durandal_1707> What's wrong with this afero guy? [01:31:24 CET] <llogan> What's wrong with these top-posting guys? [01:34:43 CET] <TD-Linux> what's wrong with these guys that use IRC as mailing list backchannel [01:37:02 CET] <iive> something's wrong with the world today [01:37:07 CET] <iive> i don't know what it is... [02:30:11 CET] <Shiz> x/w 48 [04:07:07 CET] <cone-281> ffmpeg 03Petru Rares Sincraian 07master:68e5976543e9: Refactor libavutil/parseutils.c [04:38:22 CET] <rcombs> how _about_ on iOS [04:41:56 CET] <rcombs> Shiz: error: failed to read memory from 0x30. [06:56:10 CET] <rcombs> https://trac.ffmpeg.org/ticket/5366 <-- is there actually any compelling reason to use AU filters over the built-in lavfi ones [06:56:18 CET] <rcombs> like, are they actually higher-quality or faster or something [06:56:50 CET] <rcombs> (the codecs actually present a significant perf difference) [08:27:54 CET] <omerjerk> I'm working on my proposal for ALS encoder. [08:28:13 CET] <omerjerk> I want to know how does the review process take place ? [08:28:38 CET] <omerjerk> review process for adding the encoder in the official codebase. [08:44:26 CET] <JEEB> omerjerk: you can ask very initial comments by linking your repo here, and when your patch is in a state in which you think you can post it for actual review, you post it on the ML [09:18:46 CET] <linuxuser9000> Hi! I'm trying to properly configure my git clone of ffmpeg such that I can use most of the disabled-by-default encoders, is there a better way to do this than to just give the --encoder* option for every encoder? [09:28:44 CET] <ethe> linuxuser9000: you probably want #ffmpeg [10:53:18 CET] <JEEB> btw, did the work on the subtitles without end timestamps also affect picture-based subtitles? [11:23:18 CET] <ubitux> JEEB: what work? [11:23:26 CET] <ubitux> oh you mean the realtime in the cc decoder?" [11:24:00 CET] <JEEB> yeah, that you actually get the subtitle packets even if no endtime was gotten [11:24:41 CET] <ubitux> it's just a private option in that decoder, i don't remember what value was decided to be used [11:24:58 CET] <ubitux> i don't remember if some other do that [12:44:03 CET] <cone-493> ffmpeg 03Michael Niedermayer 07master:50d017a28171: avformat/mpegtsenc: Keep track of the program for each service [12:44:04 CET] <cone-493> ffmpeg 03Michael Niedermayer 07master:26811fd9468d: avformat/mpegtsenc: Fix used service [12:46:08 CET] <Razva> guys, any RHEL/CentOS repo with ffmpeg 3.0 available? [12:47:01 CET] <JEEB> not a -devel related thing, please don't spam channels [12:56:43 CET] <ubitux> http://sprunge.us/BMND i wonder if a prefetch could help [13:48:24 CET] <nevcairiel> rcombs: AT patches have broken building on all iphone targets (ie. arm darwin) [13:51:08 CET] <wm4> does iphone not have this API [14:05:12 CET] <cone-493> ffmpeg 03Rostislav Pehlivanov 07master:72e13600079e: vc2enc: optimize and simplify quantization [14:07:40 CET] <wm4> does fate not stop on the first error? [14:15:30 CET] <ubitux> running with -jX or -k? [14:16:27 CET] <wm4> either [14:38:45 CET] <nevcairiel> sure stops for me [14:38:48 CET] <nevcairiel> just dont run with -k or -i [14:39:47 CET] <wm4> oh it's -k [14:39:55 CET] <wm4> so much for mindlessly copying command line options [14:59:23 CET] <ubitux> re: codecpar; are we going to introduce the codec time_base in it for subtitles? [14:59:39 CET] <ubitux> that's one of the last "meta" issue, isn't it? [15:12:58 CET] <Daemon404> ubitux, i would really like to avoid that since it's a public struct [15:13:05 CET] <Daemon404> :/ [15:15:40 CET] <wm4> I don't see where the problem is, I had to set an explicit timebase on subtitle converters before [15:15:52 CET] <wm4> while no other decoders required this [15:18:03 CET] <wm4> does anyone know how ffmpeg.c drains filter frames on midstream format changes? [15:20:16 CET] <Daemon404> wm4, its not a real fix though [15:20:31 CET] <Daemon404> it's a shitty hack to passtimebase in a place its not supposed to be [15:20:41 CET] <Daemon404> it's not a "codec parameter" [15:20:52 CET] <Daemon404> (the real fix is avframe subtitles IMO) [15:21:05 CET] <wm4> there are other things that use timebase and require it to have a specific value in decoders [15:21:18 CET] <Daemon404> such as? [15:22:01 CET] <Daemon404> (it's worth noting nobody has been able to provide a legitimate explanation of what a "codec timebase" is, except that it's convenient for passing info that may or may not be an actual timebase) [15:22:13 CET] <Daemon404> (most of the shit i see in lavf uses it for e.g. bitstream *framerate*) [15:22:28 CET] <wm4> ah maybe I'm wrong [15:23:15 CET] <wm4> seems like the general requirement is that they're linear integers and that AV_NOPTS_VALUE is special [15:23:58 CET] <wm4> there are still some weird reference to time_base in utils.c though [15:24:19 CET] <Daemon404> weird is the key word here [15:24:22 CET] <Daemon404> i doubt it is correct [15:24:36 CET] <Daemon404> e.g. the use in the framr duration computations is insane [15:24:50 CET] <Daemon404> because codec timebase is used interchangeably with framerate in lavf [15:25:01 CET] <Daemon404> like 90% of the fate failures were that so far [15:25:15 CET] <wm4> I only see this in the encoding path: [15:25:16 CET] <wm4> avpkt->duration = ff_samples_to_time_base(avctx, [15:25:16 CET] <wm4> frame->nb_samples); [15:25:49 CET] <Daemon404> iaybe its not utils.c where teh func im thinking of is [15:25:55 CET] <Daemon404> but same idea [15:26:18 CET] <Daemon404> i havent found a use yet where it's seemed 'correct', in any case [15:26:52 CET] <durandal_1707> just change fate [15:27:17 CET] <Daemon404> durandal_1707, no the correct fix was mostly to access the correct, framerate, fields [15:27:33 CET] <Daemon404> fate had a good refence for most. [15:30:09 CET] <nevcairiel> changing fate is usually not the right answer, unless you can clearly proof that the old value was wrong and the new one is better (which happens) [15:30:20 CET] <nevcairiel> ie. every single fate change has to be properly explained, or its a bug [15:30:24 CET] <Daemon404> yes [15:41:07 CET] <cone-493> ffmpeg 03Ico Doornekamp 07master:e3e6a2cff4af: avformat/rtpdec_jpeg: fix low contrast image on low quality setting [15:50:33 CET] <Daemon404> huh.. daala uses the term 'golden frame' too [15:50:39 CET] <JEEB> funky [15:52:59 CET] <wm4> doesn't this AGPL guy make a convincing argument [15:54:25 CET] <Daemon404> im not even reading that thread [15:54:34 CET] <Daemon404> people who argue for AGPL arent even worth my time to read [15:55:06 CET] <wm4> I mean his argument about openjpeg vs. internal decoder [15:55:19 CET] <JEEB> did he actually find copied code? [15:55:22 CET] <nevcairiel> why do we attract the obnoxious nutjobs lately [15:55:44 CET] <Daemon404> ... lately? [15:55:53 CET] <nevcairiel> dunno, seems to be getting worse [15:55:54 CET] <Daemon404> this project is founded on that [15:55:58 CET] <Daemon404> us included [15:56:04 CET] <Daemon404> (you have to be nitjobs to enjoy working with multimedia) [15:56:08 CET] <wm4> seems accurate [15:56:09 CET] <Daemon404> nut* [16:01:41 CET] <rcombs> nevcairiel: huh, I guess iOS just doesn't provide VBR [16:05:09 CET] <durandal_1707> so no tep2, still? [16:05:57 CET] <Daemon404> you are fere to help [16:05:59 CET] <Daemon404> free* [16:06:10 CET] <Daemon404> i dont think much will get done on easter eekend [16:06:13 CET] <Daemon404> since, well, it is easter. [16:07:46 CET] <rcombs> should AVOptions not available on a given platform be removed or "stubbed"? [16:13:02 CET] <Daemon404> rcombs, whats this related to? [16:13:16 CET] <rcombs> Daemon404: http://fate.ffmpeg.org/log.cgi?time=20160324030457&log=compile&slot=arm64-d… [16:13:33 CET] <Daemon404> right. well API-wise, i dont think it matters if there is a stub or not. [16:13:40 CET] <Daemon404> it's a key/value dict, so. [16:14:23 CET] <rcombs> some things error if you pass unknown AVOpts, but I guess if you're passing an opt that doesn't exist you'll probably want to know about it [16:14:34 CET] <Daemon404> what errors? [16:14:44 CET] <Daemon404> maybe im not remembering correct, but i thought they were ignored [16:15:44 CET] <Daemon404> if they errored on unknown option, then we'd have to bump API major a lot more [16:17:01 CET] <rcombs> oh, I thought ffmpeg errored but I guess it just logs [16:17:12 CET] <rcombs> (ffmpeg_opt) [16:17:58 CET] <Daemon404> who would be running ffmpeg cli on ios [16:19:00 CET] <wm4> Daemon404: multimedia professionals? [16:19:29 CET] <Daemon404> technically, you legally cant [16:19:43 CET] <Daemon404> e.g. certain fate clients which run ios use rooted phones [16:19:52 CET] <Daemon404> with samba and exec scripts [16:19:55 CET] <Daemon404> it's rather very complicated [16:37:11 CET] <ubitux> Daemon404: i don't think it's wise to wait for the subtitles frame api for the merge [16:37:22 CET] <ubitux> we have way too much delay now on libav commits already [16:37:38 CET] <ubitux> we can just do the usual "fields below are private" in codecpar [16:39:40 CET] <Daemon404> ubitux, ok [16:40:28 CET] <ubitux> also, add a big fat warning saying that this time base should be exclusively used for decoded subtitles because of current API limitations [16:41:18 CET] <rcombs> are there any subtitle codecs where 1/1000 isn't suitable? [16:41:45 CET] <rcombs> (sure ASS is 1/100 but it doesn't hurt anything to store it as 1/1000 internally) [16:44:08 CET] <wm4> hm, would that be compatible? [17:08:59 CET] <Daemon404> ... sigh [17:09:01 CET] <Daemon404> NIH zlib [17:09:09 CET] <Daemon404> this may be one of the dumbest patchsets ever [17:12:27 CET] <nevcairiel> its also super ugly [17:12:36 CET] <Daemon404> yes [17:13:24 CET] <wm4> I can get behind wanting to NIH zlib, but not this ifdef soup [17:13:51 CET] <Daemon404> i cant get behind wanting to NIH zlib [17:14:09 CET] <Daemon404> Windows on MSVC is more or less the only place you do not have an easily accessible zlib [17:14:59 CET] <sfan5> mingw has zlib out-of-the-box? [17:15:01 CET] <sfan5> did i miss something [17:15:06 CET] <sfan5> i always had to build zlib seperately [17:15:33 CET] <nevcairiel> then you used a bad mingw package, the most basic libraries should be coming with it [17:15:33 CET] <Daemon404> sfan5, most chains do, or: ./configure && make [17:15:48 CET] <Daemon404> and its still retarded to NIH it for that [17:15:48 CET] <sfan5> "bad mingw package" [17:15:52 CET] <sfan5> i built it myself [17:15:54 CET] <sfan5> it was a PITA [17:16:00 CET] <Daemon404> wut? [17:16:10 CET] <nevcairiel> its just like manually building gcc on linux [17:16:12 CET] <Daemon404> iirc its like one command [17:16:14 CET] <wm4> use msys, do pacman -S zlib-whatever? [17:16:20 CET] <Daemon404> make -fwin32/Makefile.mingw [17:16:21 CET] <mabreedlove> wanted to say these new builds I've done have been doing a much better job at fully taxing my cpu, so much appreciated [17:16:22 CET] <wm4> (the only hard part is finding the package name) [17:16:23 CET] <nevcairiel> anyone who is sane would just get a package that suits their needs :D [17:16:24 CET] <Daemon404> optionally with PREFIX [17:16:32 CET] <sfan5> yeah no [17:16:40 CET] <Daemon404> then youre doing it massively wrong [17:16:41 CET] <sfan5> i did linux from scratch but with mingw [17:16:44 CET] <wm4> anyway that's no real argument [17:16:57 CET] <wm4> zlib is one of the most ubiquitous libs ever [17:17:02 CET] <nevcairiel> there are plenty of pre-packaged mingw environments that include what most projects need [17:17:08 CET] <nevcairiel> if you build your own, you are of course also responsible [17:17:11 CET] <mabreedlove> sfan5: theres a project that builds ffmpeg and its libs native on windows if you want, just asks which libs you'd like [17:17:22 CET] <nevcairiel> on that ground you might argue that we should include a compiler =p [17:17:27 CET] <sfan5> nah i don't need ffmpeg built for windows [17:17:47 CET] <sfan5> nevcairiel: why stop there? include a kernel! [17:17:50 CET] <sfan5> :P [17:18:13 CET] <Daemon404> libav* including libc or a kernel has been a running joke [17:18:14 CET] <mabreedlove> micro or monolithic? :p [17:18:16 CET] <Daemon404> but obvious reasons [17:18:28 CET] <Daemon404> mabreedlove, in libav* fashion it will be monolithic, but claim to be micro [17:18:35 CET] <mabreedlove> haha [17:19:02 CET] <mabreedlove> you guys amuse me everytime I come in here [17:20:24 CET] <nevcairiel> Daemon404: should just package miniz with ffmpeg to make reimar and his broken buildenv happy [17:20:30 CET] <nevcairiel> its api compatible to libz and just a single c file! [17:20:46 CET] <wm4> dump it into compat/ ? [17:20:51 CET] <wm4> or what [17:21:06 CET] <mabreedlove> have you guys tried slim-lto builds using --enable-lto on your buildchain? [17:21:15 CET] <mabreedlove> our of curiosity [17:21:18 CET] <nevcairiel> lto is rather broken [17:21:39 CET] <mabreedlove> it works. just not like....without spending a ton of time screwing with it [17:21:48 CET] <RiCON> last time i tried if failed building, didn't try too much [17:21:58 CET] <nevcairiel> it conflicts with some of our inline assembly [17:22:09 CET] <mabreedlove> thats what Im referring to ;) [17:22:27 CET] <wm4> nevcairiel: interesting [17:22:39 CET] <mabreedlove> I built it awhile back for my own usage but required disabling avx/avx2, etc which I didnt have native support for [17:23:00 CET] <mabreedlove> this was before gcc 5+ had better thin LTO support though [17:23:09 CET] <RiCON> unless it considerably improved performance, i'm not very interested in marginal space savings [17:23:32 CET] <mabreedlove> saves a fairly significant amount of space on a static build [17:24:21 CET] <drv> everything old is new again [17:24:23 CET] <mabreedlove> I haven't done proper benchmarks so I can't disagree with you in terms of performance though [17:24:23 CET] <drv> http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/46754 [17:25:33 CET] <mabreedlove> I have a feeling thin-lto build may fix the assembly errors, however [17:41:37 CET] <Daemon404> mabreedlove, lto generally doesnt work for ffmpeg [17:41:47 CET] <Daemon404> due to us relying on DCE to disable arch-specific stuff [17:41:58 CET] <Daemon404> so LTOing on x86 will complain about missing ARM symbols [17:42:04 CET] <Daemon404> at least, last time i tried. [17:42:19 CET] <nevcairiel> why would lto not manage to figure this out [17:42:30 CET] <nevcairiel> seems like a rather important optimization if you actually want to save binary space [17:42:48 CET] <Daemon404> bevause it is broken [17:42:53 CET] <nevcairiel> all i ever got from it was fail from inline asm [17:42:54 CET] <mabreedlove> usually those problems were due to it using fat-lto-objects with references that couldn't be resolved when they were recompiled [17:43:00 CET] <Daemon404> ive had this happen with LTO on 3 different compiles, nevcairiel [17:43:07 CET] <Daemon404> compilers* [17:43:50 CET] <mabreedlove> I know what you're referring to, though. I disabled lto on my last build since I couldn't hassle with it at the time. I'm going to try it with thin LTO objects this time and will let you guys know if you like [17:44:08 CET] <mabreedlove> but yes, it is a complete pain in general to get working properly [17:47:44 CET] <mabreedlove> need to go through the new configure script and figure out what all of this stuff is useless for me [17:51:30 CET] <mabreedlove> SDL is just for ffplay mainly, right? [17:51:48 CET] <wm4> yes, but also libavdevice [17:54:33 CET] <mabreedlove> not that many options on windows for it anyway. can probably dump it without noticing [17:54:46 CET] <mabreedlove> thanks [19:09:00 CET] <Angus__> Hey guys [19:09:25 CET] <Angus__> quick question with getting some stuff setup, [19:09:37 CET] <Angus__> this list here: [19:09:38 CET] <Angus__> https://ffmpeg.org/doxygen/2.8/dir_c323f0049949ddd42f5d8cc7327305d9.html [19:09:45 CET] <Angus__> and this list: [19:09:45 CET] <Angus__> http://packages.ubuntu.com/xenial/amd64/libavformat-dev/filelist [19:09:52 CET] <Angus__> are not quite the same thing [19:10:10 CET] <Angus__> what am I doing wrong here [19:10:21 CET] <Daemon404> the former is a list of all internal files. [19:10:24 CET] <JEEB> ^this [19:10:31 CET] <JEEB> those get built into the .so and .a [19:10:36 CET] <Daemon404> https://ffmpeg.org/doxygen/2.8/ [19:10:39 CET] <JEEB> and then you have the public headers [19:10:39 CET] <Daemon404> start exploring here. [19:11:46 CET] <Angus__> ah, alright, I should be cloning the source if I need the files for development? [19:13:22 CET] <JEEB> in-FFmpeg development, yes [19:13:33 CET] <JEEB> using FFmpeg for your own thingy, you can use those public things [19:14:20 CET] <Angus__> I'm trying to port a citrix plugin from 0.7/0.8 to 2.8 [19:17:04 CET] <JEEB> then you are something using the public APIs [19:19:10 CET] <Angus__> has url.h in libavformat been moved out of the public APIs between 0.8 -> 2.8? [19:19:19 CET] <Daemon404> url.h is not publci api [19:19:23 CET] <Daemon404> it was wrong to use it in teh first place [19:20:40 CET] <Angus__> citrix seems to have written their own "FFDecode.h/cpp" files :( [19:20:54 CET] <wm4> so, what is this citrix thing? [19:21:23 CET] <Angus__> https://www.citrix.com/downloads/citrix-receiver/receiver-for-linux-sdks/li… [19:22:14 CET] <wm4> another instance of <corporate>-sdk==bad time [19:23:03 CET] <Daemon404> Angus__, you can probably replace it using avio_* [19:23:06 CET] <Daemon404> i havent looked though [19:23:12 CET] <Daemon404> also, what wm4 said [19:24:05 CET] <Angus__> heh [19:24:42 CET] <Angus__> this is the wrapper they made: typedef int ( _PAPI PFN_av_register_protocol )( URLProtocol *protocol ); [19:24:54 CET] <Daemon404> that's... odd [19:25:04 CET] <JEEB> http://pastebin.com/7UaeED2Q [19:25:13 CET] <JEEB> their AVC decoding thingamajig through avcodec et al [19:25:28 CET] <Angus__> you found it. [19:25:29 CET] <Daemon404> er what [19:25:32 CET] <wm4> lol [19:25:34 CET] <Daemon404> why are they registering a protocol [19:25:36 CET] <Daemon404> for a bufr [19:25:39 CET] <Daemon404> you can use avio callbacks [19:25:41 CET] <Daemon404> to do exactly that [19:25:59 CET] <wm4> I want to know why they dlopen everything [19:26:46 CET] <Daemon404> also there is next to no error checking [19:26:57 CET] <Daemon404> but eh. corporate code. [19:27:30 CET] <JEEB> also they did their NAL unit parser or whatever [19:27:53 CET] <wm4> I don't get it [19:27:58 CET] <wm4> why are they using libavformat at all [19:28:07 CET] <JEEB> or well, "parser" since it really doesn't do jack shit :D [19:28:09 CET] <durandal_1707> what's status of our j2k decoder? [19:28:35 CET] <nevcairiel> it kind of works on dcinema files, probably not all that well on any others [19:29:02 CET] <Daemon404> it really doesnt fully. [19:29:07 CET] <Daemon404> iirc it cant do full 4k [19:29:13 CET] <Angus__> I have not a clue, I just started working on this thing a yesterday [19:29:36 CET] <JEEB> anyways, if they have buffers you can just do custom AVIO [19:29:51 CET] <JEEB> heck, I was able to make a wrapper for Windows IStreams and I'm dumb :P [19:30:26 CET] <JEEB> https://github.com/jeeb/matroska_thumbnails/blob/master/src/istream_wrapper… [19:31:56 CET] <JEEB> https://github.com/jeeb/matroska_thumbnails/blob/master/src/matroska_thumbn… and then I just used the thing :D [19:34:08 CET] <Angus__> right now, I'm just trying to get this to compile, see what it does [19:35:36 CET] <JEEB> well, good luck :P it doesn't do too much of anything, but it looks like a classic corporate dump [19:35:50 CET] <Angus__> is there any reason why I shouldn't copy the devel headers to /usr/include/etcetc ? [19:36:00 CET] <Daemon404> because its not piblic api [19:36:13 CET] <Daemon404> you are using stuff that is not exposed nor intended to be used [19:36:20 CET] <Daemon404> if it doesnt work, we wont support it [19:36:40 CET] <JEEB> nothing there requires any of the private APIs as far as I can tell :P [19:37:03 CET] <Daemon404> nope [19:37:13 CET] <Daemon404> they use private apis to implement something the public api has had for many years [19:37:21 CET] <Daemon404> it's clear no docs were read [19:38:10 CET] <Angus__> lol [19:44:30 CET] <cone-289> ffmpeg 03Petru Rares Sincraian 07master:f707042c931c: Added more tests to libavutil/parseutils.c [19:44:30 CET] <cone-289> ffmpeg 03Michael Niedermayer 07master:54f43984e1e4: avutil/parseutils: mark args as static const [20:20:06 CET] <jkqxz> Why would RTP JPEG clip away quality 100? (Seeing it in that patch which went past a few hours ago.) RFC 2435 allows 1-99 and says 100-127 are "reserved", but 100 is well-defined so deliberately decoding it incorrectly seems kindof pointless, unless there is some other constraint I've missed. [20:23:58 CET] <Angus__> +JEEB, you posted this a bit ago: [20:23:59 CET] <Angus__> http://pastebin.com/7UaeED2Q [20:24:12 CET] <Angus__> do you have any idea what the BufferContext is in there? [20:24:40 CET] <Daemon404> its their own struct. [20:24:45 CET] <Daemon404> it's user data. [20:25:45 CET] <Angus__> hmm [20:33:22 CET] <Angus__> https://www.citrix.com/downloads/citrix-receiver/linux/receiver-for-linux-l… [20:33:34 CET] <Angus__> oh ho ho, "FFmpeg Source Package" [20:35:09 CET] <Angus__> "The FFmpeg library has been modified by Citrix Systems, Inc. The difference file between the initial FFmpeg source and the Citrix modified FFmpeg source is 'Citrix-FFmpeg-DIFF.txt'." [20:36:17 CET] <JEEB> because of course [20:37:22 CET] <Daemon404> where is said txt file? [20:39:01 CET] <JEEB> https://www.citrix.com/downloads/citrix-receiver/linux/receiver-for-linux-l… [20:39:05 CET] <Angus__> ^ [20:39:22 CET] <JEEB> blergh, not a straight url [20:39:23 CET] <Angus__> "Additional components" [20:39:27 CET] <JEEB> yeah [20:39:40 CET] <Daemon404> this is technically noncompliat [20:39:47 CET] <Daemon404> it violates the lgpl to force me to agree with a eula [20:40:03 CET] <JEEB> yeah, seems to be a generic shithole they have :P [20:40:24 CET] <Daemon404> dling at 32 kb/s [20:40:26 CET] <Daemon404> nice [20:40:59 CET] <JEEB> anyways, they seem to be using lavc purely for AVC/JPEG (?) sw decode if their crappy hw decoder doesn't work [20:41:03 CET] <JEEB> so not much use I guess? [20:41:16 CET] <JEEB> unless they implemented their hwdec in their fork of FFmpeg, which I doubt [20:41:44 CET] <BtbN> the changes are minimal [20:41:47 CET] <BtbN> and look broken [20:41:54 CET] <Daemon404> BtbN, so like the rest of the code? [20:42:00 CET] <BtbN> no idea what the hell they are doing there, except for the ripping out all crypto [20:42:17 CET] <Daemon404> they cant legally export strong crypto probably [20:42:19 CET] <Daemon404> because USA USA USA [20:42:29 CET] <BtbN> Yeah, but there are other changes, which make no sense [20:42:38 CET] <llogan> old ass source [20:42:41 CET] <Daemon404> well their api-using code makes no sense either [20:42:41 CET] <Daemon404> so. [20:43:47 CET] <Angus__> you guys ever had to deal with legal team code reviews? [20:43:54 CET] <Angus__> horrible. horrible stuff. [20:44:12 CET] <JEEB> yeah, usually at the point where a dependency is added into some stack [20:44:24 CET] <JEEB> what license the thing is under, etc [20:45:19 CET] <Daemon404> unless you are a multimedia bigcorp, then you just violate all the licneses [20:45:55 CET] <dinux5> Do the developers here use any IDE ? Is it recommended or no? [20:46:07 CET] <Daemon404> you can use whatever editor or ide you want [20:46:15 CET] <Daemon404> not sure why you wouldnt [20:46:31 CET] <JEEB> if you can set up an IDE with the build system and you feel like an IDE brings anything to you then feel free [20:46:37 CET] <JEEB> everyone likes their own thing here methinks [20:47:44 CET] <dinux5> Which one specifically ? The eclipse one for C is not much developed. [20:48:27 CET] <Daemon404> most of us probably use vim or emacs. [20:48:36 CET] <JEEB> you're going to get N amounts of answers here which is why I'm just telling you to use whatever you like/want to use [20:48:46 CET] <JEEB> if you can set the FFmpeg build system into your IDE, feel free [20:48:59 CET] <JEEB> I did that once with Qt Creator and L-SMASH, but never for FFmpeg [20:49:43 CET] <Angus__> I'm using eclipse here just for the editor [20:50:17 CET] <dinux5> Alright. Thanks :) [21:06:01 CET] <Angus__> Do you guys know if this still exists? [21:06:01 CET] <Angus__> https://www.ffmpeg.org/doxygen/0.6/libavformat_2raw_8c-source.html [21:07:23 CET] <Daemon404> uh... context? [21:07:33 CET] <Daemon404> that file is internal to avformat [21:08:00 CET] <JEEB> and as has been noted N times here already, none of what that AVC thing did required internal APIs [21:08:38 CET] <Angus__> I'm trying to patch the citrix changes to a newer build of FFMpeg, [21:08:44 CET] <JEEB> ugh [21:08:54 CET] <JEEB> seriously, are there *any* useful things? [21:09:13 CET] <JEEB> as far as I could tell, it's a goddamn scenario of decoding AVC from a buffer [21:09:29 CET] <JEEB> anything else it does with lavf/lavc? [21:09:39 CET] <Angus__> I have no idea [21:10:20 CET] <JEEB> as far as I could see that thing could be reimplemented just fine with whatever 16.04 comes with [21:10:51 CET] <JEEB> the thing in its current state is FUBAR [21:12:08 CET] <Angus__> I have no idea what it actually does, so porting it over to use the proper API isn't going to be trivial [21:12:44 CET] <JEEB> hacking up random/bad changes from ancient FFmpeg to current FFmpeg blindly is not gonna be any better [21:13:01 CET] <JEEB> + that thing used private APIs that's going to end up in tears as well [21:13:38 CET] <JEEB> you might think it's simpler to go on blindly trying to make it compile, but that's not going to happen [21:13:59 CET] <Angus__> you're probably right [21:14:17 CET] <JEEB> map how much avcodec/avformat is used in that code dump [21:14:29 CET] <JEEB> then you will see how much of a job it would be [21:15:36 CET] <JEEB> and if you're doing this for a company, consider getting someone on board who can habla these things [21:16:01 CET] <JEEB> unless you have the time to learn this yourself of course [21:16:03 CET] <Angus__> I'll probably be giving citrix a call on tuesday [21:16:27 CET] <JEEB> looking at that thing, citrix doesn't really habla FFmpeg [21:16:30 CET] <Angus__> idea is 4 weeks to hack this thing, and VDPAU together [21:16:32 CET] <JEEB> which is what I was meaning [21:16:54 CET] <Angus__> we have someone in the company who knows VDPAU stuff pretty well [21:17:21 CET] <Angus__> but no-one here is from Citrix [21:18:58 CET] <JEEB> anyways, if that is all it does with FFmpeg (decodes AVC), then you could probably get this done in a week and then spend the rest on your VDPAU infra :P [21:19:21 CET] <JEEB> there's a VDPAU hwaccel but you still have to handle a lot of stuff on your side since hwaccel != decoder [21:19:28 CET] <Angus__> that's the idea [21:19:58 CET] <JEEB> not sure how much citrix would help you with regards to the FFmpeg part due to how crudely it was made [21:20:21 CET] <JEEB> but at least it doesn't seem like they are actually doing much at all with FFmpeg :P [21:20:40 CET] <Angus__> I don't even know how much of this they remember, it's not exactly new [21:41:46 CET] <Angus__> https://trac.ffmpeg.org/wiki/HWAccelIntro [21:41:54 CET] <Angus__> hmm, looks like there is a VDPAU implementation [21:42:02 CET] <Angus__> but apparently shouldn't be used? [21:43:14 CET] <JEEB> there was something old, but that has been since removed, there is a newer vdpau hwaccel - that should be utilized [21:44:03 CET] <JEEB> hwaccels are a way of integrating those into the avcodec infrastructure. you will have to do some plumbing on your side as well, though. this differentiates it from the the plain decoders/encoders [21:44:58 CET] <JEEB> quite a few applications, including ffmpeg.c, use the VDPAU hwaccel, so you should have no shortage of examples of its usage [21:45:08 CET] <Angus__> ah [21:45:11 CET] <Angus__> https://ffmpeg.org/doxygen/2.8/group__VDPAU__Decoding.html [21:45:17 CET] <Angus__> is this the deprecated module? [21:45:39 CET] <JEEB> no [21:45:48 CET] <JEEB> I'm pretty sure the old thing is long gone [21:46:01 CET] <JEEB> since even mplayer should be using the new one for years [21:46:19 CET] <JEEB> and it was the only reason to keep the old thing in FFmpeg for as long as it was kept [21:47:16 CET] <Angus__> I see [21:53:14 CET] <wm4> JEEB: it's not gone [21:53:27 CET] <JEEB> oh [21:53:29 CET] <JEEB> jesus christ [21:53:33 CET] <wm4> Angus__: that looks like the new hwaccel stuff [21:53:38 CET] <JEEB> yeah [21:53:43 CET] <JEEB> esp. since it had the HEVC stuff [21:53:45 CET] <wm4> Angus__: I'd stop trying to read the doxygen generated pages [21:54:00 CET] <wm4> it's worse than the source code [21:54:07 CET] <Angus__> :( [21:54:14 CET] <Angus__> makes enough sense though [21:54:28 CET] <wm4> I don't think any of the frequent devs use this [21:54:47 CET] <rcombs> I do sometimes [21:54:52 CET] <wm4> (the doxygen in the source can still be useful, but, well, you can just read it as part of the source) [21:54:52 CET] <rcombs> just for searching/browsing [21:55:03 CET] <rcombs> I end up in the source whenever I want any detail, though [21:55:07 CET] <wm4> rcombs: I use my modern refactoring tools like grep [21:56:04 CET] <Daemon404> wm4, i use the doxy [21:56:08 CET] <Daemon404> but probably only me [21:56:35 CET] <wm4> ok maybe it's not as uncommon... personally I feel like the doxy omits a lot of information [21:56:37 CET] <Daemon404> i google func names and ffmpeg doxycomes up [21:56:48 CET] <Daemon404> less effort than grep [22:02:08 CET] <JEEB> I use the doxy sometimes as it pops up on google for specific public structs/vars [22:11:35 CET] <Angus__> typedef int ( _PAPI PFN_avcodec_decode_video ) (AVCodecContext *avctx, AVFrame *picture, int *got_picture_ptr, const uint8_t *buf, int buf_size ); [22:11:37 CET] <Angus__> strange... [22:13:47 CET] <wm4> ancient api [22:13:59 CET] <wm4> long gone [22:14:17 CET] <Angus__> ahh. so that's why I can't find it [22:14:46 CET] <Angus__> there's no direct replacements for it, is there? [22:15:17 CET] <wm4> avcodec_decode_video2 is direct enough [22:15:24 CET] <wm4> but you need to setup a avpacket [22:15:57 CET] <Angus__> hmm [22:18:34 CET] <JEEB> just look at the decoding example under docs in the git repo [22:31:01 CET] <Angus__> https://ffmpeg.org/doxygen/0.6/avcodec_8h.html#812b0ba18511f3c8787294bce7b0… [22:31:16 CET] <Angus__> my god, it was deprecated by 0.6 [22:33:14 CET] <JEEB> Angus__: you'd probably get much more out of looking at https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/demuxing_decoding… for a bit [22:33:46 CET] <JEEB> since the citrix thing essentially is reading and decoding AVC [22:33:57 CET] <Angus__> what exactly does demuxing do? [22:34:29 CET] <Angus__> as in, what are the inputs/outputs of the demuxing? [22:34:35 CET] <wm4> Daemon404: oh, now I remember... I confused time_base with pkt_timebase [22:34:52 CET] <JEEB> Angus__: takes input and reads it with a demuxer of choice into chunks you can throw into avcodec [22:35:02 CET] <wm4> Angus__: it turns a file (byte stream) into a stream of AVPackets [22:35:12 CET] <wm4> but why your citrix thing does that... who knows [22:35:32 CET] <wm4> maybe to split it into individual h264 packets (but a parser could do this) [22:35:59 CET] <JEEB> basically it most probably was doing what the h264 demuxer for raw AVC does? [22:36:27 CET] <Angus__> hmmm [22:36:38 CET] <Angus__> the Citrix thing streams video to decode [22:37:21 CET] <JEEB> yes, so you would be making a custom avio thing for avformat that feeds it bytes from the buffer (mark it as non-seekable) and then have avformat read it as "h264" [22:37:28 CET] <JEEB> then pass those avpackets into avcodec and voila [22:37:32 CET] <JEEB> you get AVC decoding [22:38:21 CET] <Daemon404> wm4, lulz. [22:43:53 CET] <Angus__> what's the difference netweem modifying the code to use avcodec_decode_video2 and getting it to use avio? [22:44:41 CET] <JEEB> either manually creating avpackets [22:44:46 CET] <JEEB> or getting the framework do it for you [22:45:19 CET] <dinux5> What does sample_buffer contain [22:45:20 CET] <JEEB> because you need AVPackets for the decoder, and demuxers output them [22:45:49 CET] <dinux5> is it an array or just a memory location? [22:48:14 CET] <JEEB> Angus__: this https://github.com/jeeb/matroska_thumbnails/blob/master/src/matroska_thumbn… until line 235 is my babby's first with custom AVIO and avformat+avcodec video demuxing and decoding [22:49:20 CET] <Angus__> mhmmm [22:49:24 CET] <JEEB> I defined the custom seek and read functions, in your case seek wouldn't be needed [22:50:18 CET] <JEEB> you could of course simplify it somewhat even, because you know exactly what you're receiving :P [22:50:35 CET] <JEEB> you don't need to probe or pick "best" streams per se [22:52:00 CET] <Angus__> yep, exactly [22:52:53 CET] <JEEB> and the only reason for having avformat there would be to make sure that the data fed to avcodec matches what it expects [22:53:23 CET] <JEEB> not sure if making custom AVPackets and setting parsing to full in the AVC decoder would suffice [22:53:49 CET] <Angus__> let me figure out how this citrix thing is creating the context, and if it's easier to let it do it's thing, vs taking the bytestream and making it an avformat [23:02:56 CET] <Angus__> I'm putting off the decision of keeping with the custom avcodec_decode_video2 vs avio for now [23:03:26 CET] <Angus__> how does FFMpeg decide what decoder to use? [23:05:23 CET] <Angus__> or would it be easier just to trace the program when it's being run? [23:07:07 CET] <cone-399> ffmpeg 03Michael Niedermayer 07master:0cd9ff4e3aa2: avcodec/libutvideodec: copy frame so it has reference counters when refcounted_frames is set [23:07:54 CET] <Daemon404> and so it lives on [23:07:59 CET] <Daemon404> as the undead [23:08:53 CET] <wm4> this is the first decoder ever that checks thgis field [23:08:55 CET] <wm4> (lol) [23:09:06 CET] <wm4> gotta go fast when using legacy API [23:10:01 CET] <wm4> also this patch was for 30 mins on the ML and then pushed without any replies, what's the point of sending it to the ML? [23:11:31 CET] <Daemon404> carl ok'd it privately according to the msg [23:11:34 CET] <Angus__> anyway, im heading out of work for the night. Thanks for the help, appreciate it. [23:11:40 CET] <Daemon404> i dont care, it's not my problem [23:11:42 CET] <Daemon404> @ am4 [23:11:47 CET] <Daemon404> wm4* [23:12:04 CET] <Daemon404> (that was unfortunate timing) [23:13:06 CET] <wm4> I'm just amazed how this created yet another special case [23:13:29 CET] <wm4> probably harmless, but still [23:13:41 CET] <wm4> (also might encourage users to use legacy API) [00:00:00 CET] --- Fri Mar 25 2016

