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July 2015

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ffmpeg-devel.log.20150725
by burek 26 Jul '15

26 Jul '15

[00:28:27 CEST] <jamrial> i seriously wonder what happened with libdcadec's developer [00:28:33 CEST] <jamrial> it's been almost two months since he was last active [00:28:38 CEST] <cone-609> ffmpeg 03Michael Niedermayer 07master:dee551bbd280: avcodec/dvdec: skip 3rd stage ac decoding when the headers indicates that the data is inconsistent [00:30:10 CEST] <wm4> jamrial: all while he promised to make a release soon...? [00:30:46 CEST] <jamrial> yeah, he was fixing some bugs and was about to tag a first release. he even added placeholder version defines and all [00:33:03 CEST] <rcombs> isn't the standard procedure at this point to have everything you need for your own application finished, get bored, and abandon the project? [00:33:58 CEST] <wm4> ffmpeg should adopt it by copying it into its source tree [00:34:07 CEST] <wm4> (50% joking, 50% serious) [00:35:19 CEST] <rcombs> it's already LGPL2.1+ [00:38:12 CEST] <jamrial> wm4: actually that'd be great [00:38:40 CEST] <jamrial> lgpl, better than current ffdca [00:39:12 CEST] <wm4> ffdca is gpl? [00:40:10 CEST] <wm4> it's lgpl too [00:40:54 CEST] <wm4> so I guess you mean libdcadec doesn't have a worse license [00:40:55 CEST] <jamrial> i mean that it can easily be added to ffmpeg's tree to replace the current decoder [00:41:03 CEST] <jamrial> yes [00:41:24 CEST] <wm4> so we'd have 2 dca decoders [00:43:39 CEST] <jamrial> keyword replace [00:46:24 CEST] <wm4> but ffmpeg isn't goimg to delete these valuable speed optimizations in ffdca [00:48:17 CEST] <jamrial> in exchange of getting actual lossless dca decoding and actual maintainable non-spaghetti code, why not? [00:48:22 CEST] <jamrial> it's not like new asm couldn't be written [00:58:06 CEST] <wm4> I'm not the one who you need to convince [00:58:28 CEST] <wm4> I just firnly believe ffmpeg never deletes anything, not even useless code [01:03:47 CEST] <jamrial> libmpcodecs was deleted in the end, right? so not always :p [01:04:34 CEST] <cone-609> ffmpeg 03Michael Niedermayer 07n2.6.4:HEAD: avcodec/dvdec: skip 3rd stage ac decoding when the headers indicates that the data is inconsistent [01:11:23 CEST] <wm4> it took only half a decade [01:25:26 CEST] <philipl> wm4: send a patch. :-) [01:38:24 CEST] <wm4> for libdcadec? we don't know about the author yet [01:50:17 CEST] <philipl> wm4: an attempted fork can often provoke a lazy author into responding. [02:42:32 CEST] <cone-609> ffmpeg 03Zhang Rui 07master:c0a4af408ee5: avformat/async: move more code into locked area in background thread [02:42:33 CEST] <cone-609> ffmpeg 03Zhang Rui 07master:8a173351895e: avformat/async: wake up main thread before exit background thread [05:23:37 CEST] <cone-609> ffmpeg 03Michael Niedermayer 07master:1909a9151cce: swscale/output: Fix "warning: assignment from incompatible pointer type" [08:30:43 CEST] <nevcairiel> its not like the dcadec author slowly became less active, he just up and vanished one day [08:30:52 CEST] <nevcairiel> at first i thought, maybe he went on vacation [08:30:55 CEST] <nevcairiel> but two months is long [09:19:44 CEST] <durandal_1707> you .... [11:56:41 CEST] <durandal_1707> anybody working on subtitle support in lavfi? [12:29:22 CEST] <durandal_1707> hmm I will add overlap option for acrossfade [12:49:21 CEST] <rcombs> durandal_1707: *points