1 0

ffmpeg.log.20160324
by burek 25 Mar '16

25 Mar '16

[00:05:24 CET] <durandal_170> AlexQ: yes; directly [00:06:31 CET] <AlexQ> I'm afraid that will preserve chapter info and whatnot, but whatever... [00:06:47 CET] <AlexQ> Let's see [00:06:56 CET] <AlexQ> Checked the original file and it plays well [00:07:21 CET] <AlexQ> It this doesn't work, maybe I'll try to mux the original file with the FLAC I extracted or sth [00:09:42 CET] <AlexQ> remuxing fixed the issue. [00:10:23 CET] <AlexQ> both audio and vid from the same file that was wrong, just muxed again. Ehh, so strange [00:10:40 CET] <AlexQ> maybe it was the filters that created the issue, durandal_170 [00:11:25 CET] <durandal_170> now try with demuxer one to do filters [00:11:42 CET] <durandal_170> *remuxed [00:12:01 CET] <AlexQ> um, I am done, actually: the FLAC file was the result of the volume amp filter [00:12:28 CET] <AlexQ> FLAC stream* [00:13:13 CET] <AlexQ> interestingly, Foobar's Dobly Headphone DSP produced 10 dB louder output than VLC's headphone DSP [01:02:09 CET] <isoboy> Hi there [01:02:49 CET] <isoboy> I'm trying to build a ffmpeg sample code from https://ffmpeg.org/doxygen/trunk/encoding-example_8c-source.html [01:03:13 CET] <isoboy> I've managed to get it working, but I'm getting timestamp error [01:03:21 CET] <isoboy> http://pastebin.com/hwGQAGgp [01:04:34 CET] <isoboy> I've tried setting the AVFrame::pts and AVPacket::pts but they did not work. What unit are the timestamps in? milliseconds? seconds? [01:04:55 CET] <isoboy> I'd really appreciate it if someone could point me in the right direction. [01:05:29 CET] <llogan> not many library usage answers here. may want to try the FFmpeg libav-user mailing list [01:09:09 CET] <isoboy> I see. Thanks llogan. [04:02:20 CET] <kuroro> i want to concatenate 3 short videos ( 2 of which are loud, while 1 has low volume). I want to normalize their volumes to be similar to each other. my current attempt of using dynaudnorm filter didnt really work (last clip still has low volume) [04:02:45 CET] <kuroro> here's my ffmpeg command that's not working (http://pastebin.com/raw/euKxspxV) [04:35:21 CET] <kuroro> trying to use dynaudnorm on audio file of concatenated mp4 didnt work either (the results from "sox file.wav -n stat") were the same before and after the ffmpeg command below was executed [04:35:36 CET] <kuroro> ffmpeg -v debug -i norm_tAy6wnTbQUFY_vQ.wav -af dynaudnorm=m=100.0:r=0.3 dynanorm_tAy6wnTbQUFY_vQ.wav [04:35:48 CET] <kuroro> http://pastebin.com/raw/b2mGZJ4X [06:59:43 CET] <parrot1> hi..while trying to link against ffmpeg library I got error /usr/bin/ld: /usr/local/lib/libavdevice.a(xcbgrab.o): undefined reference to symbol 'xcb_xfixes_query_version_reply' . [06:59:45 CET] <parrot1> Any idea? [07:15:06 CET] <thebombzen> parrot1: did you install libxcb correctly? [07:15:34 CET] <thebombzen> make sure you install the development library as well (libxcb-dev or xcb-devel usually) [07:19:05 CET] <parrot1> thebombzen: checking.... [07:25:35 CET] <parrot1> thebombzen: looks like libxcb-devel is installed. I'm using Fedora 23 [07:26:10 CET] <thebombzen> try dynamically linking ffmpeg. that is what you should be doing anyway [07:33:35 CET] <parrot1> thebombzen: ok..thanks for the help :-) [08:52:14 CET] <liquid-silence> hi all [08:52:23 CET] <liquid-silence> ffmpeg -i 1frame.mp4 -ss 4.522133333333334 -t 1 -s 800x600 -f image2 imagefile.jpg [08:52:31 CET] <liquid-silence> extracts the frame after those seconds [08:53:02 CET] <liquid-silence> so if burnin on the video is 00:00:04:15 the image is 00:00:04:16 [08:53:05 CET] <linuxuser9000> Hi. I'm trying to build ffmpeg from source and I was wondering if anyone has a generic command they use for the configure script? I want to be able to convert gif to webm, so I want to enable vp8 or vp9 for example, but I'd like to know if any of you have a better configure command than that [08:53:07 CET] <liquid-silence> its consistent [08:53:26 CET] <liquid-silence> but how do I extract the frame on those seconds? or get the 15th frame on 4th second of the video? [08:59:01 CET] <furq> liquid-silence: -filter:v "select=gte(n\,115)" -frames:v 1 [08:59:04 CET] <furq> assuming your video is 25fps [08:59:11 CET] <liquid-silence> yeah [08:59:27 CET] <liquid-silence> this only works when we add a keyframe to every frame in the video [08:59:45 CET] <furq> it shouldn't make a difference if you're extracting to jpeg [08:59:47 CET] <liquid-silence> we are using an html5 video player [08:59:51 CET] <furq> s/extracting/converting/ [08:59:55 CET] <liquid-silence> that is reporting the incorrect timecode [09:00:08 CET] <liquid-silence> those seconds come from the html5 video tag's current time [09:00:17 CET] <liquid-silence> which is inherently broken [09:00:28 CET] <liquid-silence> so we will never be accurate? [09:00:45 CET] <furq> no idea, i've never had to deal with browser video timecodes [09:00:59 CET] <furq> just typing "browser video timecodes" gave me a headache [09:01:16 CET] <liquid-silence> its a pain in the ass [09:01:18 CET] <liquid-silence> honestly [09:02:20 CET] <furq> doubtless once you do figure it out you'll discover that firefox and chrome do it completely differently [09:03:55 CET] <furq> and that safari doesn't support it, and IE claims to support it but is actually just returning CryptGenRandom() [09:04:04 CET] <furq> web development! [09:09:38 CET] <liquid-silence> pain in my ass man [09:16:55 CET] <linuxuser9000> well i tried configuring everything i could from the encoder section. this should be good [09:23:56 CET] <furq> linuxuser9000: you generally don't need to do much [09:25:07 CET] <furq> you only really need to enable stuff that you intend on using for encoding [09:26:08 CET] <furq> there are only a few obscure-ish codecs which don't have built-in decoders [09:29:48 CET] <linuxuser9000> furq: Thanks [09:29:56 CET] <linuxuser9000> I'm going to retry tomorrow [09:30:46 CET] <linuxuser9000> So the encoding, if I only want to make .aac, .opus, .mp4 and .webm, do i only need encoders for those? and vice versa for decoders for the inputs i use? [09:31:20 CET] <furq> all you need is libopus and libvpx [09:32:37 CET] <linuxuser9000> make clean returns the directory to as if I'd just done a git-pull, right? [09:32:59 CET] <furq> it might be make distclean, i forget now [09:33:06 CET] <furq> one or the other [09:33:28 CET] <linuxuser9000> Thank you [09:33:35 CET] <furq> https://ffmpeg.org/general.html#Video-Codecs [09:33:44 CET] <furq> only codecs with an E under decoding need an external library [09:33:58 CET] <furq> which is maybe three or four audio codecs and no video codecs [09:34:58 CET] <linuxuser9000> argh I need to enable vp8 and vp9 [09:35:04 CET] <linuxuser9000> thanks for the link [09:35:47 CET] <furq> if you want to make mp4 then you probably also want libx264 [09:36:01 CET] <furq> and fdk-aac is a better aac encoder than the builtin one [09:36:17 CET] <linuxuser9000> Thanks again for those tips I'll enable those [09:36:51 CET] <linuxuser9000> I'm going to just ctrl-c my current make and re-issue a new configure command [09:37:11 CET] <furq> don't forget to run make -j [09:37:35 CET] <linuxuser9000> that didn't do anything [09:37:49 CET] <furq> it should be running faster [09:41:33 CET] <NicolasRaoul> Using latest ffmpeg (N-53602-g65cff81-static from johnvansickle) on Ubuntu I get: ALSA lib ../../src/conf.c:3357:(snd_config_hooks_call) Cannot open shared library libasound_module_conf_pulse.so [09:42:01 CET] <NicolasRaoul> Any idea how to solve this? It works if I record only video and not audio. [09:44:22 CET] <liquid-silence> are math question here guys [09:44:40 CET] <liquid-silence> 6.6715555 = 00:00:06:14 @ 25fps [09:44:44 CET] <liquid-silence> how do I remove one frame [09:45:16 CET] <liquid-silence> so when I do [09:45:34 CET] <liquid-silence> ffmpeg -i test.mp4 -ss 6.6715555 -t 1 -s 800x600 -f image2 imagefile.jpg I actually get the following burnin on the video [09:45:40 CET] <liquid-silence> 00:00:06:156 [09:45:45 CET] <liquid-silence> 00:00:06:15 [09:45:47 CET] <liquid-silence> not 14 [09:46:01 CET] <liquid-silence> so I need to subtract one frame from 6.6715555 [10:06:26 CET] <debianuser> NicolasRaoul: Maybe you're running 32-bit ffmpeg on a 64-bit system and you don't have 32-bit alsa-pulse plugin installed? [10:07:00 CET] <NicolasRaoul> debianuser, I downloaded and use http://johnvansickle.com/ffmpeg/builds/ffmpeg-git-64bit-static.tar.xz [10:08:20 CET] <NicolasRaoul> the error does not happen with the ffmpeg from the official Ubuntu 2015.10 repository (which is too old) [10:09:38 CET] <t4nk455> hey :), is there a command to show an images size in ffmpeg? [10:09:47 CET] <JEEB> ffprobe file [10:10:05 CET] <JEEB> you can get json output etc from it as well if you want to probe things within an application [10:11:13 CET] <debianuser> NicolasRaoul: Maybe it looks for pulse plugin in the different directory then? I mean it could be /usr/lib/x86_64-linux-gnu/alsa-lib/libasound_module_pcm_pulse.so or /usr/lib/alsa-lib/libasound_module_pcm_pulse.so. The search path depends on your alsa-lib, but since that's a static build it depends on that static alsa-lib. [10:11:41 CET] <debianuser> NicolasRaoul: You can workaround that by putting an explicit path to that module in your alsa config. [10:12:23 CET] <relaxed> NicolasRaoul: my builds lack pulse support, if that's what you're trying [10:12:39 CET] <t4nk455> thank you :), can you tell me, how to get the json output ? [10:13:40 CET] <NicolasRaoul> relaxed, are you johnvansickle? :-) [10:13:51 CET] <relaxed> yes [10:14:15 CET] <debianuser> NicolasRaoul: put in your ~/.asoundrc something like: pcm_type.pulse { lib "/usr/lib/x86_64-linux-gnu/alsa-lib/libasound_module_pcm_pulse.so" } that would hopefully make it working. [10:14:50 CET] <NicolasRaoul> cool! Thanks for the builds. I am on Ubuntu where pulse is the norm if I understand correctly. I will try that now, thanks! [10:20:09 CET] <debianuser> NicolasRaoul: That may break sound in 32-bit apps like wine/skype/adobe-flash. So if that solves ffmpeg problem but breaks 32-bit apps, you'll need to use ffmpeg-specific config instead: rename it to ~/.asoundrc.64bit for example, and run ffmpeg like: env ALSA_CONFIG_PATH=/usr/share/alsa/alsa.conf:$HOME/.asoundrc.64bit ffmpeg ... [10:20:11 CET] <NicolasRaoul> debianuser, I just created that .asoundrc but still the same "Cannot open shared library libasound_module_conf_pulse.so"... I would create a symbolic link but I don't know where it is trying to open it in the first place. [10:21:19 CET] <debianuser> Yeah, symbolic link may work too. Most probably it looks for /usr/lib/alsa-lib. To check that run: strace ffmpeg ... 2>&1 | grep libasound_module [10:23:43 CET] <debianuser> (hm... it looks for _conf_pulse.so, not _pcm_pulse.so, so the config won't help anyway) [10:29:16 CET] <NicolasRaoul> open("/usr/lib/alsa-lib/libasound_module_conf_pulse.so", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) [10:31:41 CET] <relaxed> t4nk455: man ffprobe-all|less +/json [10:32:29 CET] <NicolasRaoul> Symbolic link created, now I just get Segmentation fault [10:33:19 CET] <gnome1> I got to say that, so far, ffmpeg is working quite well for muxing/merging different streams. bundling subtitles in mkv, or maintaining subtitles when transcoding just works. [10:38:37 CET] <debianuser> NicolasRaoul: `sudo ln -s x86_64-linux-gnu/alsa-lib /usr/lib/alsa-lib` ? [10:43:10 CET] <NicolasRaoul> debianuser, cd /usr/lib/alsa-lib/ ; sudo ln -s /usr/lib/x86_64-linux-gnu/alsa-lib/ libasound_module_conf_pulse.so [10:43:32 CET] <NicolasRaoul> sudo ln -s /usr/lib/x86_64-linux-gnu/alsa-lib/libasound_module_conf_pulse.so libasound_module_conf_pulse.so [10:43:55 CET] <NicolasRaoul> (correct line is the second one) [10:44:11 CET] <debianuser> you'll also need a symlink for libasound_module_ctl_pulse.so and libasound_module_pcm_pulse.so then [10:47:01 CET] <NicolasRaoul> symlinking the whole directory is a better idea indeed! I just did this, same segfault though. Still with ffmpeg -f alsa -ac 2 -i hw:0 -f video4linux2 -framerate 10 -video_size 1280x720 -i /dev/video0 test.mkv [10:49:18 CET] <NicolasRaoul> I wonder if there is any easier way to use ffmpeg 3.0 on Ubuntu... [10:50:19 CET] <NicolasRaoul> I tried compiling yesterday, video quality was not as good, for some unknown reason, it was pixelated. [10:53:11 CET] Action: debianuser guesses the easiest way would be to remove/move /usr/share/alsa/alsa.conf.d/*pulse*.conf files somewhere. [10:56:54 CET] <debianuser> NicolasRaoul: To workaround pulse bug you can try a custom standalone config. Put pcm.myhw { type hw; card 0 } into ~/.asoundrc.ffmpeg and then test `env ALSA_CONFIG_PATH=$HOME/.asoundrc.ffmpeg ffmpeg -f alsa -ac 2 -i myhw ...`. Not sure if that would work, but it should bypass all pulse configs and use just your "myhw" config alone. [10:59:37 CET] <NicolasRaoul> [alsa @ 0x50b8f20] cannot open audio device myhw (No such file or directory) [11:02:01 CET] <debianuser> that's with ALSA_CONFIG_PATH=$HOME/.asoundrc.ffmpeg and pcm.myhw definition in it? [11:07:20 CET] <liquid-silence> can someone here explain drop frames please [11:08:24 CET] <relaxed> liquid-silence: frames that aren't encoded to the output [11:25:25 CET] <lukesan> morning all [11:25:36 CET] <lukesan> first time here [11:30:07 CET] <lukesan> can I post a little script to explain a problem? [11:47:08 CET] <relaxed> lukesan: pastebin.com the script [12:34:46 CET] <lukesan> http://pastebin.com/CZ0gCYpu [12:35:53 CET] <lukesan> I'd like to create a top bottom dual video, with a drag&drop input [12:36:34 CET] <lukesan> for simple conversion works perfectly, but with this video filter something goes wrong [12:41:42 CET] <J_Darnley> don't pad [12:41:55 CET] <J_Darnley> don't use movie source filter [12:42:03 CET] <J_Darnley> do use vstack [12:42:43 CET] <J_Darnley> and "something goes wrong" is not an error message [12:44:42 CET] <lukesan> I need to be more precise [12:45:59 CET] <Razva> guys, any RHEL/CentOS repo with ffmpeg 3.0 available? [12:46:03 CET] <lukesan> [Parsed_movie_1 @ 0000000002d8d6e0] Failed to avformat_open_input 'V' [12:46:03 CET] <lukesan> [AVFilterGraph @ 0000000002d51680] Error initializing filter 'movie' with args ' [12:46:03 CET] <lukesan> V:IReXCOMP04_COMPOSITINGPROGETTITeatro_trampolo_elasticoRENDERTeatro_trampolo_el [12:46:03 CET] <lukesan> astico_R.mp4' [12:46:03 CET] <lukesan> Error opening filters! [12:46:04 CET] <lukesan> Premere un tasto per continuare . . . [12:46:25 CET] <furq> lukesan: https://ffmpeg.org/ffmpeg-filters.html#vstack [12:46:25 CET] <lukesan> soory [12:46:45 CET] <furq> Razva: http://johnvansickle.com/ffmpeg/ [12:46:52 CET] <furq> not a repo but it'll probably be much less hassle [12:46:53 CET] <J_Darnley> Another reason to not use the movie filter: you need to escape lots on Windows [12:47:02 CET] <furq> based on my suppressed memories of centos [12:47:15 CET] <lukesan> http://pastebin.com/vCG33JkK [12:48:40 CET] <J_Darnley> And I point back to my previous line [13:00:53 CET] <lukesan> thanks J_Darnley I'm going to try the vstack that seem to be interesting [13:07:24 CET] <satinder___> Hi , I am getting following errors when using drawtext ffmpeg filter with Textfile and reload = 1 [13:07:49 CET] <satinder___> [Parsed_drawtext_0 @ 0x3a88ee0] [FILE @ 0x7fff05b4e4a0] Error occurred in mmap(): Invalid argument [13:07:49 CET] <satinder___> [Parsed_drawtext_0 @ 0x3a88ee0] The text file '/home/satinder/OverlayInfo' could not be read or is empty [13:07:49 CET] <satinder___> Failed to inject frame into filter network: Invalid argument [13:07:49 CET] <satinder___> [video4linux2,v4l2 @ 0x3a84460] Some buffers are still owned by the caller on close. [13:08:12 CET] <satinder___> please any body help me , what I am doing wrong [13:08:24 CET] <satinder___> my command is following [13:08:51 CET] Action: relaxed whispers pastebin.com [13:09:00 CET] <satinder___> ffmpeg -re -i /dev/video0 -vf drawtext='fontsize = 20 : fontfile = /usr/share/fonts/truetype/freefont/FreeSansBold.ttf : textfile = %s : reload = 1' -f v4l2 /dev/video1 [13:09:19 CET] <satinder___> %s = Textfile [13:11:02 CET] <relaxed> try removing all the spaces from the filtergraph [13:11:23 CET] <relaxed> if that doesn't work pastebin.com the command and all console output [13:12:57 CET] <satinder___> relaxed : Sir that is working for 16 mins after that giving above errors [13:14:08 CET] <satinder___> that means not any issue in command that generating error due to some other reasons [13:14:16 CET] <satinder___> but I am not sure [13:14:18 CET] <satinder___> :( [13:15:14 CET] <J_Darnley> You could always read the error message. [13:15:32 CET] <J_Darnley> I assume this is your non-atomic updating coming back to bite you. [13:16:24 CET] <satinder___> J_Darnley : Sir , I think you said right [13:16:50 CET] <satinder___> because the error console show [Parsed_drawtext_0 @ 0x3a88ee0] The text file '/home/satinder/OverlayInfo' could not be read or is empty [13:17:15 CET] <satinder___> J_Darnley : what I can do resolve that issue , Sir [13:17:32 CET] <J_Darnley> Someone told you yesterday (or whenever) [13:17:45 CET] <J_Darnley> "rename" was the answer [13:18:10 CET] <t4nk455> Does anyone have an Idea, how to begin with the search how to stream from the virtual cameras inside a 3D Engine ? [13:18:26 CET] <satinder___> J_Darnley : yes sir [13:18:42 CET] <satinder___> but I am using a C program for updating that file [13:18:57 CET] <satinder___> that is following [13:19:19 CET] <satinder___> while (1) { [13:19:19 CET] <satinder___> writeStructure(OverlayParam); [13:19:20 CET] <satinder___> EXIT_CODE = CheckOverlayValues(OverlayParam); [13:19:20 CET] <satinder___> if (EXIT_SUCCESS == EXIT_CODE) { [13:19:20 CET] <satinder___> file = fopen(TextFile, "w+"); [13:19:20 CET] <satinder___> if (NULL == file) { [13:19:22 CET] <satinder___> CAMERA_ERROR("Overlay output file not opened "); [13:19:24 CET] <J_Darnley> OMFG! [13:19:24 CET] <satinder___> exit(EXIT_FAILURE); [13:19:26 CET] <satinder___> } else { [13:19:28 CET] <satinder___> CAMERA_DEBUGL("OverLay outfile opened successfully"); [13:19:30 CET] <satinder___> EXIT_CODE = Write_TextFile(file, OverlayParam); [13:19:30 CET] <J_Darnley> FUCK OFF [13:19:32 CET] <satinder___> if (EXIT_SUCCESS == EXIT_CODE) { [13:19:34 CET] <satinder___> fclose(file); [13:19:36 CET] <satinder___> } [13:19:38 CET] <satinder___> } [13:19:42 CET] <satinder___> } [13:19:44 CET] <satinder___> sleep(1); [13:19:46 CET] <satinder___> } [13:19:53 CET] <J_Darnley> ignored [13:20:03 CET] <satinder___> ?? [13:20:09 CET] <JEEB> satinder___: for future reference, never ever paste long text onto an IRC channel [13:20:18 CET] <JEEB> you will get hated and condemned [13:20:29 CET] <JEEB> use a pastebin-like sane service that you can link to [13:20:47 CET] <satinder___> JEEB : sorry !! I don't not [13:20:50 CET] <satinder___> Sir [13:21:00 CET] <satinder___> Next time that will not happened [13:21:20 CET] <satinder___> But please help what is wrong in that method [13:22:54 CET] <IntelRNG> It floods the IRC clients of everybody in the room [13:24:27 CET] <t4nk455> Does anyone have an Idea, how to begin with the search how to stream from the virtual cameras inside a 3D Engine ? [13:24:29 CET] <furq> weren't you advised to do an atomic update of the file [13:24:31 CET] <jkqxz> Write to a temporary file ('file = fopen("blah", ...);') and then rename it to the thing you want atomically after you've finished writing it ('fclose(file); rename("blah", TextFile);'). [13:24:55 CET] <furq> e.g. `echo foo > /tmp/overlay; mv /tmp/overlay /home/satinder/OverlayInfo` [13:24:55 CET] <J_Darnley> t4nk455: like any other thing: use the ffmpeg libraries [13:25:01 CET] <furq> or yeah what jkqxz said [13:25:38 CET] <satinder___> J_Darnley : that is possible renaming a file in c [13:26:10 CET] <furq> http://linux.die.net/man/3/rename [13:26:20 CET] <satinder___> ok [13:26:52 CET] <satinder___> furq : you mean update value in file and then rename it [13:27:13 CET] <satinder___> thanx if that will works for me , Sir [13:29:55 CET] <satinder___> But I don't understand what happened when I am truncating Textfile and writing new values in this [13:30:18 CET] <satinder___> furq : ?? [13:45:24 CET] <lukesan> vstack still doesn't work with windows variables [13:46:13 CET] <J_Darnley> If that is the only thing you changed then you completelyignored the error [16:37:14 CET] <fatpelt> good afternoon all! i'm having an issue with ffmpeg version git-2016-03-08-b60dfae (should be a recent build). i'm using overlay to put 4 video streams in a mosaic. that works fine, but as soon as i put an audio stream in my complex filter ffmpeg dies when one of the content switches ac3 audio layout mid-stream. http://pastebin.com/VHx9si6m if i remove line 10 from the command and rerun it'll run all day long. i've tried a ton of different filters but i [16:37:16 CET] <fatpelt> "Input stream #1:1 frame changed from rate:48000 fmt:fltp ch:2 chl:stereo to rate:48000 fmt:fltp ch:6 chl:5.1(side) [16:37:16 CET] <fatpelt> [Parsed_overlay_28 @ 0x42fe540] [framesync @ 0x1f86fe8] Buffer queue overflow, dropping.drop=58 speed=0.983x" [16:39:44 CET] <durandal_1707> fatpelt: better use vstack/hstack for mosaic [16:40:01 CET] Action: fatpelt googles that filter [16:40:20 CET] <fatpelt> ooh [16:40:21 CET] <fatpelt> tasty [16:40:34 CET] <fatpelt> let me re-write this command and see what happens [16:47:21 CET] <momomo> anyone here used clappr ? [16:48:10 CET] <andrey_utkin> is it correct that i can concatenate H.264-in-MP4 videos from different origins if input clips were processed with "dump_extra" bitstream filter? I know that joining works if you convert all clips to AnnexB-in-MPEGTS and then back, now it seems dump_extra does that more efficiently, is this correct? [16:49:00 CET] Last message repeated 1 time(s). [16:50:38 CET] <andrey_utkin> oh sorry for repetition [16:51:30 CET] <lroe> Does anyone know if there is a way to enable a 'record' button on the controls of an HTML5 video stream? [16:52:35 CET] <lroe> I'm serving an rtsp stream using the html5 native player. I have enabled controls so people can go back in time a bit, but I'd love to allow people to download a clip they're watching [16:54:43 CET] <andrey_utkin> is anybody seeing my message? seems my xmpp-irc gateway misbehaves [16:54:50 CET] <lroe> andrey_utkin, I see it [16:54:55 CET] <andrey_utkin> thanks lroe [16:55:18 CET] <andrey_utkin> about dump_extra too? [16:56:49 CET] <fatpelt> ok. i've rewritten my complex filter to use *stack instead. i've got the one audio stream with only an anullsink to see if i can reproduce [16:57:40 CET] <andrey_utkin> (just in case. Sorry if you've already received this.) is it correct that i can concatenate H.264-in-MP4 videos from different origins if input clips were processed with "dump_extra" bitstream filter? I know that joining works if you convert all clips to AnnexB-in-MPEGTS and then back, now it seems dump_extra does that more efficiently, is this correct? [17:01:22 CET] <fatpelt> durandal_1707: still crashes on the ac3 layout change [17:02:07 CET] <pyro25> hi there! I've got this funny problem: a platform with no video4linux2 on it and I need to capture the webcam. I see so many applications, like streamer, with which I can capture but I just can't figure what they use =P [17:02:09 CET] <durandal_1707> you mean it changes channels midstream? [17:03:58 CET] <explodes_> My team is working on a video player for Android. The great thing about it is that it supports variable speed playback. We just released our app using the library to 20% of our users. Unfortunately, the crash rate is ridiculous (7%) and they're all C-level errors [17:04:52 CET] <explodes_> My team and myself is pretty much inexperienced in the way of C, so tracking down what I believe to be memory errors has been an arduous process for us. [17:06:54 CET] <durandal_1707> so you ask for free support? [17:14:07 CET] <fatpelt> durandal_1707: yeah. the ac3 audio layout changes mid stream. [17:16:16 CET] <durandal_1707> fatpelt: than you need to transcode it first [17:17:17 CET] <fatpelt> durandal_1707: i've tried a couple of different ways, and i'm sure i'm doing it wrong. i've got -ac 1 *before* all the -i inputs. shouldn't that transcode it to single channel? [17:18:14 CET] <sfan5> most likely not [17:18:17 CET] <durandal_1707> no, you need it after inputs [17:18:29 CET] <sfan5> it would tell ffmpeg to "interpret" the inputs as with 1 audio channel [17:26:06 CET] <fatpelt> ok. so, i moved the -ac2 to after the inputs and now the showvolumes is showing more than 2 channels of audio [17:27:12 CET] <fatpelt> oh&. i moved it back up to the top and it shows audio on showvolumes still anyway [17:29:16 CET] <explodes_> durandal_1707: I don't know of a better place to go looking for someone to hire than at or near the source, so I'm here [17:36:35 CET] <fatpelt> Input stream #3:1 frame changed from rate:48000 fmt:fltp ch:6 chl:5.1(side) to rate:48000 fmt:fltp ch:2 chl:stereo [17:36:54 CET] <fatpelt> right after that i get a ton of Buffer quey overflow, dropping messages and the video dies [18:48:01 CET] <durandal_1707> fatpelt: midstream changes are simple not supported [18:56:20 CET] <petecouture> Is there anyone on that has best practices for encoding a Live HLS stream. Mine works in some players like Flash and HLS.js but desktop players like VLC say it can't detect the format. Also timed metadata/ID3 tags don't get detected. Here's my config: http://pastebin.com/V5qV9pBc [18:56:33 CET] <petecouture> ffprobe returns no errors [18:59:59 CET] <blue_misfit> does media autobuild suite actually work for anyone right now? [19:01:01 CET] <llogan> what's a media autobuild suite? [19:01:23 CET] <blue_misfit> it's a package for Windows that sets up a build environment and builds ffmpeg among other related things [19:01:30 CET] <blue_misfit> allegedly. I've never gotten it to work tho :) [19:02:37 CET] <llogan> it's not from FFmpeg. you should contact the author. [19:04:21 CET] <blue_misfit> indeed, I will. I just thought I'd ask in case others are familiar with this. [19:05:54 CET] <llogan> there is also this which is mentioned on the zeranoe ffmpeg forum now and then: https://github.com/rdp/ffmpeg-windows-build-helpers [19:08:09 CET] <Angus_> Hey guys [19:09:12 CET] <JEEB> don't spam [19:09:23 CET] <JEEB> if you've got a user question, stay here [19:09:35 CET] <JEEB> if you've got a thing regarding development of FFmpeg itself, then maybe -devel [19:18:13 CET] <petecouture> spam spam spamity spam [20:03:16 CET] <anotherRandomGuy> the docs for libmp3lame wrapper of ffmpeg say this: "Set bitrate expressed in bits/s for CBR or ABR. LAME bitrate is expressed in kilobits/s. " [20:03:34 CET] <anotherRandomGuy> does this assume that 1 kilobit equals 1024 bits, or rather 1000 bits? [20:05:12 CET] <anotherRandomGuy> or maybe a bit different question, when an MP3 file is let's say 192 kbps CBR, is it in fact 192000 bits per seconds, or actually 196608 bits per second? [20:06:04 CET] <JEEB> usually when talking of bits in multimedia a kilo means 1000 [20:11:33 CET] <anotherRandomGuy> I tried encoding the same source file using both 192000 and 196608 and it turns out, I got exactly the same file from both runs [20:11:56 CET] <anotherRandomGuy> and by that I mean the checksums do match. So I guess it doesn't matter which of them you use anyway [20:12:57 CET] <JEEB> that's a thing with the specific encoder you're using [20:13:05 CET] <JEEB> it most probably only supports specific bit rates when it bit rate mode [20:13:32 CET] <JEEB> you can't really pull parallels between that and all the other encoders available through avcodec [20:14:38 CET] <anotherRandomGuy> so what you mean is what might be true for one codec in ffmpeg, not necessarily has to be true for some other? I'll keep that in mind, thanks [20:49:18 CET] <explodes_> /buffer 2 [20:49:20 CET] <explodes_> nice. [21:12:21 CET] <anotherRandomGuy> is it possible to make ffmpeg put different encoder metadata to the output files? right now, the encoder field is set to "Lavf57.29.100", but I'd like it to say "Lavf57.29.100 libmp3lame" instead [21:13:30 CET] <anotherRandomGuy> I tried -metadata encoder="foo", -metadata ENCODER="foo", -metadata TSSE="foo", -metadata tsse="foo", none of which worked [21:13:38 CET] <fatpelt> hey all. i'm back with a question on ac3. i've got a stream that changes the layout mid stream, and when it does things all go to pot and ffmpeg starts to drop packets [21:14:00 CET] <fatpelt> within mpeg-ts, i'm not seeing that it isn't supported, [21:14:56 CET] <c_14> fatpelt: can you copy the stream to a file and reproduce with that? If yes, open a bug report on trac with the file as a sample. [21:15:40 CET] <fatpelt> c_14 i'll have to see if i can do that [21:16:30 CET] <fatpelt> oddly though, it only crashes when i add something like "[0:a] SOMEFILTERINGSTUFF" to my complex filter. it even crashes when i use anullsink by itelf [21:16:34 CET] <c_14> anotherRandomGuy: it seems to work for me with the stream metadata field but not the global one [21:16:46 CET] <fatpelt> if i don't have that in the filter, it works just peachy and will run all day long [21:17:02 CET] <c_14> fatpelt: reencoding the audio? [21:17:11 CET] <anotherRandomGuy> c_14: you mean the g and s switches, right? I'll try with these as well [21:17:23 CET] <c_14> anotherRandomGuy: yes [21:17:59 CET] <c_14> fatpelt: hmm, actually. it might just be a problem that libavfilter doesn't support mid-stream layout changes well (or at all). Still bug-worthy though [21:21:12 CET] <anotherRandomGuy> c_14: indeed, '-metadata:s encoder="foo"' does change the stream metadata, thank you. any idea for the global one though? [21:21:22 CET] <kwivix> Hi, i'm playing around with ffmpeg and ffserver, is it possible to 'capture' a stream and 'forward' it to a ffserver? [21:22:15 CET] <fatpelt> c_14, in the short term, is there *any* way to downsample them *before* it hits libavfilter? [21:22:37 CET] <llogan> kwivix: unfortunately nobody here really knows how to use ffserver (or uses it AFAIK). it is basically unmaintained. [21:24:28 CET] <c_14> fatpelt: the only downsampling methods I know would use libavfilter... [21:24:48 CET] <fatpelt> :) heh. ok. been there and tried that with resample [21:32:55 CET] <anotherRandomGuy> c_14: adding 'fflags +bitexact' makes the TSSE tag go away completely [21:33:23 CET] <J_Darnley> strange, it usually gets degraded to "ffmpeg" when you do that. [21:34:03 CET] <anotherRandomGuy> this behavior suits me, I didn't want that tag anyway, the stream one is what I need and this one stays so it's perfect [21:37:05 CET] <JEEB> how muxers decide to follow the bitexact flag depends a lot [21:37:23 CET] <JEEB> since its main part is that it makes the output exactly the same between runs between versions [21:37:37 CET] <JEEB> so as long as the output isn't specific to the version, it flies [21:38:49 CET] <anotherRandomGuy> I also noticed the stream "encoder" field is limited to only 9 characters, so even the default "Lavc57.28.103 libmp3lame" becomes "Lavc57.28" [22:55:24 CET] <Wader8> hello [22:58:06 CET] <Wader8> I'm wondering, is 2 video stream merging possible anywhere out there, specifically 2 videos with different aspect to create a video which has extra pixels from a secondary video, so it would work that a priority has to be selected which one retains the main part [22:58:24 CET] <J_Darnley> yes [22:59:59 CET] <Wader8_> either way you always get to lose some pixels, whichever you watch so it kinda makes it half-baked [23:00:08 CET] <Wader8> and I didnt' get any msgs since my connection went out [23:00:15 CET] <J_Darnley> You need to be more precise. [23:00:26 CET] <J_Darnley> "create a video which has extra pixels from a secondary video" [23:00:41 CET] <J_Darnley> where should those "extra pixels" go? [23:01:09 CET] <Wader8> well there where the primary source video doesn't have them [23:01:19 CET] <Wader8> I will explain [23:01:21 CET] <J_Darnley> Huh? [23:01:34 CET] <J_Darnley> The "primary" video has pixels everywhere [23:01:41 CET] <J_Darnley> There are no holes [23:01:55 CET] <J_Darnley> Yes you should explain. [23:08:48 CET] <Wader8> actually, i need to do a quick test, I already have it written, might really work, moment [23:09:14 CET] <Wader8> there's a problem that I didn't foresaw in my thinking, only noticed now, so I need to do a test [23:11:04 CET] <pzich> do you want to share your command and thinking so we can familiarize ourselves with it while you're testing? [23:19:29 CET] <Wader8> yeah, the point is, if it doesn't pan out in practise manually just one screenshot then it won't work even if i do the best explanation [23:29:24 CET] <Wader8> okay just as I suspected, the zoom issue, so it wouldn't fill the entire top and bottom horizontal space, at least in 16:9 and 4:3 case [23:31:11 CET] <Wader8> and the codec would have to do some content analysis to calibrate the enlargement of the secondary video [23:37:32 CET] <J_Darnley> I still don't get it. [23:37:52 CET] <J_Darnley> Are you trying to pad with another video rather than black? [23:38:05 CET] <Wader8> im writting the explanation I started earlier ... it's taking some time [23:38:19 CET] <J_Darnley> okay [23:38:34 CET] <J_Darnley> I will wait [23:53:08 CET] <Wader8> Basically, you'd have a 16:9 1080p video, and a 4:3 720x480 video, and you'd want to merge them (needs fancy name), you'd pick the 16:9 as primary in this case (priority) so that would be treated as a front layer, the other one will be in background, the codec would create a new blank video plane with extra vertical pixels in this case as there is more content on top and bottom with 4:3,... [23:53:10 CET] <Wader8> ...but that would be based on a result from a calculation, first the codec would enlarge the secondary video in a calibrating process with the primary video to establish the center point, so the secondary video is properly enlarged while keeping it's aspect intact to fit perfectly with the content in the primary video, after that, the codec would be able to see the difference how much higher... [23:53:11 CET] <Wader8> ...the secondary video's in-this-case vertical pixels differ from the first video, so it would create a new video with these dimensions to accomodate both videos, it would add enough height so the secondary video's top and bottom can be included, however this will always result in a video with 4 always black areas, one in each corner, as the content doesn't exist there so it's just rendered... [23:53:13 CET] <Wader8> ...as black, the codec will then offer options to enhance secondary video color properties and other things to better blend in with the primary video if the user chooses, the downside is that the video cannot be displayed on a native screen, you'd always need higher resolution monitor to display native 1080p and the extra vertical area, otherwise the video will get downscaled to fit... [23:53:14 CET] <Wader8> ...perfectly with the added vertical content [23:53:33 CET] <Wader8> J_Darnley, there we go [23:53:47 CET] <Wader8> http://i.imgur.com/kdI7VDM.png [23:54:12 CET] <Wader8> Varied Aspect Ratio Video Content Merging - VARVCM :p [23:55:29 CET] <J_Darnley> I think I was right when I said "pad with another video rather than black". [23:55:45 CET] <J_Darnley> But I have no clue how "codec" comes into play. [23:56:32 CET] <Wader8> whatever, the program, that's gonna do this beforer the codec, I'm not sure if this could be possible without codec having some kind of idea about it [23:56:43 CET] <J_Darnley> or maybe this is "pan and scan" [23:57:20 CET] <Wader8> I was looking for the first term, first i took Dissimilar, but then I recalled it could simply be called variable VCM [23:57:43 CET] <Wader8> ala VVCM [23:59:32 CET] Action: llogan lost some IQ trying to read that [00:00:00 CET] --- Fri Mar 25 2016

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ffmpeg-devel.log.20160323
by burek 24 Mar '16

24 Mar '16

[01:15:28 CET] <jkqxz> Without a really very strong justification, a total 0.1% speedup for a nebulous change is far too small to avoid Producing Wrong Data Without Doing Anything Obviously Wrong <http://www-plan.cs.colorado.edu/diwan/asplos09.pdf>. [02:36:25 CET] <newbe-user> \help [07:54:46 CET] <midgetspy> Hey guys I'm working with libav* inside Kodi and noticed that ffmpeg doesn't implement cropping in its demuxer (https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/matroskadec.c#L432) Is this intentionally omitted? If I were to add it and submit a patch would that be welcome? If so I'd appreciate any guidance about where to start, diving right into the codebase is a little daunting [07:54:46 CET] <midgetspy> :0) [08:08:18 CET] <wm4> midgetspy: there's no API concept for cropping and thus can't be implemented [08:13:55 CET] <midgetspy> wm4: what do you mean by "there's no API concept for cropping"? [08:14:28 CET] <wm4> no way to export cropping etc. [08:17:04 CET] <midgetspy> I'm not familiar with the ffmpeg architecture so sorry if I'm being dense... do you mean that the ffmpeg API has no concept of cropping and so there's no point reading it from the matroska container? [08:17:07 CET] <ubitux> apparently, at least cinedec is exporting crop metadata in the stream [08:17:30 CET] <ubitux> so the same could be done for matroska [08:17:49 CET] <ubitux> but i suppose the new trending way of exporting data is through side data [08:20:00 CET] <midgetspy> My (naive) thought after brief investigation was that I would add it to AVStream and then populate it in the matroska demuxer... of course I had a feeling it wouldn't be that simple (not that that was going to be simple, heh) [08:20:42 CET] <midgetspy> it seemed to me that it's a property of a video stream so it'd make sense to have it in AVStream along with width/height [08:20:52 CET] <wm4> sure the demuxer could just export it (even in a demuxer-specific way), but I think it shiuld be added to AVFRame too [08:21:53 CET] <wm4> there's also a question how matroska cropping works (its not clearly defined at all), and how it interacts with codec cropping [08:23:05 CET] <midgetspy> in my case I have been attempting to follow the VLC implementation for Kodi [08:24:11 CET] <wm4> following the mpc-hc implementation would be a better idea [08:25:03 CET] <wm4> anyway, what I find worrisome here is how to add it to AVFrame [08:25:17 CET] <wm4> so far everything assumes that there's no cropping [08:25:39 CET] <wm4> and codec cropping (like in h264) is handled by adjusting the width/height and the plane data pointers [08:26:02 CET] <wm4> (the latter doesn't work for hw decoding, so top/left cropping is omitted) [08:27:56 CET] <midgetspy> ok I'll take a look at mpc-hc. So is your suggestion I implement it in the matroska demuxer specifically and then put it in the AVStream->side_data? I'm not sure I understand how side_data is packed just looking at the headers but I'll look into it. I also see AVStream->metadata but that seems like not what it's meant for? [08:29:16 CET] <wm4> AVStream side data might be fine [08:30:12 CET] <wm4> anyway, my worry is about how to nicely integrate it into ffmpeg's ecosystem so that the result isn't just a mess that somehow sometimes allows to get cropping params while it doesn't work in general, or so [08:36:12 CET] <midgetspy> Makes sense, I'm glad I asked here before starting hacking around in the code. I have only the very narrow technical problem that I'm using ffmpeg as a demuxer in Kodi and it gives me no way to access the matroska crop values. If you have guidance of how I can solve that problem in a way that fits into the ffmpeg ecosystem I'm all ears. I'd like to find a solution that'd be accepted into [08:36:12 CET] <midgetspy> ffmpeg master :0) [08:37:47 CET] <midgetspy> (In fact my real problem is that I want to crop black bars off a video file without re-encoding it and my plan is to solve that with matroska's cropping params as that's the only solution I could find) [08:38:44 CET] <JEEB> ugh matroska cropping [08:39:15 CET] <JEEB> originally meant for avc parameter value override in the demuxer [08:40:13 CET] <JEEB> and then people noticed it and started using it for stuff like cropping a 4:3 image out of a 16:9 encoded blu-ray [08:40:59 CET] <JEEB> also the cellar people seem to have noticed how badly it's defined [08:42:13 CET] <midgetspy> Heh, I want to use it for cropping a 2.40 image out of a 16:9 blu-ray [08:43:03 CET] <JEEB> yes, same thing and most probably wouldn't even work with the demuxer that defined the feature :) [08:43:55 CET] <JEEB> because that is way more than what you can override by hacking parameter set values [08:44:39 CET] <JEEB> wonder if cellar will just drop it or of it will stay as a badly defined thing in there, too [08:45:16 CET] <JEEB> i wonder if the cropping-like metadata in isobmff is in any way better [08:46:55 CET] <midgetspy> well personally I'd still like to have the demuxer able to read it so that the player is able to make assumptions about its implementation if it so chooses ;-P [08:49:10 CET] <JEEB> I'm not against having well defined side data for cropping or whatever, but I just have a thing against the matroska field [08:51:35 CET] <midgetspy> ok well I will proceed and see if I can figure out how to read it in the demuxer and add it to the side channel. What is the desired way to get feedback on it as I work, ask in here? Mailing list? I hope in the end it will be accepted into master so I'd like to get some feedback sooner than later so I don't waste my time if the change isn't wanted [08:54:38 CET] <midgetspy> What's the difference between AVStream->metadata and AVStream->side_data? Other than the data structure... [08:57:34 CET] <JEEB> as for the side data it shouldn't leave the player guessing [08:58:00 CET] <JEEB> it should define at which data the cropping it should be applied [08:59:01 CET] <JEEB> decoded data, decoded and decoder cropped data, image after aspect ratio has been applied are at least three different points of possible cropping [09:12:49 CET] <wm4> it's still not clear how matroska and codec aspect ratio alone interact either (without any cropping present) [09:15:13 CET] <JEEB> I recommend looking at how 14496-12 defines cropping if it has anything like that [09:32:51 CET] <cone-649> ffmpeg 03Paul B Mahol 07master:b098e1a46975: avcodec/dvaudiodec: now that we got samples, fix 12bit case [11:36:11 CET] <michaelni> midgetspy, it wasnt mentioned but as its related see AVPanScan, the struct can defines per field croping rectangles. (The name pan scan originates from mpeg1/2 terminology) [11:37:21 CET] <michaelni> also there can be multiple such rectangles in side data of AVFrames [11:40:34 CET] <michaelni> atomnuker, theres a patch for a qualification task for truhd on ffmpeg-devel (0321 21:29 Disha Singh (135K) [FFmpeg-devel] [PATCH]Addition of MLP encoder), you are listed as Qualification Mentor on https://trac.ffmpeg.org/wiki/SponsoringPrograms/GSoC/2016-Qualis [11:43:45 CET] <kierank> so, who has outreachy students? [12:17:42 CET] <wm4> how is ffmpeg CLI supposed to behave if timestamps go backwards? drop the frame or apply non-working magic? [12:21:15 CET] <atomnuker> michaelni: I know, I keep in touch with Disha [12:21:28 CET] <atomnuker> still has some problems left to solve [12:22:54 CET] <michaelni> atomnuker, i meant someone should review the patch on ML [12:25:51 CET] <jkqxz> wm4: IMO, apply non-working magic and warn about it. I don't think you should ever drop frames unless the user has explicitly said you can (excepting cases where constrained resources force you to). [12:26:47 CET] <wm4> well it seems it's normal, and just doesn't happen in fate-cavs because of crazy interactions of insane code [12:33:08 CET] <ismail> wm4: semi-related, do you know if cavs used in the wild? [12:33:51 CET] <kierank> neeraj: where is fuzz3.avi? [12:33:54 CET] <durandal_170> people mail me to send them ffmpeg builds.... [12:38:21 CET] <neeraj> sorry the name of the file was fuzz2 which i pasted [12:38:28 CET] <neeraj> sorry uploaded [12:40:16 CET] <wm4> ismail: no and I don't care, I just had to deal with its fate test [13:33:49 CET] <BBB> my fingers are one big race condition today [14:57:23 CET] <ubitux> yay hscale aarch64 working. [15:23:06 CET] <ac_slater> hey all. I've made an openmax h264 encoder interface for libavformat. My only use case for this currently is to mux with mpegts and write to a file or send via udp/unix socket. With all three of these outputs, I'm seeing a 0-20second lag from input to output. ffmpeg is claiming my encoder is running at >= 1.00x. Is there a buffer between encoder and output mux (mux and output) that I can control? [15:39:35 CET] <Compn> ac_slater : you want zero latency [15:40:16 CET] <Compn> udp sounds like it would be slow [16:49:56 CET] <rcombs> that was fast [16:51:07 CET] <atomnuker> rcombs: is the apple aac encoder chosen over the native one by default? [16:51:12 CET] <rcombs> no [16:51:13 CET] <jamrial> wait, these audiotoolbox decoders and encoders have more priority than the native ones? [16:51:19 CET] <jamrial> that shouldn't happen [16:51:27 CET] <rcombs> or, if it does, something's badly wrong [16:52:20 CET] <rcombs> oh, he did `--disable-encoders` [16:52:29 CET] <rcombs> so yeah, it's the only one available so it gets picked [16:53:03 CET] <jamrial> funny he request that it should fallback to another encoder instead of exit with error, then :p [16:53:13 CET] <jamrial> when he purposely disabled every other encoder [16:53:19 CET] <rcombs> kinda hard to do that when you don't build anything else :3 [16:59:43 CET] <rcombs> ah, it's a bitrate issue [17:26:26 CET] <rcombs> &if you ask what the applicable bitrates are for ALAC, the function returns `'who?'` [17:26:53 CET] <rcombs> erroring there makes sense, but nice return code [18:01:28 CET] <ac_slater> Compn: sorry for the late reply. My h264 encoder is doing baseline 3.1 at 15 fps with intraperiod of 15 (IDR frame every second). Isn't this what x264 does? If not, then that's probably my next step [18:10:39 CET] <cone-224> ffmpeg 03Ganesh Ajjanagadde 07master:8dbffda0f940: lavc/psymodel: check for av_malloc failure [18:49:32 CET] <TD-Linux> ac_slater, how small do you need the delay to be? [19:08:53 CET] <cone-224> ffmpeg 03Rostislav Pehlivanov 07master:14b41e061db6: vc2enc: macro out slice size rounding [19:08:54 CET] <cone-224> ffmpeg 03Rostislav Pehlivanov 07master:a010e36f23df: vc2enc: correctly zero out the slice size cache [19:08:55 CET] <cone-224> ffmpeg 03Rostislav Pehlivanov 07master:a14d4c076dde: vc2enc: simplify calc_slice_sizes() [20:29:32 CET] <cone-224> ffmpeg 03Michael Niedermayer 07master:fe3de6bc6222: fate: Add test that also decodes packed_bframes.avi [20:34:00 CET] <ac_slater> TD-Linux: sorry for the delay mate, < 2 seconds consistent [20:34:41 CET] <ac_slater> TD-Linux: Somehow I can't seem to determine where the growing latency is coming from ... :( [21:06:18 CET] <rcombs> man, I forgot about the clusterfuck that is 7.1 AAC [21:07:33 CET] <nevcairiel> 7.1 without PCE is silly [21:08:42 CET] <rcombs> pity audiotoolbox doesn't provide a way to encode PCE [21:09:10 CET] <rcombs> there's a whole infrastructure for passing encoders a list of channels in a specific order [21:09:27 CET] <rcombs> and if you use it with AAC it gives a "format unsupported" error [21:11:45 CET] <rcombs> nevcairiel: btw, in re: audiotoolbox encoders existing, at_aac is an order of magnitude or two faster than the internal encoder at the moment [21:12:04 CET] <rcombs> well, probably less so if I built the internal encoder with more optimizations enabled [21:12:11 CET] <rcombs> but still it's fast [21:12:34 CET] <atomnuker> 1-2 orders of magnitude means 10 to 100 times faster [21:12:54 CET] <rcombs> yeah, it was here, but I said that and then remembered I have most optimizations disabled [21:13:36 CET] <nevcairiel> ie. you dont have any real numbers? :D [21:14:01 CET] <rcombs> if nothing else it's useful for comparisons [21:15:32 CET] <wm4> rcombs: faster at same quality? (also, how do you measure quality) [21:32:38 CET] <wm4> oh, nice spam attempt [21:36:57 CET] <rcombs> alright, at similar bitrates AT is between 2x and 5x faster depending on channel count (and probably other parameters but I cba to test more right now) [21:37:16 CET] <RiCON> rcombs: do those enc/decs work with apple application support in windows or is it osx-only? [21:37:25 CET] <RiCON> i assume no [21:37:45 CET] <wm4> rcombs: I guess the internal encoder still has a long way to go... [21:37:48 CET] <rcombs> RiCON: in theory you can probably link this against the Windows AudioToolbox DLL, but I don't have anything setup to do that in configure [21:38:05 CET] <wm4> or maybe Apple is actually good at doing fundamental multimedia components [21:38:12 CET] <rcombs> and then the redist police would come after you [21:38:41 CET] <rcombs> wm4: their AAC encoder is like the only good encoder they've ever done [21:39:11 CET] <wm4> what about the other encoders? [21:39:27 CET] <rcombs> there's ALAC, which is uh& [21:39:34 CET] <wm4> oh ok lol [21:39:43 CET] <rcombs> I haven't really perf-tested it but it's a lossless codec and I don't really expect much differentiation there [21:39:44 CET] <wm4> I also see pcm alaw/ulaw [21:39:50 CET] <wm4> and whatever ilbc is? [21:39:58 CET] <RiCON> internet low bitrate codec [21:39:58 CET] <rcombs> and then PCM alaw/mulaw which are just there because they're available [21:40:01 CET] <rcombs> not really useful [21:40:04 CET] <RiCON> webrtc stuff [22:11:54 CET] <JEEB> rcombs: I thought they just licensed Adobe's? [22:12:00 CET] <JEEB> uhh [22:12:01 CET] <JEEB> not adobe [22:12:05 CET] <JEEB> dolby [22:13:22 CET] <rcombs> did they? [22:13:34 CET] <JEEB> that's what I heard [22:17:48 CET] <JEEB> seems to be a rumor/whatever from circa 2004, some really old audio FAQ notes that "The iTunes/apple codec is based on the original Dolby code and has also been tuned for quality significantly, despite the fact that it is CBR only the quality is quite good." [22:23:35 CET] <rcombs> JEEB: well it's not CBR-only anymore [22:23:51 CET] <rcombs> so either it's been modified quite a bit, or it's been replaced [22:24:15 CET] <RiCON> having everyone use itunes likely helped [22:25:03 CET] <JEEB> rcombs: it probably just got updated :) [22:26:02 CET] <JEEB> that line seems to now that I see even start with "Apple's AAC: CBR only with quicktime <7, quicktime 7 and later include VBR AAC." [22:26:34 CET] <JEEB> tl;dr possibly tweaked dolby's AAC [22:26:57 CET] <JEEB> in any case an encoder made (relatively) easily available [23:47:22 CET] <llogan> i wonder if there is a way to lock certain bug reports (such as the expired "art contest") [23:47:41 CET] <llogan> because that one seems to get more spam attachments [23:48:57 CET] <llogan> also, i immediately know it was spam because the username was female, which is kind of funny and sad at the same time. [23:49:17 CET] <atomnuker> michaelni: any way to reproduce the overflow on the aac encoder you reported? [23:53:42 CET] <michaelni> atomnuker, clang with ftrapv but you can just add a if() or av_assert() there to check for INT_MIN return [23:54:02 CET] <michaelni> atomnuker: http://fate.ffmpeg.org/report.cgi?time=20160323212826&slot=x86_64-freebsd10… [23:59:08 CET] <cehoyos> michaelni: Hi, I have moved two proposals to the "ignored" list, I hope this was ok. What about "Multi - Client Video Streaming over WLAN": Is there a chance that this will be an FFmpeg project? [23:59:53 CET] <cehoyos> atomnuker: Did you see that you have two candidates? Do they know that only one can be accepted? [00:00:00 CET] --- Thu Mar 24 2016

1 0

ffmpeg.log.20160323
by burek 24 Mar '16

24 Mar '16

[00:06:02 CET] <nownot> I have an avi file and when I try to convert I get Format avi detected only with low score of 1, misdetection possible! test.avi: Invalid data found when processing input .... thoughts on how to fix this? [00:10:24 CET] <J_Darnley> What's wrong with that? [00:10:38 CET] <J_Darnley> You said it was an avi file and ffmpeg thinks it is an avi file. [00:11:10 CET] <nownot> J_Darnley : it won't convert, I get that error when doing ffmpeg -i test.avi output.mp4 [00:11:23 CET] <livingBEEF> error: No output pad can be associated to link label 'v1'. [00:11:40 CET] <livingBEEF> Is there any way to get something more informative? [00:12:26 CET] <nownot> J_Darnley : https://gist.github.com/akatreyt/5fcba513707064090903 [00:13:44 CET] <J_Darnley> Try increasing -probesize, start with 20M and see if it helps [00:15:29 CET] <livingBEEF> http://sprunge.us/IiAZ [00:15:42 CET] <nownot> J_Darnley : didn't work :/ [00:16:09 CET] <livingBEEF> that's the command [00:18:09 CET] <J_Darnley> nownot: can anything play your avi file? [00:18:32 CET] <J_Darnley> livingBEEF: are you sure? there is no "v1" there [00:18:42 CET] <nownot> J_Darnley : no. was hoping that ffmpeg would be able to get the data out :/ [00:19:26 CET] <J_Darnley> If the file has been damaged in some way you might be out of luck. [00:19:45 CET] <J_Darnley> Where did it come from? [00:19:54 CET] <livingBEEF> Damn. I'm an idiot. I edited just the debug output [00:20:01 CET] <livingBEEF> (it's a bash script) [00:20:44 CET] <nownot> meh, thanks for the help [00:36:50 CET] <livingBEEF> I'm trying to cut off last frame, but I can't find anything... [00:36:53 CET] <livingBEEF> any ideas? [00:37:26 CET] <livingBEEF> trim only accepts absolute nubers, and it has no variables as far as I can tell [00:47:16 CET] <llogan> why do you want to do that? [00:47:57 CET] <livingBEEF> Timing. [00:48:18 CET] <livingBEEF> I'll probably just cut the frame from the start... [00:48:42 CET] <livingBEEF> works well enough [00:48:55 CET] <llogan> you could try counting the frames with ffprobe then using -frames:v in ffmpeg. probably won't work but worth a try. [00:49:35 CET] <llogan> ffprobe -v error -count_frames -select_streams v:0 -show_entries stream=nb_read_frames -of default=nokey=1:noprint_wrappers=1 input.foo [00:49:52 CET] <livingBEEF> The problem with that is that it's in the middle of filter_complex.... [00:50:22 CET] <llogan> that would have been good to know beforehand [00:50:44 CET] <llogan> one reason why we are so adamant about asking for the ffmpeg command and the complete console output [00:50:57 CET] <llogan> i was lazy, so therefore ended up wasting time. [00:51:01 CET] <livingBEEF> Yeah. [00:51:05 CET] <livingBEEF> Sorry about that [00:56:57 CET] <FiloSottile> I think I am stepping into a bug. I can't use the loop filter in a filtergraph [00:57:00 CET] <FiloSottile> http://pastebin.com/Sf30pamh [00:57:15 CET] <FiloSottile> [AVFilterGraph @ 0x7febdc000940] No such filter: 'loop' [00:57:43 CET] <FiloSottile> while, well, it's supposed to exist: https://ffmpeg.org/ffmpeg-filters.html#loop_002c-aloop [00:58:22 CET] <FiloSottile> (I know I can workaround this with -loop, but I want the loop to be applied AFTER the scale filter, or performance will be abysmal) [00:58:34 CET] <llogan> FiloSottile: your ffmpeg is too old. the filter is newer than the 3.0 branch. [00:59:07 CET] <llogan> general users are recommended to use a build from git master instead of releases which are mainly for distributors [00:59:24 CET] <FiloSottile> llogan: ah, that makes sense [00:59:47 CET] <FiloSottile> but 3.0 is the latest I'll find in any package, correct? [01:00:04 CET] <llogan> what do you mean? [01:00:42 CET] <FiloSottile> llogan: I meant that 3.0 is the latest "cut" version, right? I'm ok with using master, just checking [01:01:00 CET] <llogan> 3.0 is the latest release branch [01:01:17 CET] <llogan> you can download a build from http://www.evermeet.cx/ffmpeg/ [01:01:28 CET] <FiloSottile> so, before the introduction of the loop filter, how was one supposed to do a still background? [01:01:43 CET] <llogan> ...or use the --HEAD option in homebrew (I think. not an OS X user) [01:01:58 CET] <FiloSottile> (yeah, running --HEAD right now...) [01:02:32 CET] <llogan> you use the -loop image file demuxer input option [01:02:48 CET] <llogan> -loop 1 -i foo.jpg [01:03:08 CET] <llogan> since you're recompiling consider using libfdk-aac instead of libfaac [01:03:26 CET] <llogan> and remvoe enable-hardcoded-tables [01:03:46 CET] <FiloSottile> yeah, I got -loop working, but it causes the poor scale filter to take most of the CPU [01:05:10 CET] <FiloSottile> llogan: thanks for the suggestions, maybe you could check out https://github.com/Homebrew/homebrew/blob/master/Library/Formula/ffmpeg.rb and if you have improvements submit them there? [01:05:28 CET] <FiloSottile> it would improve a LOT of users' experience [01:05:37 CET] <FiloSottile> for example, --enable-hardcoded-tables is hardcoded [01:07:51 CET] <FiloSottile> (for fdk-aac instead I'm offered the option) [01:08:50 CET] <FiloSottile> why would I want to disable hardcoded-tables, just out of curiosity? [01:28:35 CET] <petecout_> Is there a way to specify which channel metadata gets encoded on? I can't find any documentation on that and Flash has an audio only channel for ID3 metadata [02:37:33 CET] <newbe-user> \help [02:38:57 CET] <newbe-user> Hi [02:41:47 CET] <J_Darnley> What help are you looking for? [03:22:57 CET] <mydarkside> hello! can someone please help me? I'm trying to capture the screen + Webcam + microphone + system audio , I can do ALL that the only problem I've is the webcam video is too forward ...any ideas how to match the timestamps or any other solution? thanks [03:58:54 CET] <newbe-user> Can I ask for help here? [04:02:46 CET] <newbe-user> this is my question -> http://stackoverflow.com/questions/36168994/how-can-i-combine-two-mp4-files [07:03:36 CET] <KuZon> hello [07:03:43 CET] <KuZon> i need some help regarding ffmpeg [07:04:26 CET] <KuZon> i want to access each frame of a hevc video and do some calculations on it [07:04:51 CET] <KuZon> i'm kinda stuck with it since 2 days please help me [07:06:14 CET] <relaxed> what have you tried? [07:09:46 CET] <KuZon> ffmpeg -i input -c:v libx265 -preset medium -x265-params crf=28 -strict experimental -r 50/1 output.mp4 [07:09:59 CET] <KuZon> i used this to extract frames of an 50fps video [07:10:22 CET] <KuZon> but the frame output is some kind of corrupted [07:10:39 CET] <TD-Linux> um what, that's reencoding, probably to h.264 [07:10:58 CET] <TD-Linux> maybe you want something like a PNG sequence? [07:11:08 CET] <relaxed> -c:v libx265 [07:11:26 CET] <furq> KuZon: it sounds like you want -c copy -frames:v 1 [07:14:12 CET] <relaxed> KuZon: describe exactly what you want [07:15:13 CET] <KuZon> sorry i have pasted wrng command [07:15:33 CET] <KuZon> ffmpeg -i input -c:v libx265 -preset medium -x265-params crf=28 -strict experimental -r 50/1 filename%03d.jpg [07:15:56 CET] <furq> yeah that's completely broken [07:16:06 CET] <furq> get rid of everything between input and filename%03d.jpg [07:16:17 CET] <KuZon> i want to extract frames of a hevc video and calculate psnr value of two consecutive frames [07:16:25 CET] <furq> that also isn't extracting frames, that'll reencode to jpeg [07:16:35 CET] <furq> if you want to calculate psnr difference then i guess png will work [07:16:51 CET] <furq> there's probably a better way though [07:18:04 CET] <KuZon> tell me other way [07:18:16 CET] <furq> if i knew it i wouldn't have said probably [07:18:31 CET] <KuZon> when i remove libx265 will it re encode it to x265 and give the result [07:18:37 CET] <KuZon> x264* [07:18:43 CET] <furq> ?? [07:18:50 CET] <KuZon> when i remove libx265 will it re encode it to x264 and give the result [07:18:54 CET] <furq> why would it do that [07:19:55 CET] <KuZon> because when i use ffmpegon a x265 video with out enabling libx265 say for changing framerate...it reencodes into x264 and gives the result [07:20:13 CET] <furq> it does that if the output format is mp4 because that's the default codec for mp4 if you don't specify [07:20:22 CET] <furq> it obviously doesn't do that if the output format is png [07:23:19 CET] <KuZon> ohh [07:24:00 CET] <furq> if you want to copy the input codec then use -c:v copy [07:24:34 CET] <furq> s/codec/stream/ [07:32:27 CET] <KuZon> thank you for the help [09:32:37 CET] <t4nk360> when I concatenate two files like this : "concat:file1.ts|file2.ts" file2.ts has no audio anymore (file1 never had one) , is there a way to prevent this ? [09:33:00 CET] Last message repeated 1 time(s). [09:33:40 CET] <t4nk360> when I instead use "concat:file2.ts|file1.ts" the audio of file2 is still there. file1 never had audio [09:50:46 CET] <t4nk360> it states my audio as am mp2 [09:56:35 CET] <wizonesolutions> Hi I've been trying the patch at https://trac.ffmpeg.org/ticket/4437#comment:21. What I am seeing is that things are better when recording and ffmpeg is the only thing recording, but if I try to record the screen when another application is also using the microphone, then the audio comes out poorly. Is it some sort of contention issue? It's Chrome that [09:56:36 CET] <wizonesolutions> is also using the microphone over WebRTC. [13:06:39 CET] <flux> well, is ffmpeg is recording only audio it cannot know if some other app is doing video capture :), so it certainly sounds like a cpu/scheduling issue [13:08:13 CET] <flux> wizonesolutions, perhaps increasing ffmpeg's priority (how to do it depends on the platform) or adjusting recording buffer sizes (depends on used capture device) would help [13:09:35 CET] <wizonesolutions> flux: I'm on OS X. Both applications are recording audio on the microphone. What happens it that the audio then comes out sped up and generally choppy on ffmpeg's side. [13:09:59 CET] <flux> "sped up" sounds like an ffmpeg issue [13:11:06 CET] <flux> (I suppose it could be an OS X issue as well.) [13:16:32 CET] <DHE> both apps slicing up the audio samples and splitting them between them? [13:16:44 CET] <DHE> like two apps trying to read from the same pipe [13:17:21 CET] <wizonesolutions> DHE: well, Chrome using the mic over WebRTC and then trying to record that on the screen + the microphone audio with ffmpeg, basically [13:17:41 CET] <wizonesolutions> for archival purposes basically of what's happening on the call [13:49:21 CET] <t4nk242> when I concatenate two files like this : "concat:file1.ts|file2.ts" file2.ts has no audio anymore (file1 never had one) , is there a way to prevent this ? when I instead use "concat:file2.ts|file1.ts" the audio of file2 is still there. file1 never had audio [14:16:17 CET] <Azrael_-> hi, i try this: ./ffmpeg -re -threads 3 -i ../programme/test.mp4 -c:v libx264 -b:v 300k -c:a libfdk_aac -ar 44100 -ac 1 -f mp4 rtmp://localhost:1935/myapp but always get the message: Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument i switched from "-f flv" to "-f mp4" according to some documentation. [14:16:24 CET] <Azrael_-> why do i get the error? [14:16:39 CET] <flux> wizonesolutions, can you verify that it works with some other audio capture program pair? [14:16:49 CET] <furq> Azrael_-: use flv for rtmp [14:17:12 CET] <furq> also which documentation told you to use mp4 because it's wrong [14:18:42 CET] <Azrael_-> furq: the mp4-documentation was some generic one telling me i should use mp4 container over flv for rtmp, so i tried to accomplish this with ffmpeg. found the "-f mp4"-switch here: https://trac.ffmpeg.org/wiki/Encode/H.264 [14:19:17 CET] <furq> well yeah rtmp only supports flv [14:19:34 CET] <Azrael_-> did you ever encounter stuttering with the live-stream? currently my video stream and sounds stops every few seconds for 1-2sec and then resumes [14:19:47 CET] <Azrael_-> ok, good to know the other doc wasn't right [14:20:27 CET] <furq> is the encoding going at the same speed as the input file [14:21:53 CET] <Azrael_-> i started it with "-re" and it is running with 0.996x [14:22:10 CET] <furq> no idea then [14:22:50 CET] <Azrael_-> so many components working together which could be the problem and i have no clue how dig into it further [14:39:03 CET] <nadavrub> I am trying to build ffmpeg using MinGW64: configure --enable-shared --enable-libx264 --enable-libfaac --enable-nonfree --enable-gpl --enable-zlib --enable-memalign-hack --disable-static --extra-ldflags="-L/usr/local/lib" --extra-cflags="-I/usr/local/include", the resulting "avcodec-57.def" include libfaac exports rather than the expected av_*.* exports, might this B a problem w/ the latest ffmpeg @ github ?? [14:39:10 CET] <nadavrub> EXPORTS faacEncClose faacEncEncode faacEncGetCurrentConfiguration faacEncGetDecoderSpecificInfo faacEncGetVersion faacEncOpen faacEncSetConfiguration [14:41:40 CET] <atomnuker> why would you want faac? it's literally the worst aac encoder since a few years [14:44:44 CET] <nadavrub> @atomnuker -> What is the alternative for AAC encoding ? [14:46:03 CET] <atomnuker> the ffmpeg native aac encoder [14:46:26 CET] <nadavrub> Should I use any specific ./configure switch for that ? [14:46:51 CET] <atomnuker> nope, it's already built-in [14:48:52 CET] <nadavrub> Ok, TnX, will try [14:56:18 CET] <nadavrub> Yup, ditching libfaac makes ffmpeg compile, [14:56:24 CET] <nadavrub> TnX [15:01:18 CET] <ac_slater> hey guys. I've had a problem for a while and it's been annoying me. When I stream v4l via udp, the stream gradually (but quickly) falls behind. Meaning, there is a 0-20second delay overtime. I'm using ffplay/vlc/mpv as the players. I've narrowed out ffmpeg's udp as the issue by replacing it with netcat. Clues? Is my encoding too slow? ffmpeg is reporting > 1.00x encoding speed [15:02:24 CET] <ac_slater> my command line looks a little something like `ffmpeg -r 15 -f video4linux2 -video_size 640x480 -framerate 15 -f /dev/video0 -f mpegts udp://127.0.0.1?pkt_size=752` [15:11:06 CET] <t4nk242> maybe you could use the -re command [15:16:18 CET] <ac_slater> t4nk242: good point. I'll try that again really quick. I assume you mean on input [15:17:20 CET] <ac_slater> t4nk242: when I do that, I get "Past Duration 0.xxxxx too large" ... I think it's from the mpegts muxer [15:17:49 CET] <ac_slater> I should mention that the encoder is a custom h264 encoder, I forgot to include it in the command line. [15:18:10 CET] <ac_slater> but, all of these same things happen with x264. [15:18:30 CET] <ac_slater> t4nk242: (-re didnt help the latency) [15:19:00 CET] <t4nk242> ok, sorry tha nI don't know, I am still a beginner ^^ [15:19:16 CET] <ac_slater> thanks anyways [15:50:59 CET] <t4nk242> when I concatenate two files like this : "concat:file1.ts|file2.ts" file2.ts has no audio anymore (file1 never had one) , is there a way to prevent this ? when I instead use "concat:file2.ts|file1.ts" the audio of file2 is still there. file1 never had audio [15:53:15 CET] <J_Darnley> Oh actually, I might understand [15:53:37 CET] <J_Darnley> Increase probesize until ffmpeg sees the audio in the second file. [16:05:38 CET] <t4nk242> I'll try J_Darnley , ty :) , anyways, here is the pastebin : http://pastebin.com/L0kC5zRj [16:06:58 CET] <t4nk242> how to increase the probesize ? [16:08:34 CET] <relaxed> t4nk242: if that doesn't work just add a silent audio stream to the first file that matches the second. [16:14:14 CET] <t4nk242> @relaxed I considerd this, but it does not seem optimal to me ^^ [16:16:06 CET] <t4nk242> how to whisper ? [16:21:01 CET] <gnome1> hello all, I've been playing a bit with video transcoding lately, mostly trying to get video files smaller with libx264. out of curiosity, I tried to compare with HandBrakeCLI and the closest settings I got still give me a ~180 MiB bigger file, I wonder if this means I overlooked some setting I should have used... does someone know which ffmpeg options would correspond to some of the handbrake builtin [16:21:07 CET] <gnome1> profiles? [16:22:51 CET] <t4nk242> probesize increase didn't help :/ [16:27:32 CET] <durandal_170> for how much was it increased? [16:27:49 CET] <wizonesolutions> flux: you mean e.g. record with QuickTime at the same time as Chrome? ok [16:29:45 CET] <shincodex> why does ext4 not defautly install as nodelayalloc [16:30:57 CET] <t4nk242> I tried it with 100, 1000 and 10000 [16:36:57 CET] <J_Darnley> Wow. One hundred bytes. One thousand bytes. Ten thousand bytes. [16:37:07 CET] <J_Darnley> I think the default is five million bytes. [16:39:52 CET] <petecout_> Is it possible to injected Timed ID3 metadata into an HLS output? [16:40:12 CET] <petecout_> Using ffmpeg [16:40:15 CET] <Filarius> how to pass to ffmpeg same video several times, but splited by time, like 4 parts of same length, but without actually spliting original ? [16:41:00 CET] <J_Darnley> what? [16:41:17 CET] <petecout_> ^ [16:41:27 CET] <J_Darnley> Perhaps you should describe what you are trying to achieve [16:41:38 CET] <J_Darnley> Then we can tell you how to do it. [16:42:16 CET] <Filarius> me or petecount_, or both ? [16:42:22 CET] <J_Darnley> you. [16:42:31 CET] <J_Darnley> I remember petecout_ from the other day [16:42:41 CET] <J_Darnley> I still don't know how to solve his problem. [16:44:48 CET] <Filarius> okay, I have many small videos of same resolution, I'm looking for simple way to place all videos on one, but grid-like. Like 3x3 grid playing videos at same time and grid cells switching to new video if previous is ended. [16:46:06 CET] <Filarius> I already know how to make grid-video, but from separate files from start to end [16:47:32 CET] <Filarius> so I thought is it possible to send to ffmpeg same file several times, but like say to ffmpeg to take from 0 to 1/4 of total time from first, from 1/4 to 2/4 from second input [16:47:36 CET] <Filarius> and etc [16:48:13 CET] <Filarius> and concatinate all small files to one long [16:48:18 CET] <Filarius> before this [16:48:52 CET] <J_Darnley> Sounds overly ambitious for the horror that is ffmpeg's filtering [16:50:28 CET] <wizonesolutions> flux: one interesting thing. FFmpeg is often OK for roughly 10-15 seconds before it starts getting particularly noticeable. It isn't consistent. [16:50:37 CET] <wizonesolutions> flux: btw, yeah, QuickTime Player can record just fine. [16:50:48 CET] <wizonesolutions> no distortion, audio and video are in sync [16:50:55 CET] <Filarius> so, it is possible to say to ffmpeg to take only 1/4 of total time from some input, by setting start and end time not in seconds, but like size relative to total time [16:51:57 CET] <Filarius> like 0.2 or 0.5 [16:52:05 CET] <J_Darnley> no [16:52:25 CET] <J_Darnley> you can only instruct ffmpeg to seek based on absolute time. [16:54:49 CET] <Filarius> okay [16:58:07 CET] <petecout_> J_Darnley: Lol no one knows the answer to my problem. It's something that is just starting to pick up speed. [17:30:06 CET] <PlanC> is it possible to reduce the bitrate of an MP3 without having to re-encode it? [17:30:43 CET] <JEEB> no [17:31:02 CET] <PlanC> I have a few hundred MP3s from my lectures and some have bizarre bitrates, but it takes such a time to reduce them to something reasonable like 128 kbps [17:31:11 CET] <PlanC> JEEB: that's a shame [17:31:23 CET] <gnome1> is there some way to specify width:heigth but keep the aspect ratio? (that is, specify maximum frame width and maximum frame height)? [17:32:32 CET] <JEEB> PlanC: not sure how you'd expect to be able to do anything but cutting them without recompression :) [17:32:43 CET] <PlanC> JEEB: yeah, you're right [17:32:58 CET] <PlanC> do you know of any tricks that'll speed the process up at least? [17:33:20 CET] <gnome1> or maybe I should just specify the width (w:-1), should work unless it's not landscape [17:34:14 CET] <Demp> ffmpeg hangs after about 3000 frames when using this command: "ffmpeg -i "http://62.90.90.56/iba_vod/_definst_/smil:iba-xDsEA1-sRRo.smil/playlist.m3u…" -c copy -bsf:a aac_adtstoasc bla.mkv". any ideas? [17:34:28 CET] <PlanC> gnome1: why not just get the width and height with ffprobe first [17:34:32 CET] <PlanC> calculate the aspect ratio [17:34:48 CET] <J_Darnley> Let ffmpeg do that for you. [17:34:49 CET] <PlanC> calculate the new, smaller width and height based on the aspect ratio? [17:35:10 CET] <J_Darnley> Read the other examples of scale expressions [17:35:33 CET] <gnome1> PlanC: I was looking for a simpler command line that'd not have to be tweaked for every single file [17:36:24 CET] <PlanC> gnome1: I guess that's handy if you're entering it straight into the shell [17:36:39 CET] <gnome1> how else, that's how I have always used ffmpeg [17:37:07 CET] <PlanC> BASH, Python, PHP, etc [17:37:30 CET] <gnome1> bash's a shell... [17:38:26 CET] <PlanC> no kidding [17:38:38 CET] <PlanC> what I'm trying to say is that you could make a bash script [17:38:55 CET] <PlanC> but there seems to be a feature for this so you might not need it [17:38:56 CET] <PlanC> https://ffmpeg.org/ffmpeg-filters.html#Options-1 [17:39:08 CET] <J_Darnley> Or you could read the fucking manual! [17:39:16 CET] <J_Darnley> There it is ^ [17:41:40 CET] <PlanC> works in this case, but sometimes the whole process doesn't fit in one line [17:42:12 CET] <gnome1> I'm not trying to go that far, what I'm looking exactly is to use ffmpeg directly. and I did mention -1. nevermind that I asked. [17:42:49 CET] <J_Darnley> If -1 works for you then what's the problem? [17:43:55 CET] <PlanC> yeah, with -1 and force_original_aspect_ratio what else is there to ask for? [17:51:24 CET] <magento_rocks> can i add metadata to an audio file with ffmpeg? [17:52:13 CET] <magento_rocks> i have a .flac file that i converted to .wav, then converted to .m4a (ALAC) but the metadata is lost in the conversion process, i am trying to add it back to the .m4a file [17:52:35 CET] <J_Darnley> -metadata [17:52:57 CET] <J_Darnley> or if you still have the original file, be all fancy and use -map_metadata [17:53:47 CET] <Timeless_> Is it possible to restream a live MP4v2 stream to another format with FFmpeg? [17:54:12 CET] <Timeless_> Currently when I try to restream it waits for the stream to end and then converts the file. [17:54:13 CET] <J_Darnley> probably [17:54:28 CET] <Timeless_> I would like to do it on the fly [17:55:51 CET] <Timeless_> ffmpeg -i http://addres:port/test.mp4 -f m4v -r 2 -vcodec mpeg4 test.mp4 [17:56:16 CET] <Timeless_> This waits for the input stream to end and then conv the file. [17:56:27 CET] <Timeless_> But I like to have it on the fly. Am I missing something here? [17:56:36 CET] <J_Darnley> Yes. [17:56:51 CET] <J_Darnley> You cannot use an mp4 file until it is finished. [17:56:59 CET] <magento_rocks> thanks J_Darnley ill try that [17:57:15 CET] <Timeless_> So there is no way of live streaming it? [17:57:25 CET] <Timeless_> sorry"live input" [17:57:51 CET] <Timeless_> Or do you mean that my output cannot be mp4? [17:57:51 CET] <gnome1> if it cannot be used until it is finished, how is it suitable for streaming? it isn't? [17:58:09 CET] <gnome1> oh, that's plain http. [17:58:50 CET] <Timeless_> The problem is that my input can only be MP4v2 because the source is proprietary [18:01:34 CET] <JEEB> Timeless_: if whatever is generating your input can use the "fragments" feature then you will be able to start quicker, but without it the index will be at the end [18:01:40 CET] <JEEB> and without an index the file can't be read. at all [18:02:11 CET] <kepstin> well, it's not really the index, it's just that there's some codec init stuff stuck in with the index [18:02:19 CET] <kepstin> so it can't write that until it has the index ready :/ [18:03:01 CET] <JEEB> yes, although the result is the same - no index, not usable [18:04:12 CET] <Timeless_> I have 3 options. seperate frames(aka jpeg) Payload and Raw [18:04:28 CET] <Timeless_> the last 2 are detected as m4v by vlc and ffmpeg [18:04:40 CET] <Timeless_> but this could be just a default [18:05:20 CET] <Timeless_> But if I append the MOOV ATOM data to the m4v stream. would it them be possible use it as live input? [18:05:29 CET] <Timeless_> then* [18:05:45 CET] <JEEB> m4v is probably just some raw video stuff, that would be readable right away [18:06:02 CET] <JEEB> wouldn't have audio of course, but not sure if you have that anyways [18:06:06 CET] <JEEB> also wouldn't have timing information [18:06:07 CET] <kepstin> 'm4v' is typically just an alternate extension for 'mp4', and is the normally the same format [18:06:30 CET] <JEEB> "detected by ffmpeg as 'm4v'" usually means MPEG-4 Part 2 raw bitstream :P [18:06:48 CET] <Timeless_> JEEB: yeah thats correct. [18:06:56 CET] <JEEB> DE m4v raw MPEG-4 video [18:07:05 CET] <kepstin> hmm, I'm getting that confused, then :/ [18:07:18 CET] <JEEB> the extension is used by apple for ISOBMFF with video though [18:07:33 CET] <kepstin> oh, m4a is just mp4 without a video stream, i thought they did the same thing with video [18:07:38 CET] <Timeless_> I tought that m4v is mp4(without audio) [18:07:53 CET] <JEEB> Apple uses the *file* *extensions* m4a and m4v [18:08:08 CET] <JEEB> which as I said are ISOBMFF with either only audio or video (and possibly audio) [18:08:49 CET] <JEEB> ISOBMFF = ISO Base Media File Format aka what you usually call "mp4" :P [18:08:56 CET] <Timeless_> But because of the "index" (meta data?) it is not possible to use it as live input? [18:09:13 CET] <JEEB> the raw bit stream you can as-is, since it's just a raw bit stream [18:09:32 CET] <JEEB> the container'ized thing requires the index with the init stuff etc [18:09:43 CET] <JEEB> unless it can do fragments, which I guess it doesn't :P [18:09:53 CET] <Timeless_> No it cannot do fragments :( [18:10:09 CET] <Timeless_> Normally this source is used to write video to a file [18:10:22 CET] <Timeless_> which I now pipe to a http stream [18:10:27 CET] <kepstin> Timeless_: if correctly made, an mp4 (iso) file can have the 'moov' moved to the start of the file so it can be used as a live input. The issue is that you can't use an mp4 as a live *output*. (i.e. you can't read one as it's being written) [18:11:05 CET] <Timeless_> sorry. it seems that my question is unclear [18:11:08 CET] <Timeless_> what I ment is: [18:11:41 CET] <Timeless_> m4v (mp4 - raw) as input ---> whatever coded(container) is capable of live stream as output [18:11:51 CET] <Timeless_> coded = codec* [18:12:07 CET] <JEEB> yes, you just lose the timestamps and any audio if there is any [18:12:11 CET] <JEEB> since that's not in the raw bit stream [18:12:23 CET] <JEEB> but yes, raw bit stream is supposed to be readable and decode'able A=> [18:12:27 CET] <JEEB> *A=>B [18:13:05 CET] <Timeless_> ok great [18:13:10 CET] <kepstin> Timeless_: you can a couple of options there for live streamable output; e.g. mkv is streamable, you could use mpeg-ts, you could use fragmented output like dash or hls. [18:13:27 CET] <JEEB> kepstin: he has a closed thing that outputs either raw MPEG-4 Part 2 or MPEG-4 Part 2 in mp4 [18:13:30 CET] <JEEB> or MJPEG [18:13:46 CET] <Timeless_> yeah thats correct JEEB [18:13:50 CET] <JEEB> and he is asking if any of those is something he can push through HTTP to achieve live streaming :P [18:14:25 CET] <Timeless_> the output is not my problem right now. that isn't difficult :) [18:14:48 CET] <kepstin> hmm, so i'm looking at the wrong end of the problem, then? [18:14:54 CET] <Timeless_> JEEB: so if the input is indeed RAW I only have to set some option during the CLI action and that should make it work? [18:15:08 CET] <Timeless_> kepstin: yup :) [18:15:27 CET] <Timeless_> JEEB: so options like framesize bitrate fps ? [18:17:49 CET] <JEEB> Timeless_: the raw bit stream should have that info, it's not raw video as such [18:17:53 CET] <JEEB> it's raw compressed bit stream [18:18:06 CET] <JEEB> so it should have the init info at every IDR picture, hopefully [18:18:21 CET] <JEEB> if not, you're fucked unless you start from the beginning of the stream :P [18:24:21 CET] <Timeless_> JEEB: ok great. I will give that a try. [18:24:38 CET] <Timeless_> Currently I first start the pipe --> http and then ffmpeg [18:24:50 CET] <Timeless_> so maybe Im missing the initial init info [18:25:07 CET] <Timeless_> I have to go now. thanks for your help people :) [19:07:38 CET] Action: Keshl facedesks a few thousand times. "So, I have an image sequence starting with, say, "Frame_0001.bmp", except it's 25,216 frames long. The program that made these files kept a consistent numbering scheme up until frame 9,999, and then added an extra digit to get to 10,000. Is there any "easy" way to handle throwing this at ffmpeg?" [19:22:45 CET] <gnome1> wonderful. [19:32:15 CET] <Keshl> Whoo! Got it! Easier than I thought! [19:32:59 CET] <Keshl> ffmpeg -i "Frame_%04d.bmp" -i "Frame_1%04d.bmp" -i "Frame_2%04d.bmp" -i "Sound.wav" -vcodec (Etc) [19:34:50 CET] <Keshl> Oh, duh. Don't forget -r 144 (Or whichever your framerate's at.) [19:38:37 CET] <gnome1> if that's a one-time problem and the machine has "rename" installed, it may be worth to "rename Frame_ Frame_0 Frame_????.bmp". ffmpeg won't be the only tool to have a trouble with that kind of numbering [20:47:45 CET] <bubbely> question: im trying to write an app to stream my Spotify to my symbian nokia 808 phone. is something like that possible with ffmpeg [20:48:10 CET] <bubbely> maybe a webserver with the current song im listening to being streamed and parsed by ffmpeg on my phone with a qt app i write [20:48:36 CET] <J_Darnley> ffmpeg is a tool for turning one sort of format into another. [20:49:00 CET] <J_Darnley> if you don't need to do that it is probably overkill [20:50:54 CET] <bubbely> the webpage says it can "stream" [20:50:55 CET] <bubbely> whats that about [20:51:18 CET] <J_Darnley> It can output a contious stream [20:51:31 CET] <bubbely> what can i use to listen to it [20:51:41 CET] <J_Darnley> that can be watched/listen lived [20:51:44 CET] <J_Darnley> *live [20:51:52 CET] <bubbely> J_Darnley: is there an example of how to set up the server and a client ? [20:52:21 CET] <bubbely> well.. how does someone listen live ? [20:52:23 CET] <J_Darnley> there must be hundreds of examples of streaming using ffmpeg [20:52:59 CET] <J_Darnley> Not that I know any [20:53:07 CET] <J_Darnley> I think streaming is a cancer [20:53:48 CET] <bubbely> hey, J_Darnley, how does one listen to the stream? [20:54:00 CET] <J_Darnley> open it in any good media player [20:54:04 CET] <gnome1> I'd say there's one use for streaming, and that's live content. For everything else, streaming is security through obscurity or a misunderstanding of how things work. [20:54:07 CET] <J_Darnley> (I assume) [20:54:32 CET] <bubbely> I need a Qt example of someone streaming from a FFMPEG server [20:54:36 CET] <bubbely> any idea gnome1 ? [20:54:45 CET] <J_Darnley> I don't even think spotify will let you easily get music out of it anyway. [20:54:52 CET] <gnome1> no, I don't have experience with Qt [20:55:31 CET] <andrey_utkin> bubbely: elaborate plz [20:55:59 CET] <bubbely> andrey_utkin: I want to create an FFMPEG server streaming my PC playback device [20:56:11 CET] <bubbely> andredy_utkin: and i want to listen to it from a Qt app on my symbian device [20:56:21 CET] <bubbely> andrey_utkin* [20:56:23 CET] <bubbely> makes sense? [20:57:29 CET] <andrey_utkin> Streaming your pc input device you mean? [20:57:56 CET] <bubbely> streaming my pc playback, the music playing on my computer [20:59:19 CET] <bubbely> andrey_utkin: it seems i need to find a way to play .asf files on my phone [20:59:36 CET] <kepstin> bubbely: you might be better off just getting the 'libspotify' sdk and building a native spotify player for your phone. [21:00:10 CET] <andrey_utkin> Getting data of what goes out of your earphones is separate task, i d look at pulseaudio, jack or alsa stuff [21:00:22 CET] <bubbely> kepstin: I had that idea [21:00:22 CET] <gnome1> there's probably some alsa setting to do that [21:00:29 CET] <bubbely> kepstin: but i figured streaming would be faster [21:00:38 CET] <kepstin> ffmpeg can use the pulseaudio playback device as a capture source, that's not hard to do [21:00:41 CET] <gnome1> I recall seeing examples of config files where the "pcm" output was available (by software) as an input for recording [21:00:46 CET] <gnome1> well, for capturing [21:00:53 CET] <gnome1> some cards have this on hardware, some don't [21:01:10 CET] <kepstin> if you're using alsa without pulseaudio, you'd need either a card with a 'what you hear' hardware recording device, or use the 'aloop' module to make a virtual soundcard. [21:01:16 CET] <kepstin> pulseaudio makes it easier :) [21:01:27 CET] <gnome1> yeah, I believe it's aloop [21:01:42 CET] <bubbely> kepstin: do u have a Qt example of playing a .asf stream [21:02:39 CET] <kepstin> bubbely: not a qt example. you'd want to check if there's any os apis for it (although I doubt it for asf), or you could always use ffmpeg to decode it... [21:03:05 CET] <bubbely> kepstin: qt example of any form of ffmpeg ? [21:03:24 CET] <andrey_utkin> bubbely: see above, guys figured you can get raw audio, personally no idea what asf is [21:03:46 CET] <J_Darnley> Shit from microsoft [21:04:08 CET] <andrey_utkin> bubbely: ffmpeg api i just c, no big sense to ask for qt specific examples [21:04:22 CET] <bubbely> andrey_utkin: Ok [21:04:23 CET] <kepstin> they have apis for getting the audio if you have an api key, i don't think it's actually that hard. [21:04:33 CET] <bubbely> kepstin: spotify ? [21:04:39 CET] <kepstin> they used to even have a native symbian app for spotify, but that's gone now [21:04:49 CET] <andrey_utkin> It is like jquery for math in js, not needed :-) [21:04:54 CET] <kepstin> https://news.spotify.com/us/2009/11/23/spotify-for-nokia-and-more/ is just a bunch of dead links :) [21:07:08 CET] <andrey_utkin> bubbely: you could serve at last single connection by rtsp or http protocol from ffmpeg command line. Should be enough to start streaming to your phone [21:07:29 CET] <andrey_utkin> Is vlc available on your phone? [21:08:02 CET] <andrey_utkin> Or any rtsp or http player [21:10:41 CET] <bubbely> i have DLNA play [22:09:51 CET] <AlexQ> Why did ffmpeg encode 16-bit WAV to 24-bit FLAC when remuxing to MKV? At least that's what mediainfo says. I have playback issues. [22:10:09 CET] <AlexQ> VLC says it's 32 bit FLAC xD [22:12:33 CET] <AlexQ> That's really strange. How do I force bit depth? [22:12:40 CET] <J_Darnley> If you told it to copy then it copied. [22:15:01 CET] <AlexQ> I did exactly the same thing on the other file before with no problems. Maybe that's because the original DTS track from the video-source MKV was 24-bit? Somehow ffmpeg got confused? [22:15:53 CET] <AlexQ> BTW. There is that sofalizer plugin - is it available in official ffmpeg build or not? And how do I decide which profile file fits me best, I'd need to listen to all of them? [22:22:43 CET] <kepstin> AlexQ: the output audio bit depth that ffmpeg uses is automatically selected from the input and filters used, so in some cases it might do a higher depth flac depending on the input format. You can use the -sample_fmt option to force a specific output format (it might cause an extra conversion filter to be added) [22:23:16 CET] <AlexQ> kepstin: Might that have been caused by the "volume" filter? [22:24:07 CET] <kepstin> could be; I think the 'volume' filter only works in floating point, so you get 32bit float by default [22:24:27 CET] <kepstin> (and it picks the flac format that loses the fewest bits of precision) [22:30:24 CET] <JEEB> the audio/video formats IIRC are only decided by a list for a container ;P [22:30:57 CET] <JEEB> in other words, you really don't want to not define a "codec" for your streams [22:31:18 CET] <JEEB> if you don't define it and don't set a global copy - you get what you asked for, which is russian roulette [22:31:47 CET] <AlexQ> JEEB: I defined codec as "FLAC" but with no sample rate. Fair enough, that's couse of the filter [22:32:25 CET] <JEEB> kepstin was just commenting as if the sample format would affect the "codec" decision [22:32:35 CET] <JEEB> which as far as I know it doesn't [22:32:38 CET] <AlexQ> So how should I use volume and FLAC output codec so that I achieve 16-bit FLAC? My input is going to be a 32-bit WAV [22:33:15 CET] <JEEB> use a resample filter at the end of your filter chain [22:33:31 CET] <Azrael_-> hi [22:33:38 CET] <AlexQ> and sample_fmt option would be risker than that? [22:33:59 CET] <Azrael_-> does anybody use ffmpeg to stream to wowza and can tell me how to send the proper source/password for the rtmp-publishing? [22:34:02 CET] <JEEB> that most probably does the resampling logic behind the curtain [22:34:22 CET] <JEEB> as in, "if input isn't this, add resample filter to the end of the filer chain" [22:36:06 CET] <JEEB> *filter [22:36:39 CET] <AlexQ> JEEB: Though to make sure the resampling happens on the end of the filter chain it would be best to use resample filter manually then, yeah? [22:37:06 CET] <JEEB> well it would happen in the end anyways [22:37:19 CET] <JEEB> so in this case it most likely doesn't matter [22:40:59 CET] <AlexQ> JEEB: What's the difference between s16 and s16p? Both are 16-bit [22:41:20 CET] <JEEB> how the data is handled [22:41:25 CET] <J_Darnley> not-planar and planar [22:41:41 CET] <JEEB> p is planar, as in first you get the samples for one channel and then the other and so forth [22:42:05 CET] <JEEB> the other is packed, which means sample/sample/sample from each of the channels one after another [22:43:27 CET] <AlexQ> So which should I choose for a regular FLAC stream? [22:43:49 CET] <J_Darnley> it doesn't matter but the flac encoder only accept packed [22:45:29 CET] <AlexQ> So -sample_fmt s16 [22:46:26 CET] <J_Darnley> yes [22:47:50 CET] <AlexQ> When I do: ffmpeg -i pcm_f32le.wav -af "volumedetect" -f null /dev/null I do get Stream mapping: Stream #0:0 -> #0:0 (pcm_f32le (native) -> pcm_s16le (native)) [22:48:06 CET] <AlexQ> But I do hope that the analysys happens before conversion to s16le, doesn't it? [22:48:26 CET] <J_Darnley> almost certainly [22:48:51 CET] <J_Darnley> you can add -loglevel verbose or debug and it should show you where the resample is added [22:49:46 CET] <AlexQ> What about that sofalizer filter? It isn't available in standard ffmpeg distributions? [22:50:45 CET] <J_Darnley> I don't know [22:50:51 CET] <J_Darnley> check how it was built [22:51:03 CET] <durandal_1707> you need netcdf [22:52:35 CET] <durandal_1707> its for downmixing surround to binaural stereo [22:54:40 CET] <AlexQ> So which one is better, durandal_1707? [22:56:02 CET] <durandal_1707> AlexQ: which is another? [22:56:14 CET] <AlexQ> sofalizer [22:56:46 CET] <durandal_1707> and another? [22:57:18 CET] <AlexQ> what is that NetCDF? Is it a filter or what? [22:57:23 CET] <AlexQ> There is also e.g. Dolby Headphone [22:57:44 CET] <AlexQ> there is some filter from VLC, though I don't know what that is [22:58:06 CET] <durandal_1707> Netcdf is library you need to have sofalizer filter [22:58:41 CET] <AlexQ> you can use Dolby Headphone with Foobar [22:59:14 CET] <AlexQ> to convert files [22:59:38 CET] <durandal_1707> with headphone? [22:59:57 CET] <AlexQ> ? [23:00:58 CET] <durandal_1707> how you can convert files with foobar and Dolby headphone? [23:01:14 CET] <AlexQ> there is a Dolby Headphone DSP plugin for Foobar [23:01:19 CET] <AlexQ> you need to have the DLL [23:01:36 CET] <AlexQ> and Foobar supports audio conversion with DSP chains [23:01:47 CET] <durandal_1707> ah [23:02:32 CET] <durandal_1707> for sofalizer you can pick various files to suit you taste [23:02:54 CET] <AlexQ> though technologies using HRTF for in-ear monitors are probably better I guess? Yes, I think that's cool [23:02:58 CET] <gnome1> when scaling down video, does the method matter that much, should the default be good enough? [23:03:39 CET] <AlexQ> Okay, I've got to go. Thanks and bye! [23:04:03 CET] <AlexQ> durandal_1707: So basically I need to get that library, enable sofalizer in ffmpeg build flags and then recompile ffmpeg? [23:04:09 CET] <AlexQ> to play around with sofalizer? [23:04:24 CET] <durandal_1707> yes [23:04:42 CET] <J_Darnley> gnome1: if you use bicubic or better it makes little difference. [23:04:57 CET] <AlexQ> and I should be able to check these presetes without the need to encode to files with ffplay? [23:06:01 CET] <durandal_1707> you need just ffplay to use audio filter, no need to encode, it should be pretty fast [23:06:25 CET] <gnome1> J_Darnley: bicubic is the default, right? [23:06:33 CET] <J_Darnley> no idea [23:06:39 CET] <AlexQ> Or even better, is there any GUI player that allows the playback of surround files and supports sofalizer output filters? To check which profile provides best effect. I should try these hrtf_b profiles I guess? [23:06:52 CET] <J_Darnley> I might expect bilinear given the age of swscale [23:07:39 CET] <durandal_1707> AlexQ: there are many sofa files, each with pros and cons [23:08:21 CET] <durandal_1707> There's no GUI afaik, only patch for vlc [23:13:22 CET] <AlexQ> durandal_1707: Well, for the best effect I would need to have a HRTF for my own body I believe [23:14:17 CET] <durandal_1707> there are pretty good generic ones [23:15:04 CET] <AlexQ> do these profiles/sofalizer itself use LFE as well? [23:15:32 CET] <AlexQ> Dolby Headphone ignores it, which is stupid as good headphones are able to transfer these freqs, or at least part of them [23:19:20 CET] <AlexQ> Egrrrh, that problem had nothing to do with the FLAC being 24-bit or whatever. It is just a strange sync issue [23:19:23 CET] <durandal_1707> yes, lfe is passed as mono [23:19:59 CET] <AlexQ> the video starts with no audio [23:20:25 CET] <AlexQ> then the video freezes, audio begins to play up to the moment when it synces with video and the video starts to play as well [23:20:37 CET] <AlexQ> and then after a minute or so the audio is gone. What's up? [23:21:28 CET] <durandal_1707> sample? [23:21:38 CET] <AlexQ> Where should I upload? [23:22:39 CET] <gnome1> if this weren't #ffmpeg, I'd guess you used mencoder somewhere [23:23:15 CET] Action: gnome1 was never able to get a video with decent a-v sync in mencoder [23:23:33 CET] <AlexQ> The original MKV file has chapters or whatever. Maybe they cause the issue. How can I ignore chapters when remuxing? [23:26:55 CET] <i10k> Hey everyone. I've stumbled upon a little problem with video made with ffmpeg and I hope you can help. [23:26:58 CET] <durandal_1707> ordered chapters? [23:27:02 CET] <i10k> I have two flv files. One of them has just the video track, the other one has both audio and video. As a result I'd like to have a file that has the video from the former and audio from the latter. But there's a caveat. [23:27:05 CET] <i10k> Video is 55 seconds shorter than the audio, so I tried using -itsoffset to make the video start after first 55 seconds. [23:27:09 CET] <i10k> But when I open the resulting file in VLC I only hear audio with no video and it throws an error "VLC does not support the audio or video format "undf"" [23:27:13 CET] <i10k> Here's the command I'm using http://pastebin.com/f7apAJDh [23:27:17 CET] <i10k> While I was playing with arguments I noticed that file becomes unreadable by players when -itsoffset is bigger then 5 seconds. Am I doing something wrong? Thanks [23:27:53 CET] <BtbN> players probably just probe the first 5 seconds, and if they don't find a video stream then, they decide that it's audio-only [23:29:26 CET] <AlexQ> i10k: You could try -ss to seek one of the streams instead of the -itsoffset maybe? I used it once to resync a/v [23:29:31 CET] <AlexQ> without transcoding [23:29:32 CET] <i10k> I thought that ffmpeg stretches the video track by filling the "skipped" part with the first frame when using -itsoffset. [23:30:35 CET] <AlexQ> Maybe, dunno [23:31:15 CET] <AlexQ> durandal_1707: Umm, just some chapters or whatever. Can I ignore them or sth? Really wanted to have properly downmixed audio to play on my mobile device using headphones [23:32:02 CET] <durandal_1707> dunno, without sample I can't tell [23:32:39 CET] <AlexQ> So where should I upload one? [23:32:58 CET] <AlexQ> I tried null decode and there is a lot of errors with -loglevel debug [23:33:20 CET] <AlexQ> I mean, there is something about DTS, while the DTS stream is gone and replaced with the FLAC stream [23:33:40 CET] <AlexQ> but that only appears with -loglevel debug [23:36:20 CET] <durandal_1707> try remixing it with codec copy [23:36:29 CET] <durandal_1707> *remuxing [23:36:49 CET] <AlexQ> You mean, remuxing it again? Without any audio filters this time? [23:37:27 CET] <AlexQ> Maybe I'll try to remux it with mkvmerge? [23:39:04 CET] <AlexQ> extracted first 3 mins of FLAC and it played well [23:40:02 CET] <AlexQ> if I do extract h264 and FLAC and then mux these two, all other info should be gone I presume? Chapters etc., even sync info [23:42:40 CET] <durandal_1707> does sync issues happen only when using filters? [23:43:26 CET] <AlexQ> Um, haven't tried without [23:44:07 CET] <AlexQ> I'm extracting FLAC and h.264 now... [23:48:40 CET] <AlexQ> mkvmerge told me my h264 file has unknown type. To what filetype should I extract then? [23:49:36 CET] <durandal_1707> can't you use ffmpeg to remux? [23:49:37 CET] <AlexQ> I'll check if ffmpeg will be able to mux these two.. [23:50:01 CET] <AlexQ> just wanted to try something else to increase the chances of everything going well [23:50:43 CET] <AlexQ> do I need to use map when I use vid-only file first and audio-only file as second when muxing using ffmpeg? [23:52:41 CET] <llogan> AlexQ: probably not [23:52:45 CET] <AlexQ> Does the .h264 file contain all the info neccessary to play/mux it? Maybe I chose the wrong output format [23:52:52 CET] <AlexQ> Couldn't remux [23:53:23 CET] <durandal_1707> directly from input format [23:55:30 CET] <AlexQ> durandal_1707: You mean that I should remux directly from input mkv? [23:57:49 CET] <AlexQ> http://paste.ubuntu.com/15483500/ [23:58:10 CET] <AlexQ> Guess I can't mux .h264 without specifying the parameters manually or sth? [00:00:00 CET] --- Thu Mar 24 2016

1 0

ffmpeg-devel.log.20160322
by burek 23 Mar '16

23 Mar '16

[00:10:28 CET] <cone-127> ffmpeg 03Lukasz Marek 07master:df34b700981d: ffplay: remove redundant silence buffer [00:30:17 CET] <sumirekousa> Hey all. I'm interested in contributing to ffmpeg for GSOC, namely the subtitle task. I was just wondering how I would go about this? [00:37:12 CET] <wm4> sumirekousa: contact the listed mentor [00:40:01 CET] <sumirekousa> The thing is that the subtitle task currently has no mentor with only Nicolas George listed as a possible mentor. Should I try to contact him? [00:43:54 CET] <Compn> ask ubitux , he may know about it [00:44:07 CET] <Compn> but yes contacting nicolas might work [00:45:48 CET] <sumirekousa> I'll try both [00:45:58 CET] <sumirekousa> Thank for the help [00:51:32 CET] <cone-127> ffmpeg 03Thomas Volkert 07master:b4f32c42ab09: rtpdec: support for VC-2 HQ RTP payload format (draft v1) [00:58:14 CET] Action: Daemon404 slaps atomnuker with optimfrog [01:23:15 CET] <atomnuker> it's a matter of time before you lose your wavpack allegiance, mark my words! [01:23:51 CET] <atomnuker> maybe not, dunno [01:24:19 CET] <kierank> atomnuker: such a nightmare to build afl-clang-fast [01:30:48 CET] <michaelni> atomnuker, theres a overflow in fate-gaplessenc-pcm-to-mov-aac [01:31:50 CET] <michaelni> int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q, ROUNDING), 13); <-- quant() == -2147483648 <-- i guess float is out of int range [01:32:11 CET] <cone-127> ffmpeg 03Carl Eugen Hoyos 07master:30d1213ecd59: configure: Remove (b)zlib and iconv dependencies for videoltoolbox encoder. [01:33:46 CET] <atomnuker> michaelni: odd, will take a look at it tommorow [01:35:10 CET] <michaelni> atomnuker, thx, Q seems 67108864.000000, this is likely why it overflows [01:36:27 CET] <atomnuker> ok, might be the q_idx messing up (since Q is coming from a table) [01:43:47 CET] <michaelni> ubitux, your icc segfaults in on2avc http://fatebeta.ffmpeg.org/report/x86_64-archlinux-icc-2013/20160321171253#… [02:08:47 CET] <michaelni> ubitux, also there are some seemingly random zerocodec failures like: (seems Z_DATA_ERROR) http://fatebeta.ffmpeg.org/report/x86_64-archlinux-gcc-threads-8/2016032117… [02:13:19 CET] <rcombs> anybody have any issue with me pushing the audiotoolbox codecs? [02:14:16 CET] <rcombs> they've been pretty stable in my testing here (which admittedly is not particularly extensive), and since they're after their internal counterparts in allcodecs, they shouldn't break anything just by being present [05:18:00 CET] <Timothy_Gu> does anybody have a IEEE Xplore subscription? [09:37:57 CET] <Shiz> Timothy_Gu: yeah [09:42:53 CET] <cone-118> ffmpeg 03Michael Niedermayer 07master:14478b6c3820: fate: add audiomatch [10:23:28 CET] <ubitux> http://infocenter.arm.com/help/topic/com.arm.doc.dui0802a/ADDP_advsimd_vec_… [10:23:34 CET] <ubitux> $1M question: what does this do? [10:24:50 CET] <wm4> enter a support contract with ARM [10:25:10 CET] <wm4> (that's the only explanation I have for this crap) [10:25:40 CET] <nevcairiel> thats also a way to make money [10:25:48 CET] <nevcairiel> make your shit hard to use and dont publish docs 0p [10:26:01 CET] <ubitux> i can't believe they're describing an instruction with 2 fucking words [10:34:00 CET] <jkqxz> That's horizontal add, right? The documentation for NEON has always been atrocious; writing test programs is best way to find out what instructions do (especially all the strange scatter-gather ones, which are completely impenetrable otherwise). [10:36:26 CET] <ubitux> yeah, so apparenty, when you have Vn.8H={A,B,C,D,E,F,G,H} and Vm.8H={I,J,K,L,M,N,O,P}, you get Vd.8H={A+B,C+D,E+F,G+H,I+J,K+L,M+N,O+P} [11:14:30 CET] <cone-118> ffmpeg 03Carl Eugen Hoyos 07master:ef888de1c1d2: lavf/img2dec: Skip COM when auto-detecting jpeg. [11:46:46 CET] <wm4> ubitux: for the new encode/decode API, I also think I'd like to have AVSubtitle-in-AVFrame... [11:47:53 CET] <ubitux> yes i can understand [11:47:55 CET] <ubitux> :) [11:48:14 CET] <ubitux> it's not that trivial though [12:44:42 CET] <kierank> I wonder if J_Darnley is ok [13:00:24 CET] <kierank> J_Darnley: ah you are alive [13:00:29 CET] <kierank> good to hear [13:00:43 CET] <JEEB> always good to know people are fine :) [13:00:57 CET] <J_Darnley> Yeah, unfortunately [13:02:22 CET] <kierank> really shocking to see that terminal that we use every year for fosdem in pieces [13:02:30 CET] <J_Darnley> Huh? [13:02:44 CET] <J_Darnley> Oh [13:03:00 CET] <J_Darnley> I missed getting killed by terrorists again [13:04:20 CET] <wm4> again? [13:04:30 CET] <ismail> J_Darnley: well done. [13:08:52 CET] <kierank> michaelni: how does an in-place dwt/idwt work with buffer referencing? [13:09:29 CET] Action: ubitux wonders if it's better to reuse the same register for source and destination or use different registers, on aarch64 [13:13:55 CET] <ubitux> same question on x86 and arm actually [13:56:05 CET] <wm4> michaelni: can you explain this code in libswresample: if( av_get_planar_sample_fmt(s-> in_sample_fmt) <= AV_SAMPLE_FMT_S16P [13:56:33 CET] <wm4> why does it do a numerical comparison where it makes no sense [13:57:26 CET] <wm4> my actual problem is that it downmixes using s16 [13:57:29 CET] <wm4> which causes clipping [13:57:47 CET] <wm4> and I always wondered why users reported clipping in some cases... [13:57:57 CET] <wm4> and it's this impenetrable shit code [13:57:58 CET] <wm4> sigh [14:08:12 CET] <michaelni> wm4, it compares planar <= AV_SAMPLE_FMT_S16P as the formats are sorted by bits [14:09:44 CET] <wm4> nothing says they're sorted [14:09:59 CET] <wm4> and a hypothetical new format would be added at the end of the enum anyway [14:10:39 CET] <michaelni> i didnt say that i consider that code to be great [14:15:50 CET] <ubitux> weren't the samples format done such that we could toggle a bit to switch from planar to packed at one point? (and then it changed later) [14:21:25 CET] <cone-871> ffmpeg 03Michael Niedermayer 07master:914ad90eddce: swresample/swresample: Remove "less than" comparissions of enums [14:50:31 CET] <gerion> hi, i'm currently try to implement the binary output. is there somewhere in the framework a (predefined) function to write bitwise to a buffer or file or do i have to shift myself? [14:51:29 CET] <gerion> i mean some kind of put_bit(buf, 0), put_byte(buf, some_char), etc [14:51:42 CET] <jkqxz> libavcodec/put_bits.h ? [14:53:26 CET] <gerion> hmm thx, looks good, why it is in libavcodec? [14:58:13 CET] <jkqxz> It's used for writing coded bitstreams. What else would you want to do with it, that might justify it being outside libavcodec? [15:00:30 CET] <gerion> I want to use it for the signature filter that could write binary files. beside that, I thought, that some muxer needs such functions. [15:04:27 CET] <jkqxz> libavformat explicitly depends on libavcodec. Some things in libavfilter depend on libavcodec too, though that's on a per-filter basis. [15:06:55 CET] <gerion> ok, don't know the first one (actually searched libavutil for such a function) [15:07:19 CET] <gerion> but make sense now [15:08:53 CET] <cone-871> ffmpeg 03Rostislav Pehlivanov 07master:2aebdfb451bb: vc2enc: add non-experimental support for all video formats from spec [15:10:13 CET] <jkqxz> $(grep 'filter_deps.*avcodec' configure) - adding another one with that dependency wouldn't exactly be a disaster. [15:11:46 CET] <Daemon404> atomnuker, it would be cool if you could test ffa1 against the proprietary formats too. also decoding speed. [15:25:07 CET] <atomnuker> plenty of time for that, still having fun with the entropy encoding system [15:25:17 CET] <Daemon404> ;p [15:25:17 CET] <atomnuker> also still need a dct [15:25:39 CET] <Daemon404> for a lossless codec? [15:26:35 CET] <atomnuker> myeah, want to see how big the errors would be [15:26:49 CET] <Daemon404> er? [15:27:13 CET] <atomnuker> oh, wait, perfect integer reversible dcts exist [15:30:09 CET] <Daemon404> you could design your own dct [15:30:34 CET] <atomnuker> it's fine, I know just where to borrow one [15:30:46 CET] <Daemon404> (but really this is an excuse so i can mention the great series of blog posts about designign a dct by ryg) [15:31:06 CET] <atomnuker> was it the guy from rad video game tools? [15:31:18 CET] <Daemon404> yes that is ryg [15:31:51 CET] <atomnuker> yeah, that blog post was nice [15:32:02 CET] <Daemon404> posts =p [15:32:45 CET] <atomnuker> oh, I've only read part 1 [15:32:49 CET] <Daemon404> theres 2 [16:43:44 CET] <rcombs> last call for comments on the AudioToolbox code, else I'll push in a couple hours [17:10:03 CET] <durandal_1707> rcombs: why? [17:10:25 CET] <rcombs> durandal_1707: why what? [17:11:01 CET] <wm4> maybe asking for the meaning of life [17:11:21 CET] <durandal_1707> this is about decoders wrapper? [17:11:37 CET] <rcombs> and encoders, yeah [17:12:05 CET] <rcombs> been sitting on the ML for 3 weeks with no substantial comments (one bit that I should bump minor instead of micro and mention a trac issue in the commit message) [17:12:28 CET] <rcombs> and since they're after the internal implementations in allcodecs.c, they won't break anything by being present [17:12:29 CET] <durandal_1707> wm4: why you still haven't done multi vo window feature in mpv? [17:12:42 CET] <wm4> why would I [17:12:42 CET] <nevcairiel> i find the whole idea silly, but people prefer having features over sanity, so shrug [17:14:18 CET] <durandal_1707> wm4: because I need it [17:14:32 CET] <wm4> busy with other stuff [17:26:36 CET] <cone-871> ffmpeg 03Petru Rares Sincraian 07master:124526ba1aee: Added a selftest to libavutil/display.c [18:21:01 CET] <cone-871> ffmpeg 03Ganesh Ajjanagadde 07master:db1a642cd213: all: move ff_exp10, ff_exp10f, ff_fast_powf to lavu/ffmath.h [18:48:25 CET] <Daemon404> /g/g 42 [18:51:42 CET] <cone-871> ffmpeg 03Rodger Combs 07master:d5d328059e51: lavc: add AudioToolbox decoders [18:51:43 CET] <cone-871> ffmpeg 03Rodger Combs 07master:65cff814534c: lavc: add AudioToolbox encoders [18:52:23 CET] <wm4> whooh [18:58:46 CET] <durandal_1707> now just add dshow wrapper [19:39:23 CET] <atomnuker> what the hell is ganesh's problem? [19:49:26 CET] <nevcairiel> he must like being annoying [20:52:51 CET] <wm4> why do the fate-lavf tests start a bunch of ffmpegs for each test [20:52:58 CET] <wm4> Libav doesn't even have this stuff [20:53:06 CET] <wm4> I can't see where or why what is failing [20:54:57 CET] <jamrial> one to encode/mux a file, another to demux/decode the result, afaik [20:55:16 CET] <wm4> how can I run just one of them [20:57:12 CET] <Daemon404> V=1, copypase commands [20:57:16 CET] <Daemon404> its tedious but its how i did it [20:57:22 CET] <Daemon404> for the codecpar branch testing [21:05:25 CET] <Zeranoe> Does FFmpeg have x86 assembly specific components? [21:05:45 CET] <J_Darnley> Yes. [21:05:54 CET] <J_Darnley> All SIMD opts [21:06:05 CET] <J_Darnley> (well almost all, some is still inline) [21:06:10 CET] <Shiz> depends if you mean 'functionality specific to x86' or 'code specific to x86' [21:06:41 CET] <Zeranoe> code specific to x86 [21:06:50 CET] <Shiz> yes [21:07:37 CET] <Zeranoe> Anything other than SIMD opts? [21:08:06 CET] <J_Darnley> I think there is branchless cabac for the h264 decoder [21:08:13 CET] <J_Darnley> and probably other places [21:09:02 CET] <J_Darnley> Most of it is assembly of one type or another for speed [21:09:02 CET] <Shiz> https://github.com/FFmpeg/FFmpeg/search?utf8=%E2%9C%93&q=language%3AAssembly [21:09:24 CET] <J_Darnley> I don't believe any features are tied to x86 [21:09:52 CET] <J_Darnley> Only to x86 operating systems: x11 capture, gd1 capture, dshow input, etc. [21:10:23 CET] <Shiz> x11 ain't x86 only [21:27:28 CET] <wm4> I still don't know what this test is complaining about [21:27:34 CET] <rcombs> does vaapi count [21:28:27 CET] <TD-Linux> no, the talos power8 board can use radeon graphics with vdpau and there is a vaapi->vdpau bridge [21:29:38 CET] <fritsch> there is no nicely working bridge anywhere [21:29:57 CET] <fritsch> on linux you still need to implement both if you want to use them correctly [22:19:20 CET] <wm4> 0, 264, 264, 1, 13368960, 0xebe37433 [22:19:21 CET] <wm4> -0, 265, 265, 1, 13368960, 0xa1b5c8e1 [22:19:21 CET] <wm4> +0, 264, 264, 1, 13368960, 0xa1b5c8e1 [22:19:31 CET] <wm4> wonder if I should "fix" this... [22:19:43 CET] <wm4> ah nevermind [22:19:53 CET] <wm4> that's completely legitimate [22:50:44 CET] <wizonesolutions> Hi I've been trying the patch at https://trac.ffmpeg.org/ticket/4437#comment:21. What I am seeing is that things are better when recording and ffmpeg is the only thing recording, but if I try to record the screen when another application is also using the microphone, then the audio comes out poorly. Is it some sort of contention issue? It's Chrome that [22:50:44 CET] <wizonesolutions> is using the microphone over WebRTC. [00:00:00 CET] --- Wed Mar 23 2016

1 0

ffmpeg.log.20160322
by burek 23 Mar '16

23 Mar '16

[01:00:29 CET] <causasui> i'm using ffmpeg to livestream via twitch.tv. everything works great, except the video is from my secondary desktop not the primary (the secondary is to the left of the primary). can I add an offset to the frame size set via -s, or an offset somewhere else? [01:46:32 CET] <llogan> causasui: i don't quite understand your question, but showing your command and the complete console output will be helpful [01:53:42 CET] <jookiyaya> what is faster? $10,000 computer in 2016 or fastest supercomputer in year 2000 ? [01:55:01 CET] <causasui> llogan: all right, please hold [01:55:26 CET] <mrmenacex> does anyone know why a vp9 video in the regular html5 video tag would work on Android using chrome but not on ios 9.2.1 using chrome ? [01:55:38 CET] <causasui> llogan: here's the script http://sprunge.us/ZISA # there aren't any errors in the console output, do you still need that? [01:56:41 CET] <furq> mrmenacex: we might if you post ffmpeg/ffprobe output [01:57:03 CET] <mrmenacex> what's ffprobe ? [01:57:24 CET] <mrmenacex> there is no ffmpeg output really it's already encoded [01:57:47 CET] <furq> does mobile chrome ship with video decoders [01:57:53 CET] <furq> if it doesn't then it's down to iOS not supporting vp9 [01:58:36 CET] <mrmenacex> furq: it works in mobile chrome on android [01:58:42 CET] <J_Darnley> Apple wouldn't allow a viable alternative to safari, would they. [01:58:47 CET] <furq> well yeah but android supports vp9 [01:59:02 CET] <relaxed> causasui: the first -s should be -video_size [01:59:21 CET] <furq> iOS chrome isn't listed on caniuse so i have no idea if it's supported [01:59:23 CET] <relaxed> first -r should be -framerate [01:59:38 CET] <furq> it definitely isn't in desktop or iOS safari though [01:59:41 CET] <mrmenacex> it doesn't work in safari either [01:59:45 CET] <furq> ^ [01:59:50 CET] <furq> http://caniuse.com/#feat=webm [02:00:21 CET] <J_Darnley> Why isn't <video> just a wrapper around libavformat? [02:00:38 CET] <furq> one would hope that google would bundle a decoder for their own codec, but i wouldn't put it past apple to have prevented them [02:01:08 CET] <furq> either that or this is their way of forcing h.264 on youtube with iOS chrome so that it doesn't eat all your battery [02:02:45 CET] <mrmenacex> furq: that sucks i just saw caniuse :( [02:10:46 CET] <causasui> relaxed: ok I'll swich that. can you help me figure out how to set an offset, though? [02:36:05 CET] <relaxed> causasui: pastebin.com the command and console output [02:36:20 CET] <causasui> relaxed: command is just $scriptname $key [02:37:23 CET] <relaxed> and console output is $nowheretobeseen [02:38:01 CET] <relaxed> causasui: did you read https://trac.ffmpeg.org/wiki/Capture/Desktop ? [02:38:40 CET] <furq> causasui: -i :0.0+1920,0 [02:39:09 CET] <furq> replace 1920 with the horizontal resolution of your left monitor [02:39:45 CET] <llogan> causasui: for future reference we don't provide help for scripts here because sometimes the problem is with the script and not with ffmpeg, so we need unscriped commands and the complete console output. makes things easier to duplicate issues too. [02:40:25 CET] <furq> that is true as a general rule but i don't think it makes a difference here [02:40:36 CET] <causasui> relaxed: sorry, I had to trap it. http://sprunge.us/ecEW [02:40:37 CET] <causasui> furq: that looks like what I wanted. i'll try it, thanks [02:40:37 CET] <causasui> hmm, error: [x11grab @ 0x556ba38e6380] Capture area 1920x1080 at position 1920.0 outside the screen size 3600x1080 [02:40:37 CET] <causasui> :0.0+1920,0: Invalid argumenty [02:40:37 CET] <causasui> i think it should be 1680 though? [02:40:47 CET] <furq> 01:39:09 ( furq) replace 1920 with the horizontal resolution of your left monitor [02:40:53 CET] <furq> so yeah [02:40:56 CET] <llogan> also, apparently twitch now recommends aac instead of mp3 [02:41:24 CET] <causasui> furq: yep, 1680 did it, thank you! [02:41:55 CET] <furq> i've had some issues in the past with browsers playing back mp3 in flv, so i'd recommend aac anyway [02:42:05 CET] <llogan> http://trac.ffmpeg.org/wiki/EncodingForStreamingSites?action=diff&version=5 [02:42:49 CET] <causasui> llogan: i'll keep that in mind, thank you [02:43:11 CET] <llogan> are you sure you need a -minrate? [02:43:40 CET] <furq> https://trac.ffmpeg.org/wiki/StreamingGuide [02:43:45 CET] <llogan> and that "-strict normal" looks superfluous. [02:43:50 CET] <furq> should this page really be recommending ffserver [02:44:09 CET] <llogan> that page should eb completely rewritten. it's a mess. mostly written by one guy. [02:44:21 CET] <furq> yeah that whole second paragraph needs changing [02:44:24 CET] <furq> at least [02:45:02 CET] <llogan> too verbose. not enough structure or formatting. "typing as I think/notes to self" style. [02:45:29 CET] <furq> does -vpre even exist any more [02:45:42 CET] <llogan> IIRC, perhaps for custom presets. [02:46:02 CET] <furq> "Here is what another person once did:" [02:46:05 CET] <relaxed> it has -r 15 before the input :/ [02:46:06 CET] <furq> yeah this isn't good is it [02:46:36 CET] <llogan> furq: that sentence always bothered me but i don't have the energy to dive into that rabbithole [02:46:59 CET] <furq> if i streamed more often i might have a go at it [02:47:19 CET] <llogan> please do [02:47:43 CET] Action: relaxed votes furq should stream more [02:47:47 CET] <furq> i have no idea about constraining bandwidth or anything other than rtmp though [02:51:50 CET] <papamoose_> Anyone know how to also copy album art when copying streams to a different container? Right now I have this(http://pastebin.com/raw/PtgtWZsE) which works well, except it does not have the album artwork. [02:52:15 CET] <llogan> causasui: "threads="2" # max 6" you may want to allow libx264 to automatically choose threads number. not sure why it says "max 6". [02:55:34 CET] <c_14> papamoose_: ID3 tags? then not that I'm aware of [02:55:57 CET] <c_14> ffmpeg converts the album artwork into a video stream internally [02:56:48 CET] <papamoose_> fflogger: http://pastie.org/private/wie3muclridd2sn6dcgskg [02:57:02 CET] <c_14> That's not the command output. [03:00:03 CET] <papamoose_> c_14: i have tried -vcodec copy to copy the album artwork except that it returns an error when I add that option. [03:00:53 CET] <c_14> what error? [03:05:04 CET] <papamoose_> c_14: http://pastie.org/private/n9kvstqhrgh3mmj8swgow [03:06:45 CET] <c_14> weird, and if you don't specify -vcodec copy? [03:06:45 CET] <papamoose_> It seems to be on line 345: http://pastie.org/private/n9kvstqhrgh3mmj8swgow#345 [03:07:44 CET] <J_Darnley> ffmpeg is too clever for its own good. No files have "embedded covers" but some files do have a 1 frame video stream. [03:07:58 CET] <c_14> eh, it might be the j [03:08:17 CET] <c_14> The pix formats with j are deprecated, it might just not be in the list [03:08:35 CET] <c_14> The parser/demuxer should probably set range though... [03:08:44 CET] <papamoose_> http://pastie.org/private/9tn68djrstdbh2daupg#1 [03:09:06 CET] <c_14> papamoose_: get rid of the -vn [03:09:15 CET] <c_14> try letting ffmpeg pick the codec [03:09:44 CET] <relaxed> papamoose_: -c:v libx264 -pix_fmt yuv420p [03:11:29 CET] <papamoose_> c_14: getting rid of -vn did not seem to work. trying relaxed's method. [03:11:30 CET] <relaxed> but that will create a video stream [03:12:54 CET] <papamoose_> relaxed: hmm.. that is not ideal. [03:14:02 CET] <relaxed> papamoose_: right, use -vn and then "mp4box -itags cover=cover.jpg input.m4a" [03:16:06 CET] <papamoose_> mp4box is a different utility? [03:16:25 CET] <relaxed> yes, it's from gpac [03:18:25 CET] <papamoose_> relaxed: hmm.. so I know I can grab the cover like this: ffmpeg -activation_bytes ${ACTIVATION_BYTES} -i ${m} -an -vcodec copy cover.jpg [03:24:32 CET] <papamoose_> relaxed: i have modified my script to include mp4box to add the cover after the conversion and it looks like it is working now. [03:24:52 CET] <papamoose_> Any one know why ffmpeg cannot do this by default? [03:25:02 CET] <J_Darnley> What did I say above? [03:25:05 CET] <J_Darnley> ffmpeg is too clever for its own good. No files have "embedded covers" but some files do have a 1 frame video stream. [03:26:24 CET] <papamoose_> J_Darnely: sorry I didn't see your response. Would you mind clarifying: "ffmpeg is too clever for its own good"? [03:26:39 CET] <J_Darnley> Read the second sentence in that line. [03:26:47 CET] <TD-Linux> really I should probably file this as a bug [03:26:50 CET] <TD-Linux> someone hits this once a week [03:27:49 CET] <papamoose_> J_Darnley: I understand that, but I think I'm still missing something. If there is a 1 frame video stream, then -vcodec copy should do what I need..? [03:28:11 CET] <J_Darnley> You need a way to tell ffmpeg that it isn't a video stream. [03:28:27 CET] <papamoose_> Is there such a way? [03:28:32 CET] <J_Darnley> No!! [03:28:39 CET] <J_Darnley> That's half the problem! [03:28:54 CET] <papamoose_> ahh. I see. :( [03:29:00 CET] <J_Darnley> The other half is treating a binary data attachment as a stream. [03:29:59 CET] <J_Darnley> Another half is the concept of embedded covers in the first place [03:30:11 CET] <papamoose_> J_Darnley: Thanks for explaining that. :) [03:30:13 CET] <J_Darnley> 150% problem. [03:32:48 CET] <papamoose_> Well, thanks all for helping me solve this for now. [05:25:24 CET] <seeit> hello, does anyone know how to speed up the conversion of a .mkv file on arm? right now its is about 0.5 which is too slow for streaming [05:34:21 CET] <c_14> -preset ultrafast [07:11:13 CET] <Mista_D> Hi, passing "filter_complex" to FFmpeg from a tmp.txt; can't seem to read command. [ "-filter_complex foo,bar" ] fails, while [ -filter_complex "foo,bar" ] doesn't. How would I store and pass the filter chain via variable in bash avoiding the quotations please? [07:11:54 CET] <Mista_D> [ -filter_complex "foo,bar" ] doesn't fail ( : [08:48:26 CET] <pezet91> hello, I have encoded H264 stream from video grabber, but I can't restream this with avconv/ffmpeg. Maybe I can't set parameters to avconv/ffmpeg. Anyone can help me with this? [13:04:59 CET] <Gringham> Hey :) Do you know a way, to pass a list of files to stream via ffmpeg, so that directly after one end the next one starts (without resetting the Timestamp) ? [13:08:49 CET] <J_Darnley> Use one of the three concat options [13:09:16 CET] <Gringham> I'll look after it, ty :) [15:37:44 CET] <Gringham> Is there a way to concatenate via the concat demuxer and reading the files not from a text file but from a List ? [15:38:10 CET] <J_Darnley> What is a "List" and how can ffmpeg read it? [15:39:25 CET] <J_Darnley> Ultimately I will say the answer is no. [15:39:32 CET] <J_Darnley> Write this List to a file [15:54:04 CET] <Gringham> ffmpeg -f concat -i C:\Users\name\Desktop\int.txt -f mpegts udp://239.255.1.0:1239 [15:54:19 CET] <Gringham> this is what I tried to do now [15:54:33 CET] <Gringham> the first video is playing, than it mutters about the format [15:54:52 CET] <furq> what format are the videos [15:56:16 CET] <Gringham> ts [15:56:50 CET] <Gringham> [mpegts @ 0000002f72d30540] Non-monotonous DTS in output stream 0:0; previous: 2707320, current: 12848; [16:08:50 CET] <leogps> I am using ffmpeg to re encode rtmp stream into mpegts segmented files on Ubuntu 13. The video gets stuck for every 5 secs with a glitchy sound in the audio. [16:18:58 CET] <leogps> http://pastebin.com/SJNxbev7 [16:22:06 CET] <J_Darnley> Good to see people have no clue what they are doing. [16:28:18 CET] <satinder___> Hi , any one there [16:29:05 CET] <satinder___> if we want update over textfile for drawtext (overlay purpose) then how we can rewrite it [16:29:21 CET] <satinder___> I using echo in system command [16:29:42 CET] <satinder___> but I am not sure that is good or not but that is working [16:30:28 CET] <satinder___> but i listen system command is not good , so please anyone give any opinion , what I can do for that [16:30:31 CET] <satinder___> thanls [16:30:36 CET] <satinder___> thanks [16:35:24 CET] <petecout_> Is there a metadata value in ffprobe that shows a file is currently being encoded? I have an application that needs to monitor mpegts files as they are being generated to see if they are done. [16:36:48 CET] <kepstin> petecout_: no obvious way with mpeg-ts. You'll want to just stat the file at interval and see if the mtime keeps updating or file size keeps changing. [16:41:21 CET] <petecout_> kepstin: Thanks mate. I figured if there wasn't a flag I'd go by file count to see which one the system was currently one. [16:41:27 CET] <petecout_> on [16:41:37 CET] <petecout_> I move them to a different folder when they are done. [16:46:09 CET] <Fyr> JEEB, hey, is there news on the delay bug? [17:25:22 CET] <brick> ffmpeg -i try_%02d.png -vcodec mpeg4 try.avi produces a movie one frame long when there are dozens of images there starting with try_01.png. what am i doing wrong? [17:36:54 CET] <Gringham> when I concatenate two files like this : "concat:file1.ts|file2.ts" file2.ts has no audio anymore (file1 never had one) , is there a way to prevent this ? [17:37:31 CET] <Gringham> when I run "concat:file2.ts|file1.ts" it still play the audio [17:37:52 CET] <Gringham> plays* [17:40:22 CET] <brick> Gringham, when you do a "mix" i think you get to specify which streams to use in the output [18:15:08 CET] <explodes_> I'm trying to update my library by upgrading FFmpeg from 2.8.6 to 3.0. There are a few deprecations, and the app that uses the library now crashes. - Is it worth the upgrade process to use 3.0? [18:18:18 CET] <thebombzen> brick: try_02d.png needs there to be try_01.png, try_02.png, etc. [18:23:08 CET] <brick> thebombzen, OK, that might have been why. when i try this again i will make sure that there are no missing numbers. [18:23:09 CET] <brick> thanks!Q [20:44:19 CET] <petecout_> When your encoding mpegts segements for a HLS stream. The last segment is always 'inprogress' as it's encoded to. Is it best practice to prevent that working segment from being downloaded by the client or can HLS clients handle partial segemetns? [20:53:08 CET] <pyro25> hey guys. I started playing with ffserver and I'm realizing how cool it would be to have a database-backed configuration. anyone has ever heard of this? [20:55:12 CET] <blue_misfit> hey, so f4v is basically mp4, but is flv also comparable to mp4 or is it really its own thing? [21:17:51 CET] <kepstin> blue_misfit: flv is basically a serialization of a flash command stream, so quite a bit different [21:18:28 CET] <kepstin> (although there might be some cases where an flv stream contains an mp4 or mp4-like stream? I'm not sure entirely how that works) [21:28:30 CET] <awefoijawef> hi [21:28:57 CET] <Azrael_-> hi [21:34:02 CET] <Fyr> hi [21:36:01 CET] <petecout_> Ello [21:44:04 CET] <petecout_> Has anyone ever run into this issue before an error happened: ffmpeg exited with code 1: public/sdps/video-audio3.sdp: Invalid data found when processing input [21:44:20 CET] <petecout_> It seems to require a system restart when it occurs the first time. [21:48:06 CET] <blue_misfit> kepstin, thanks :) [22:05:43 CET] <Mista_D> what' [22:06:06 CET] <Mista_D> s the recommended gcc version please? [22:19:32 CET] <llogan> Mista_D: there is not a specifc recommended version [22:20:33 CET] <Mista_D> Understood, will play with it... [22:20:39 CET] <Mista_D> Thanks [22:27:48 CET] <llogan> Mista_D: however, there may be some versions that could recommended to *not* use. https://trac.ffmpeg.org/ticket/3970 [22:28:14 CET] <llogan> i'm unsure how relevant that ticket still is [22:29:13 CET] <llogan> also #1464 [22:32:53 CET] <Mista_D> allright, do not use gcc 4.2. got 4.4.7 with Centos... [22:40:43 CET] <squarecircle> heyo [22:41:37 CET] <squarecircle> is there a way to dismantle a video in singular frames and write them to stdout? [22:43:52 CET] <squarecircle> at the moment I'm calling ffmpeg very often to get frames at with increasing timestamps, but I would prefer to actually create a stream and parse this stream [23:01:47 CET] <petecout_> Before I test this. Is it ok to use the same uri for input as output? [23:02:45 CET] <Mavrik> squarecircle, dump raw frames to stdout? [23:02:48 CET] <Mavrik> there's raw output for that [23:07:19 CET] <kepstin> petecout_: depends entirely on what kind of uri it is. [23:07:39 CET] <kepstin> petecout_: you'll have bad times if you use the same file:/// uri for input and output, for example :) [23:07:49 CET] <kepstin> and with e.g. http, rtmp it depends on the server, not ffmpeg. [23:13:18 CET] <squarecircle> Mavrik: sounds good, do you have a hook for me, where to look? [23:15:41 CET] <Mavrik> ffmpeg -i <input> -codec:v rawvideo -f rawvideo - [23:15:44 CET] <Mavrik> should work I guess [23:15:55 CET] <Mavrik> Note that rawvideo will be whatever the frames in video are, usually YUV420p [23:16:36 CET] <livingBEEF> Are frames counted from 1 or from 0? [23:17:31 CET] <petecout_> kepstin: thnx mate. It's just a local file in path temp/vid.ts temp/vid.ts [23:17:34 CET] <petecout_> It seems to be working ok [23:18:43 CET] <kepstin> petecout_: hmm, i wouldn't expect that to work; when writing the output it should be truncating the file before writing... [23:18:51 CET] <kepstin> maybe it's removing the file and creating a new one instead [23:18:55 CET] <squarecircle> Mavrik: do you know how they are separated? [23:19:03 CET] <Azrael_-> i'm streaming with ffmpeg a live-stream to an nginx-rtmp-server but the video occasionally just stops for a few seconds and then proceeds. the upload+download isn't overloaded, cpu also free. any suggestions how i could debug this? [23:19:09 CET] <Mavrik> squarecircle, they're not afaik [23:19:21 CET] <Mavrik> squarecircle, you need to know the size of the video :) [23:19:36 CET] <kepstin> livingBEEF: I think in most places in ffmpeg frames are counted starting at 0. [23:19:54 CET] <squarecircle> so I try to ffprobe the resolution and then read in the size of each frame [23:19:55 CET] <squarecircle> ? [23:20:02 CET] <livingBEEF> kepstin: thanks [23:20:25 CET] <Mavrik> squarecircle, yes, and then calculate size in bytes for each frame [23:20:35 CET] <Mavrik> and that's what you're gonna get in sequence [23:22:36 CET] <kepstin> you have to have some knowledge of the pixel format being used to calculate the size correctly. (if you want to make it easy on yourself, you can use the -pix_fmt option to tell ffmpeg to convert everything to one format) [23:23:57 CET] <petecout_> kepstin: Could be. I'm going to test and I'll let you know if there are any issues. [23:27:56 CET] <squarecircle> Mavrik: that sounds a bit interesting [23:30:49 CET] <petecout_> kepstin: You were right lol. Can't process same file on file system. [23:40:10 CET] <squarecircle> Mavrik: Wikipedia tells me that yuv420 is "YUV420p 6 bytes per 4 pixels, reordered" [23:41:01 CET] <squarecircle> Mavrik: the rest should be easy goung [23:42:05 CET] <squarecircle> Mavrik: "-pix_fmt rgb24" should be working, or? [23:42:32 CET] <Mavrik> if you want to convert it yes :) [23:43:06 CET] <Mavrik> rgba is also common [23:43:47 CET] <squarecircle> Mavrik: I want to do some magic with python [23:43:48 CET] <squarecircle> ^^ [23:45:38 CET] <squarecircle> Mavrik: so it would be "ffmpeg -i testvid.mkv -codec:v rawvideo -pix_fmt rgb24 -f rawvideo -" [23:46:56 CET] <squarecircle> Mavrik: nah: "ffmpeg -i testvid.mkv -codec:v rawvideo -f rgb24 -" ? [00:00:00 CET] --- Wed Mar 23 2016

1 0

ffmpeg-devel.log.20160321
by burek 22 Mar '16

22 Mar '16

[00:11:03 CET] <cone-095> ffmpeg 03Neil Birkbeck 07master:e7e5c5e6c4d7: lavf/matroskaenc.c: add early support for colour elements [08:38:59 CET] <j-b> 'morning [08:42:00 CET] <llogan> 'night. about to "hit the sack" as they say in English. Really, it's not as painful as it sounds. [08:55:20 CET] <ethe> has anyone worked on reversing OptimFROG? [09:31:57 CET] <slomo> hi! what's the official way to distinguish between libav and ffmpeg at compile time and runtime? previously i was told that ffmpeg always has LIBAVCODEC_VERSION_MICRO < 100, and libav >= 100. but that's obviously not true anymore since 3.0 [09:32:36 CET] <wm4> slomo: it's not? [09:32:50 CET] <JEEB> I don't think there's an official way but check ffms2's configure checks [09:33:08 CET] <wm4> the micro thing is official [09:33:30 CET] <wm4> (as official as it could get with cranky assholes not even acknowledging the existence of Libav) [09:33:52 CET] <wm4> slomo: also, you got it the wrong way around [09:34:12 CET] <wm4> ffmpeg has micro versions >= 100 [09:34:22 CET] <slomo> wm4: https://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/version.h;h=6e42… [09:34:40 CET] <slomo> so it's actually the other way around? [09:34:44 CET] <wm4> yes [09:35:42 CET] <slomo> ok, thanks :) [09:40:42 CET] <durandal_170> ethe: nope [11:06:25 CET] <wm4> I'm really wondering what kind of shit causes code like "st->priv_pts->num != st->priv_pts->den >> 1" [11:07:45 CET] <kierank> wm4: "Fix sample #1234" [11:09:20 CET] <wm4> originally it was added as " fix obnoxious ogg_packet passing from encoder to muxer" in 2004 [11:10:29 CET] <wm4> hm I wonder if it'd possible to write a tool that eliminates all code from ffmpeg that doesn't change FATE results [11:14:06 CET] <nevcairiel> if you want to get rid of all error checks, sure =p [11:14:33 CET] <ubitux> according to coverage.ffmpeg.org you can trash 40% of the code [11:15:31 CET] <Shiz> wm4: mutation testing! [11:50:49 CET] <ubitux> so i'm setting hyScale and hcScale to some random function in sws, but it appears that on x86 the filtersize doesn't match the one on aarch64 [11:50:54 CET] <ubitux> any hint on the reason? [11:51:09 CET] <nevcairiel> there is a bunch of special conditions to masage that value [11:51:26 CET] <ubitux> where is this done? [11:51:39 CET] <nevcairiel> somewhere in sws [11:51:46 CET] <nevcairiel> dont expect me to remember where this spaghetti ends [11:51:46 CET] <nevcairiel> :D [11:52:24 CET] <ubitux> :( [11:53:40 CET] <nevcairiel> initFilter maybe? [11:54:09 CET] <nevcairiel> it things like HAVE_MMX && cpu_flags & MMX in it [11:54:36 CET] <nevcairiel> (thats in utils.c) [11:56:31 CET] <ubitux> nevcairiel: thanks [11:57:31 CET] <ubitux> const int filterAlign = X86_MMX(cpu_flags) ? 2 : [11:57:33 CET] <ubitux> PPC_ALTIVEC(cpu_flags) ? 8 : 1; [11:57:35 CET] <ubitux> aaaah [12:00:04 CET] <ubitux> actually, the one above, for hscale [12:06:53 CET] <cone-229> ffmpeg 03Rostislav Pehlivanov 07master:500dc20deede: vc2enc: fix segfault [12:06:54 CET] <cone-229> ffmpeg 03Rostislav Pehlivanov 07master:d4773c94a6b5: vc2enc: simplify count_hq_slice() and caching [13:57:12 CET] <BBB> oh poor ubitux [13:57:15 CET] <BBB> getting lost in sws [13:57:29 CET] <nevcairiel> its his own fault for caring about writing sws asm [13:58:56 CET] <ubitux> well it's fine [13:59:04 CET] <ubitux> mostly [14:28:52 CET] <kierank> I'm not sure accepting that mpeg-7 patch is a good idea [14:29:07 CET] <kierank> it's a complicated academic spec and most likely the author will disappear afterwards [14:30:07 CET] <kierank> but i guess MOAR FEATURES [14:31:49 CET] <J_Darnley> If it was good I might support it but the guy even says that transformations will produce a different signature. [14:33:55 CET] <wm4> hm yet another crazy mpeg thing [14:34:24 CET] <atomnuker> I don't like adding more GPL stuff [14:35:04 CET] Action: J_Darnley shouts "boo" [14:37:34 CET] <nevcairiel> is that what youtube uses and why some people upload videos mirrored to avoid detection? =p [14:54:58 CET] <Daemon404> i am fairly certain i had a prof in university, who was into MPEG-7... [14:55:04 CET] <Daemon404> i assumed there was a reason it went nowehere [14:55:16 CET] <Daemon404> or im rememebring some other crazy mpeg spec [14:55:26 CET] <Daemon404> nevcairiel, youtube uses proprietary stuff [14:56:10 CET] <Daemon404> wm4, i build ffmpeg with gcc and -O0 all the time [14:57:11 CET] <wm4> wut [14:57:25 CET] <Daemon404> --optflags=-O0 [14:57:27 CET] <Daemon404> builds fine [14:57:41 CET] <nevcairiel> did you verify thats what its actually using :D [14:57:50 CET] <Daemon404> with V=1 yea [14:57:59 CET] <Daemon404> and it got rid of <optimized away> [14:59:28 CET] <wm4> I guess some compilers do trivial DCE even at -O0? [14:59:36 CET] <Daemon404> who knows [14:59:57 CET] <nevcairiel> and some compilers fail at it even at full optimization (*cough*clang*cough*) [15:01:09 CET] <Daemon404> yep [15:02:05 CET] <ubitux> aren't we supposed to use --disable-optimisation? [15:02:38 CET] <Daemon404> that doesnt use -O0 iirc [15:02:44 CET] <Daemon404> but maybe i misremember [15:03:28 CET] <nevcairiel> that defaults to using no -O flag iirc [15:04:21 CET] <nevcairiel> whatever that does then [15:11:53 CET] <ubitux> it's kinda original how the hscale x86 asm proceed to increment the width [15:12:40 CET] <ubitux> instead of having a straightforward p[idx], it's doing sth like p_end[-w] and incrementing w [15:12:48 CET] <ubitux> i guess it's saving a reg or sth [15:13:00 CET] <ubitux> but damn that's confusing [15:13:30 CET] <wm4> libswscale code is engineered to create despair [15:14:27 CET] <ubitux> it's ok, just curious :) [15:14:39 CET] <kierank> ubitux: that's a normal idiom, no? [15:14:55 CET] <kierank> as much as a I like to bash swscale [15:15:36 CET] <jamrial> ubitux: that's really common in asm [15:15:36 CET] <J_Darnley> Yes it is. [15:15:59 CET] <J_Darnley> Otherwise yu need to compare against width [15:16:15 CET] <J_Darnley> or whetever your limiting value is [15:18:07 CET] <ubitux> jamrial: in x86 maybe [15:18:25 CET] <ubitux> but sure, ok :) [15:18:50 CET] <ubitux> in arm you don't really need to do that [15:29:18 CET] <wm4> so what's the right way to prevent ffmpeg from picking up winpthreads just for clock_gettime and nanosleep? [15:29:25 CET] <wm4> on mingw [15:30:01 CET] <Daemon404> yell at mingw? [15:30:12 CET] <Daemon404> i hard disable winpthreads because it is crap [15:30:25 CET] <wm4> how do you hard disable it? [15:30:34 CET] <Daemon404> i build a toolchain without it. [15:31:45 CET] <wm4> what makes me wonder is what makes ffmpeg link against the DLL at all [15:31:48 CET] <wm4> is it the -lrt? [15:33:38 CET] <michaelni> JEEB, as you are a contributor you should get voice [15:37:12 CET] <Daemon404> context? [15:37:44 CET] <wm4> well, my windows binaries are created with a dependency on libwinpthread.dll or something [15:37:57 CET] <kierank> speaking of which is there a list of people with commit access [15:38:13 CET] <Daemon404> wm4, c++ involved? [15:38:16 CET] <wm4> (despite --enable-w32threads) [15:38:19 CET] <wm4> no [15:38:28 CET] <Daemon404> mingw runtime may pull it in, i dunno [15:38:32 CET] <Daemon404> im sure they are to blame [15:38:45 CET] <kierank> isn't libwinpthread gpl? [15:40:27 CET] <wm4> hm actually -lrt isn't involved here [15:42:20 CET] <jamrial> wm4: are you using msys2's mingw package? [15:42:36 CET] <jamrial> or a toolchain like the one nevcairiel built? [15:42:39 CET] Action: Daemon404 plays around with mmap [15:43:10 CET] <jamrial> the former has a billion shared libraries and some depend on winpthreads, even if you compile ffmpeg using w32threads [15:43:35 CET] <jamrial> so you're better using a static toolchain like nev's if you want a portable ffmpeg binary [15:46:52 CET] <kierank> Daemon404: are you doing mmap on an fd? [15:46:55 CET] <kierank> and then just seeking? [15:47:03 CET] <kierank> "seeking" [15:47:37 CET] <Daemon404> writing all over the place [15:47:50 CET] <Daemon404> it's just a test program [15:47:52 CET] <kierank> it's the right way to do it [15:48:12 CET] <kierank> I remember virtualdub had a hex editor inbuilt [15:48:58 CET] <Daemon404> this isnt for my compare tool :P [15:49:38 CET] Action: durandal_170 uses dhex [15:50:19 CET] <Daemon404> last time i checked, dhex could not handle comparing files with inserted/deleted bytes [15:50:26 CET] <Daemon404> also it crashed a lot [15:51:07 CET] <durandal_170> fix it, its open source [15:52:34 CET] <Daemon404> it's code is uh... eye-bleeding [15:52:37 CET] <Daemon404> its~ [15:58:17 CET] <durandal_170> so your version is gonna be closed source? [15:58:43 CET] <Daemon404> ... wut? [15:59:53 CET] <durandal_170> Why we don't have dshow wrapper? [16:17:03 CET] <nevcairiel> wm4: as the others have said, its likely a transient dep pulled in from some other mingw crap [16:17:24 CET] <nevcairiel> dont use the msys2 toolchain for portable builds [16:18:48 CET] <wm4> what then [16:19:36 CET] <nevcairiel> use mine if you want, https://files.1f0.de/mingw/ .. or any other static toolchain from around the web which are probably build in a similar manner [17:46:13 CET] <JEEB> michaelni: so should I re-post 3/3 with the updated hash? [18:28:58 CET] <michaelni> JEEB, hmm, probably yes [18:29:05 CET] <JEEB> ok [18:29:22 CET] <JEEB> will try to run the tests at home [18:30:12 CET] <JEEB> also when I have the time I'll have to see if I can get negative cts offsets implemented [18:30:52 CET] <JEEB> i have a poc patch for it but it doesn't work [18:31:59 CET] <michaelni> ok [18:32:54 CET] <JEEB> that way cts = dts for the first sample [18:33:17 CET] <JEEB> which is what cases without edit lists require [18:33:35 CET] <JEEB> (fragmented stuff, dash, mss) [18:34:16 CET] <JEEB> i guess for vod you can have an edit list [18:34:28 CET] <JEEB> live is where you can't, though [18:43:43 CET] <wbs> JEEB: no, you can have an edit list for live just fine [18:43:59 CET] <wbs> it's only in smooth streaming where it isn't actually transmitted at all [18:45:38 CET] <JEEB> wbs: ah ok [18:46:21 CET] <JEEB> DASH livestream with edit lists would be funky I guess [18:54:38 CET] <wbs> not really [18:54:53 CET] <wbs> if you understand the way the isom timestamps work [18:55:02 CET] <JEEB> funky as in "I wonder how many implementations would understand it" [18:55:10 CET] <wbs> ah, yes, that's probably quite true :P [18:55:44 CET] <JEEB> technically it makes sense, if you have edit lists for each fragment or so (because live streams generally don't have a limited duration) [18:55:53 CET] <JEEB> and it would make some stuff heckuva lot simpler [18:56:20 CET] <wbs> it's not per fragment, but just one at the start [18:56:24 CET] <JEEB> hmm [18:56:29 CET] <JEEB> so duration can be eternal or whatever? [18:56:49 CET] <wbs> yes, in 14496-12:2012, there was an amendment that allowed that [18:56:52 CET] <JEEB> cool [18:57:02 CET] <wbs> before that it didn't work, true [18:57:07 CET] <JEEB> that's actually applaudable [18:57:24 CET] <wbs> yes, the 2012 revision fixed all the issues I had wrt streaming [18:57:46 CET] <JEEB> meanwhile DASH IF and other things are seemingly trying to get through without edit lists as far as they can [18:57:47 CET] <wbs> the same thing about non-absolute offsets in tfhd [18:58:06 CET] <wbs> probably because nobody understands the isom timetamp model :P [18:58:15 CET] <JEEB> indeed [19:49:37 CET] <TD-Linux> and no one cares about A/V sync [19:49:58 CET] <TD-Linux> or having an audio track that is the correct length for that matter [19:51:36 CET] <michaelni> TD-Linux, i do care about a/v sync [19:51:52 CET] <fritsch> we also do [19:51:55 CET] <fritsch> :-) [19:52:46 CET] <TD-Linux> unfortunately that also requires cooperation of the people who make the DASH streams :) [19:53:46 CET] <kierank> not many people care about sample level sync like I do [19:55:45 CET] <JEEB> I'm actually not sure how you'd do encoder delay with live DASH with negative CTS offsets... you have the same fragment start timestamp as video, and then all samples have a negative CTS offset and you just have a bit more samples in the fragment? [19:56:08 CET] <JEEB> or do you actually use a fragment DTS that is smaller than the one in video accordingly, and have no CTS offsets [19:57:10 CET] <fritsch> we had the biggest issues with reconfiguring our renderer [19:57:15 CET] <fritsch> on stream changes [19:57:35 CET] <fritsch> to not have visual impact - in kodi we implemented it via a binary addon interface and an inputstream.mpd addon [20:51:46 CET] <D`K_C> Hi guys, is that normal that libavformat/avio.h does not include libavformat/url.h (where struct URLContext is defined)? [20:52:07 CET] <wm4> yes [20:53:37 CET] <D`K_C> what is the reason for that? [20:53:50 CET] <wm4> the URL stuff is private API [20:53:56 CET] <wm4> while avio.h is a public header [21:00:35 CET] <Daemon404> bugs that arent ffmpeg. cool. [21:00:55 CET] <JEEB> lol [21:04:13 CET] <D`K_C> ho right, the application is using the private API... [21:05:10 CET] <D`K_C> thanks wm4 [21:14:23 CET] <cone-127> ffmpeg 03Michael Niedermayer 07master:6e65b9bb1f55: avformat/utils: scan a bit farther for a keyframe in mpeg/mpegts (7 sec instead of 5, we already scan 90sec in some cases by default) [22:44:58 CET] <yashpatel> michaelni: Sir, what is the reason for using these specific 5 contexts as mentioned in 'http://www.ffmpeg.org/~michael/ffv1.html#context'. Are these derived from some experimental values or there goes some theory behind choosing them ? Bcs as in JPEG-LS different contexts are chosen. My question is how to know which contexts to choose ? [23:06:12 CET] <michaelni> yashpatel, the reason is rather boring, "it worked well" [23:07:19 CET] <yashpatel> michaelni: Sir, I've been reading a lot of content regarding the same. Could you please guide me as to how can I improve the algorithm, as mostly things seem to be based on experimental results. [23:08:55 CET] <TD-Linux> doing experiments is pretty much the only way to do it [23:09:13 CET] <TD-Linux> because ffv1 is lossless, the experiments are pretty easy - compress a lot of video and look at the output size [23:09:29 CET] <TD-Linux> you can get a collection of videos to compress here https://media.xiph.org/video/derf/ [23:18:02 CET] <michaelni> yashpatel, if i knew of a certain and simple way to improve it, i would already have improved it. But its certain there are improvments possible, it could b from small changes, from replacing some part by a different better implementation, or from some additional layer that performs further decorrelation, or ... [23:19:40 CET] <michaelni> this problem is rather similar to implementing good P frame compression, that too will need stepwise "improving" and alot of testing various ideas ... [23:25:12 CET] <atomnuker> started doing some experiments for an ffa1 codec: https://github.com/atomnuker/FFmpeg-FFA1 [23:27:46 CET] <J_Darnley> Neat! [23:28:39 CET] <J_Darnley> Wait, was it you talking about over-engineering an audio codec or someone else? [23:29:15 CET] <cehoyos> michaelni: A user has finally uploaded a "motion jpeg 2000" sample meaning concatenated j2k samples. Would a j2k parser be a valid enhancement request? [23:29:28 CET] <cehoyos> ... concatenated j2k frames, sorry [23:39:28 CET] <atomnuker> J_Darnley: might have been me, overengineering stuff is the greatest! [23:39:42 CET] <J_Darnley> :) [23:42:42 CET] <wm4> what would ffa1 have that dozens of lossless codecs don't already have [23:43:47 CET] <nevcairiel> more alien technology powering its algorithms [23:47:22 CET] <atomnuker> will not follow the LPC + golomb/rice residuals convention [23:48:25 CET] <atomnuker> which already makes it dissimilar to almost everything else [23:48:48 CET] <atomnuker> monkey's audio is different to most other codecs but it still sucks [23:49:38 CET] <nevcairiel> the dalaa entropy coding looks extremely convoluted though, not sure thats going to find many people being fans of =p [23:50:12 CET] <atomnuker> I'm a fan of it and that's enough [23:50:55 CET] <atomnuker> if I don't get much better results than flac I'll scrap it [23:52:37 CET] <michaelni> cehoyos, probably yes, possibly the mjpeg parser could be resued, not sure, one would eed to compare the structure [23:53:12 CET] <michaelni> atomnuker, ffa1 <-- very cool idea [00:00:00 CET] --- Tue Mar 22 2016

1 0

ffmpeg.log.20160321
by burek 22 Mar '16

22 Mar '16

[00:06:06 CET] <courrier__> ffmpeg -f concat -i videos.txt -vcodec libx264 -filter:v "setpts=0.04*PTS" -filter:v hqdn3d -an out.mp4 [00:06:26 CET] <courrier__> It looks like setpts is ignored [00:06:34 CET] <courrier__> it's normal speed [00:06:56 CET] <courrier__> is there issues in combining filters perhaps? [00:08:39 CET] <furq> courrier__: -filter:v "setpts=0.04*PTS,hqdn3d" [00:10:32 CET] <courrier__> thx furq :) [00:19:13 CET] <xochilpili> hi all, anyone with mencoder experience? [00:19:23 CET] <xochilpili> im using mencoder to pass a file into a mkv output with subttiles. the issue that i have is that the video changes it's width, i have this "AVFrame.format is not set" in the output: i tried using -vf scale=720:480 but nothing happend, i still have this resolution frame [00:19:30 CET] <xochilpili> can some one please explain me how can i preserve the original aspect? [00:24:46 CET] <DHE> this is ffmpeg, not mplayer/mencoder [08:08:37 CET] <ProfessorOptics> how do i output to multiple files following a naming order, gif01, gif02, gifo3, etc? [08:09:29 CET] <furq> gif%02d.gif [08:12:47 CET] <ProfessorOptics> its just outputting as gif%02d.gif thus not output the all frames. any suggestions? [08:14:34 CET] <furq> oh i guess that doesn't work automatically with gif [08:14:40 CET] <furq> -f image2 gif%02d.gif [08:29:27 CET] <ProfessorOptics> thanks that worked [08:30:07 CET] <ubitux> why would you do that? [08:40:01 CET] <ubitux> ProfessorOptics ^ [12:08:28 CET] <pezet91> hello i have problem with avconv. When I tried to start stream video from pipe (video grabber), avconv can't start :/ [12:09:25 CET] <pezet91> cmd: ./stream /mnt/ux_0 0 | avconv -re -i - -f flv rtmp://localhost/live/0 [12:09:50 CET] <pezet91> it not work :( [12:10:12 CET] <pezet91> can anybody help me? [12:25:10 CET] <JEEB> pezet91: support for avconv goes to #libav [12:26:14 CET] <pezet91> ok, thanks [13:23:55 CET] <d-fens_> hi, can somebody please reproduce: ffmpeg -i "overlaywithalpha.png" -i "1.mp4" -filter_complex " [1:v] setsar=sar=1/1 [sared]; [sared][0:v] overlay [overlay]" -y out.mp4 creates a correct alphablended video, BUT:ffmpeg -i "overlaywithalpha.png" -i "1.mp4" -filter_complex " [1:v] setsar=sar=1/1 [sared]; [sared][0:v] overlay [overlay];[1:v][overlay] concat=n=2" -y out.mp4 creates a wrong alpha overlayed video ; somehow the concat filter [13:23:56 CET] <d-fens_> causes that [14:18:29 CET] <DEBIANIAN_> hi [14:19:31 CET] <DEBIANIAN_> iam trying to stream an audio file using ffserver but i get this error when i send a request . [14:19:55 CET] <DEBIANIAN_> Mon Mar 21 14:15:45 2016 Error writing output header for stream 'test1.ogg': Invalid data found when processing input [14:19:56 CET] <DEBIANIAN_> Mon Mar 21 14:15:45 2016 141.105.162.250 - - [GET] "/test1.ogg HTTP/1.1" 200 68 [14:20:14 CET] Action: DEBIANIAN_ oops! [14:53:50 CET] <ffmpegROOTinstal> hey everybody - I installed my FFmpeg as root in Centos 6.7 and, of course, am getting failures. Does anyone know how to change the PKG_CONFIG command properly so I can install it in /usr/local/bin/ [14:54:53 CET] <thebombzen> ffmpegROOTinstal: try building it from source. the autogenered configure/makefile will go to /usr/local/bin by default. [14:55:23 CET] <thebombzen> the latest Git Master is better than any fixed version anyway. always use git master if possible [14:56:01 CET] <ffmpegROOTinstal> thebombzen: I followed the compilation guide exactly, and now I found a readme telling me not to use $HOME in the path and change it, but diong so gives me ./configure No such file or directory [14:56:41 CET] <thebombzen> !paste [14:56:43 CET] <ffmpegROOTinstal> thebombzen: I also need additional libraries installed for my script, not sure how I would do that with the Git Master [14:57:15 CET] <ffmpegROOTinstal> http://pastebin.com/iEZ3HCeN [14:58:05 CET] <thebombzen> why are you compiling it as root? [14:58:06 CET] <thebombzen> don't do that [14:58:26 CET] <thebombzen> that causes all kinds of linking problems. kinda like the ones you're experiencing [14:58:56 CET] <ffmpegROOTinstal> because I'm a total newb [14:59:31 CET] <thebombzen> ffmpegROOTinstal: linux 101: compile things from source as your normal user. then switch to root before running "make install" [14:59:45 CET] <thebombzen> in general don't do things as root if you can avoid it. because it messes stuff up. [15:00:49 CET] <ffmpegROOTinstal> ok, so do I have to login as new user and go through compilation guide again? [15:01:02 CET] <ffmpegROOTinstal> just created a new user [15:01:04 CET] <thebombzen> no, you use your normal user account. the one without root privileges [15:01:18 CET] <thebombzen> there's no reason to create a new user. [15:01:28 CET] <ffmpegROOTinstal> the only account I have on here is root [15:01:35 CET] <ffmpegROOTinstal> its a new distro deploy [15:02:05 CET] <ffmpegROOTinstal> I know, you're probably wondering why I'm so stupid lol [15:02:29 CET] <thebombzen> no, I'm wondering why you think you don't have a user account. When you set up your CentOS box you should have created a user... right? [15:02:38 CET] <ffmpegROOTinstal> no, only root [15:02:54 CET] <ffmpegROOTinstal> but I just created one [15:03:25 CET] <ffmpegROOTinstal> do I need to assign any priviledges? [15:04:19 CET] <thebombzen> if you're brand new to Linux I'd recommend a different distribution. CentOS is not targeted at End Users [15:04:39 CET] <ffmpegROOTinstal> I know! It's freaking crazy [15:04:47 CET] <ffmpegROOTinstal> however, I wrote a script in it a couple of years ago [15:04:55 CET] <ffmpegROOTinstal> its very lucrative and it took a long time to run [15:05:00 CET] <ffmpegROOTinstal> I'm willing to pay for your help right now [15:05:04 CET] <ffmpegROOTinstal> not looking for handouts [15:05:18 CET] <ffmpegROOTinstal> if you have a paypal I can send [15:05:25 CET] <thebombzen> ... [15:05:37 CET] <ffmpegROOTinstal> wasn't trying to insult you? [15:05:42 CET] <thebombzen> if you wrote a script that runs on centos then it should run on other distributions [15:05:56 CET] <ffmpegROOTinstal> yep, but I installed ffmpeg correctly last time [15:06:09 CET] <ffmpegROOTinstal> so this time I screwed up and am looking for guidance on how to unscrew it up [15:06:35 CET] <thebombzen> centos is an enterprise distribution. it's designed to be used for servers and that kind of thing. please, do yourself a favor and get a beginners' distribution [15:06:50 CET] <ffmpegROOTinstal> once FFmpeg is installed I don't have to use it [15:07:24 CET] <ffmpegROOTinstal> Anyone out there want a consulting gig on this problem? I installed FFmpeg in /root/ with root@ user and need someone to help me get it in /usr/local/bin/ [15:07:33 CET] <thebombzen> either way, you clearly didn't read the compliation guide. If you had read it, you'd know that --prefix is used with cross compiles. [15:08:27 CET] <ffmpegROOTinstal> like I said, I screwed up [15:08:37 CET] <ffmpegROOTinstal> oh [15:08:47 CET] <ffmpegROOTinstal> so can you help me unscrew it up? [15:09:34 CET] <thebombzen> yes. here's my first bit of advice. when you go to the internet and say you've read the manual, please read the manual if you're going to say that. [15:09:46 CET] <ffmpegROOTinstal> Ok, I read the manual. [15:09:55 CET] <ffmpegROOTinstal> Every single line of the compilation guide [15:10:14 CET] <ffmpegROOTinstal> That doesn't mean I absorbed and was able to digest everything [15:10:30 CET] <ffmpegROOTinstal> I don't know how many times I can take the blame for this problem - it's my fault [15:14:28 CET] <thebombzen> here's a clue [15:14:32 CET] <thebombzen> ./configure [15:14:33 CET] <thebombzen> make [15:14:35 CET] <thebombzen> make install [15:15:52 CET] <ffmpegROOTinstal> same error [15:26:39 CET] <satinder___> how we can use reload tag for drawtext please any one can give any example [15:26:55 CET] <satinder___> please humble request [15:28:01 CET] <J_Darnley> RTFM [15:28:23 CET] <J_Darnley> reload [15:28:23 CET] <J_Darnley> If set to 1, the textfile will be reloaded before each frame. Be sure to update it atomically, or it may be read partially, or even fail. [15:30:17 CET] <satinder___> J_Darnley : Respected Sir , I already read this line but I not able to find a suitable example , I mean I am not able to use it [15:30:36 CET] <J_Darnley> Set it to 1 then write to the file! [15:30:49 CET] <J_Darnley> What don't you get about that? [15:31:04 CET] <satinder___> Please give any example , I want overlay a variable which will change in every second [15:31:17 CET] <satinder___> so I want I can do for that [15:31:22 CET] <J_Darnley> Then you need to write to the file every second. [15:31:28 CET] <satinder___> ok [15:31:33 CET] <satinder___> you mean textfile [15:37:22 CET] <satinder___> sir thanks for giving hint [15:37:27 CET] <satinder___> I will try it [15:48:54 CET] <satinder___> J_Darnley : Thanks Sir that working for me perfectly [15:49:37 CET] <satinder___> Please you can give some Idea how we can overlay new line [15:50:35 CET] <satinder___> I want more than two lines overlay [15:56:29 CET] <Gp1> Hey, I got a question, might not be entirely relatef t [15:57:56 CET] <Gp1> related to ffmpeg, I got a video codec as .ax file, i understand its a part of directshow graph, can i incorporate it in ffmpeg? [15:58:44 CET] <durandal_170> only if you RE it [15:59:04 CET] <durandal_170> what's codec name? [15:59:56 CET] <satinder___> durandal_170 : sir I happy see you again , please read my above question sir give some opinion [16:00:33 CET] <durandal_170> I'm paid per hour [16:00:55 CET] <Gp1> Elecard [16:01:17 CET] <satinder___> :( [16:03:52 CET] <satinder___> please help [16:04:53 CET] <J_Darnley> What was your question? How do you use a new line? [16:05:07 CET] <J_Darnley> By having a new line in the text file. [16:06:56 CET] <durandal_170> Gp1: thats company name [16:09:10 CET] <Gp1> Yeah, the codec itself is eavcdec_16k.ax [16:23:47 CET] <Gp1> Durandal_170? [16:24:53 CET] <durandal_170> Gp1: is it for AVC? [16:25:07 CET] <Gp1> Yes [16:25:53 CET] <durandal_170> as I already said its not possible [16:25:54 CET] <satinder___> J_Darnley : Sir I want new line in textfile , [16:26:04 CET] <satinder___> Now I did it sir [16:26:14 CET] <satinder___> only because of you [16:26:18 CET] <satinder___> thanks [16:31:00 CET] <satinder___> J_Darnley : I use echo command in system() call for write data in textfile , Is that right aproch ?? [17:20:12 CET] <c_14> no, that's not atomic [17:20:15 CET] <c_14> use rename [17:20:34 CET] <c_14> i.e. write the new text to a new file and rename it over the old one [19:18:13 CET] <thebombzen> <_< when people bite off more than they can chew. >my bash script don't work. >do you know bash. >no should i lern it [19:36:06 CET] <llogan> thebombzen: i assume "doesn't work" was the only so-called error message supplied instead of showing the script and actual console output. [19:38:10 CET] <thebombzen> yep. no description of what it was supposed to do and what it did instead. [19:38:42 CET] <thebombzen> when you start developing your own stuff your bug reports to other projects tend to get much better [19:43:37 CET] <rrva> how can I add a adaptation field with discontinuity indicator in an existing mpeg-ts file? [19:52:47 CET] <nadermx> Does the -bufsize option work when converting to mp3? Or is that only for video. [19:53:35 CET] <JEEB> it works with whatever that supports that option [19:53:42 CET] <JEEB> most probably LAME doesn\t [19:56:06 CET] <nadermx> Is LAME the best option for encoding to MP3? [19:56:36 CET] <JEEB> yes [19:56:45 CET] <JEEB> also I think if you set a bit rate it will encode in CBR mode anyways [19:56:50 CET] <JEEB> not much leeway for any bufsize [19:56:51 CET] <nadermx> I keep running into issues where when I have a input url that is over 30 or 40 mins it cuts out the song [19:57:28 CET] <nadermx> Been trying to figure out how to do fix it, but can't seem to make any progress [19:59:48 CET] <jookiyaya> does ffmpeg support -input that is DVD/Bluray instead of single file [20:15:01 CET] <llogan> jookiyaya: there is this: http://ffmpeg.org/ffmpeg-protocols.html#bluray [20:15:07 CET] <llogan> don't know anything about it though [20:15:57 CET] <RobertKm> Can anyone tell me how to how to use ffmpeg zoom pan in x11grab [20:16:15 CET] <llogan> jookiyaya: requires ffmpeg with --enable-libbluray [20:16:44 CET] <jookiyaya> llogan is it enabled for normal build [20:17:23 CET] <llogan> i don't understand what you mean [20:19:05 CET] <jookiyaya> llogan is it enabled in this: https://ffmpeg.zeranoe.com/builds/win64/static/ffmpeg-3.0-win64-static.7z [20:20:32 CET] <llogan> run it. look for --enable-libblueay in the output [20:21:27 CET] <llogan> *libbluray [20:21:49 CET] <jookiyaya> Unrecognized option '-enable-libbluray'. [20:22:04 CET] <llogan> no. run just "ffmpeg" then look at output from command. [20:22:47 CET] <jookiyaya> yes it's there [20:23:15 CET] <llogan> then it should support the protocol that i linked to. [20:23:25 CET] <jookiyaya> ok thanks [20:23:35 CET] <llogan> however, i don't actually know what the protocol does [20:23:55 CET] <llogan> and i doubt it works on anything encryted [20:24:10 CET] <jookiyaya> what about dvd then? [20:24:34 CET] <jookiyaya> which lib is for dvd [20:24:59 CET] <llogan> ffmpeg has no dvd input support, but it can decode unencrypted vob files [20:25:25 CET] <jookiyaya> wow really? [20:25:36 CET] <jookiyaya> it has bluray support but no dvd support [20:25:49 CET] <llogan> (contrary to what apple thinks who is apparently rejecting some apps that use ffmpeg because they believe ffmpeg can decode encrypted shit) [20:26:42 CET] <kepstin> well, ffmpeg can decode some encrypted stuff (certain streaming protocols mostly) - if you provide it with keys that you've gotten from some external source. [20:27:00 CET] <llogan> i was referring to dvd specifically [20:27:04 CET] <jookiyaya> why can't ffmpeg support dvd ? [20:27:19 CET] <kepstin> but really, a libdvd input to ffmpeg would probably be a useful and reasonable addition to ffmpeg [20:27:28 CET] <kepstin> libdvdnav/libdvdread [20:27:42 CET] <jookiyaya> then why isn't it already added? [20:28:03 CET] <llogan> nobody submitted a patch to do so [20:28:13 CET] <llogan> as far as I know [20:28:37 CET] <jookiyaya> kepstin does libdvdnav/libdvdread support encrypted dvd or non-ecrypted only? [20:29:08 CET] <kepstin> jookiyaya: by themselves non-encrypted, but you can optionally build them with libdvdcss support to crack encrypted disks [20:29:45 CET] <jookiyaya> what about libbluray [20:32:18 CET] <kepstin> libbluray can use libaacs or libbdplus, i think, which are able to decrypt blurays if you provide keys that you've obtained... somehow else (it can't do it on its own - it's not like dvd where the encryption has trivial to break weaknesses) [20:36:11 CET] <Carl__> could anyone tell me what this error means? http://pastebin.com/HgHrjVkH [20:36:24 CET] <jookiyaya> how come libdvdcss does not key file but libaacs does [20:37:21 CET] <c_14> because dvd encryption is trivial to break [20:37:48 CET] <kepstin> libdvdcss doesn't need keys because it uses weaknesses in dvd encryption to break it (in some cases it even brute-forces keys, which can take only a few seconds on modern computers) [20:38:14 CET] <llogan> Carl__: is that from ffmpeg? [20:38:41 CET] <jookiyaya> how long does it take to brute force aacs keys [20:38:48 CET] <Carl__> yes [20:39:34 CET] <Carl__> ret = avformat_write_header( outContext, NULL ); if (ret != 0) { av_log( NULL, AV_LOG_ERROR, "Failed to connect to RTSP server for '%s'.\n", url ); goto end; } [20:40:40 CET] <kepstin> jookiyaya: it uses 128bit aes keys, do the math yourself :/ [20:41:18 CET] <jookiyaya> i noticed KEYDB.cfg has like list of all the bluray titles [20:41:19 CET] <kepstin> jookiyaya: pretty much all existing known keys are stolen or extracted from memory images of legally licensed players, not brute-forced. [20:41:51 CET] <jookiyaya> kepstin i see [20:42:12 CET] <jookiyaya> so if you have a bluray title that is not listed in keydb.cfg, then it won't work [20:42:59 CET] <jookiyaya> ? [20:43:10 CET] <kepstin> and most newer disks also use bd+ protection to do some scrambling in addition to the aacs [20:43:42 CET] <kepstin> and libbdplus only supports some older versions of that - and requires some additional key files and player memory images too, I think. [20:44:11 CET] <jookiyaya> i came across a bluray that cannot be played [20:44:26 CET] <jookiyaya> even with keydb.cfg/liaacs.dll [20:44:28 CET] <klaxa> i came across a bluray that i could play [20:44:37 CET] <klaxa> it's the only one that worked [20:44:41 CET] <kepstin> klaxa: wow, i'm impressed. those are pretty rare ;) [20:45:00 CET] <klaxa> took me two days or something, hadn't heard of any bluray drm before that [20:45:18 CET] <jookiyaya> kepstin why is that rare [20:45:49 CET] <kepstin> of course, if you actually get past the encryption of the bd, then the files on it are all just plain mpeg-ts which you can use as input to ffmpeg directly [20:45:54 CET] <klaxa> only worked with one disc i have, the other two i tried didn't [20:46:47 CET] <kepstin> most of the disks that are supported by the open-source libraries and have keys available are fairly old. if you just pick up any new disk at a store, chances are it's probably not going to work. [20:47:57 CET] <Carl__> anyone? [20:47:59 CET] <jookiyaya> then how does player with non-opensource able to playback bluray [20:50:11 CET] <kepstin> jookiyaya: ask the people who make the player [20:50:17 CET] <kepstin> (hint: they won't tell you) [20:50:51 CET] <jookiyaya> why can't libbluray use same method that regular bluray player use [20:51:39 CET] <kepstin> jookiyaya: because doing it "the proper way" requires having a secret key that you have to keep secret [20:51:39 CET] <klaxa> because bluray player manufacturers pay good money for licenses [20:52:14 CET] <klaxa> isn't it great how you can make money at every step in the chain? :) [20:54:00 CET] <kepstin> a lot of it actually relies on the firmware of the drive itself - the drive reads every disk inserted and updates the allowed/blacklisted player keys with info from the disk. The drive is also responsible for authenticating the player software and providing some key material off a hidden section of the disk. [20:54:33 CET] <kepstin> it's interesting, because inserting a newly purchased disk can actually cause your bluray player to stop working on all disks until you get an update with a new key [20:55:24 CET] <jookiyaya> are you saying only online bluray players work properly [20:55:57 CET] <jookiyaya> what if you have don't connect internet to bluray player [20:56:02 CET] <klaxa> with the information kepstin just provided i would believe that no bluray player works "properly" [20:56:23 CET] <kepstin> only blu-ray players with an internet connection so they can receive firmware updates over the course of their life will be able to keep operating, yes. [20:56:48 CET] <klaxa> or the company decides to become evil and blocks all blurays for all time [20:56:53 CET] <kepstin> (I know that some models of a few older players that are basically useless now, since the manufacturer stopped releasing firmware updates for them) [20:57:24 CET] <jookiyaya> if that is true, then there would a be a lot of complaints [20:57:28 CET] <jookiyaya> not everybody has internet [20:58:09 CET] <jookiyaya> i know people that as no cable/internet and just watch dvd/bluray [20:58:32 CET] <kepstin> i'm pretty sure there are a lot of complaints :) [20:58:54 CET] <kepstin> in some cases, you might be able to take a player back to a store for "servicing" and they'll update the fw for you [20:59:39 CET] <jookiyaya> i don't understand why a bluray player has to be updated to work properly? doesn't blruay have a standard [21:00:54 CET] <kepstin> the standard includes provisions for blacklisting stolen player keys and revising processing key versions, and requires that players have support for updating firmware. [21:01:14 CET] <kepstin> and there's actually been several standards adding new copy protection features after the original standard [21:01:20 CET] <kepstin> BD+ in particular [21:02:46 CET] <jookiyaya> at least not all bluray has bd+ [23:12:25 CET] <blue_misfit> hey guys, what's involved in getting multithreaded decoding out of libavcodec? [23:17:36 CET] <durandal_1707> blue_misfit: it should be auto enabled [23:18:10 CET] <blue_misfit> durandal_1707, thanks - that's what I thought [23:18:55 CET] <durandal_1707> there are threads options [23:28:14 CET] <kepstin> the threads options are mostly useful to reduce threads if you're running multiple things on one system. [00:00:00 CET] --- Tue Mar 22 2016

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