vaguely in the direction of ubitux* [13:07:20 CEST] <kierank> https://github.com/kaltura/platform-install-packages/issues/392 [13:07:31 CEST] <kierank> lots of linkedin stalking after this [13:10:35 CEST] <durandal_1707> what they want? [13:11:18 CEST] <kierank> they are not happy that I am telling them they are violating the gpl and distributing very expensive software with non-free ffmpeg [13:12:44 CEST] <JEEB> lul [13:13:01 CEST] <BBB> kierank: youre too nice [13:13:07 CEST] <BBB> they should always be added to shame list by default [13:13:10 CEST] <BBB> adding/removing takes seconds [13:13:15 CEST] <BBB> talking to lawyers takes years [13:13:43 CEST] <kierank> various VPs of whatever have been stalking me on linkedin as a result [13:16:08 CEST] <durandal_1707> but we no longer have wall of shame just bug tracker [13:16:19 CEST] <Compn> its all fun and games until some company sues for defamation and loss of revenue due to hall of shame [13:16:29 CEST] <Compn> which is why i disabled the shame page [13:17:34 CEST] <durandal_1707> what's point of bugs on tracker if they never get resolved? [13:18:19 CEST] <Compn> libav removed it in 2014 it seems, https://lists.libav.org/pipermail/libav-commits/2014-September/015362.html [13:18:19 CEST] <iive> Compn: huh? [13:18:23 CEST] <durandal_1707> the ones with license violations [13:18:59 CEST] <Compn> durandal_1707 : waiting for some volunteer to step up [13:19:24 CEST] <Compn> to educate and fix companies problematic distribution [13:19:39 CEST] <durandal_1707> you mean someone to sue them? [13:20:42 CEST] <Compn> i find its easier to talk than to sue [13:20:46 CEST] <Compn> iive : https://ffmpeg.org/shame.html [13:20:52 CEST] <durandal_1707> such companies do not care [13:22:00 CEST] <iive> Compn: oh, you mean that the page might not be up to date, and because of that it would be defamation. [13:22:19 CEST] <iive> not that shaming is defamation. [13:22:27 CEST] <Compn> yes, or maybe an entry was put there in error [13:23:09 CEST] <Compn> or maybe its a valid entry, it still can take 10 years nonsense to move through courts. even with frivolous lawsuit (in usa anyway) [13:24:33 CEST] <Compn> durandal_1707 : https://ffmpeg.org/pipermail/ffmpeg-devel/2014-July/160126.html [13:24:42 CEST] <Compn> there were some that did [13:27:32 CEST] <Compn> is that violating lgpl or fdk-aac though? [13:28:14 CEST] <Compn> if they have lgpl source available, i would say they are only violating fdk-aac license ? [13:28:14 CEST] <kierank> lgpl [13:28:36 CEST] <nevcairiel> the fdk-aac license is in general quite liberal [13:28:40 CEST] <nevcairiel> its just not gpl compatible [13:28:44 CEST] <kierank> they have --enable-gpl enabled Compn [13:29:09 CEST] <Compn> do they provide ffmpeg source y/n ? [13:29:10 CEST] <nevcairiel> gpl and nonfree are exclusive though, so only one of them takes hold [13:29:41 CEST] <kierank> then they build nonfree [13:29:44 CEST] <kierank> and distribute [13:30:19 CEST] <nevcairiel> they download official ffmpeg relesae sources from ffmpeg.org in their build script [13:30:28 CEST] <Compn> see, if they provide ffmpeg source, i dont care how they compile it or what nonfree libs they use. [13:30:39 CEST] <nevcairiel> so in a way, they provide sources =p [13:30:43 CEST] Action: durandal_1707 waiting for s64 sample format [13:30:45 CEST] <Compn> but i guess thats just me. [13:31:38 CEST] <nevcairiel> kierank: do they actually distribute binaries, that github project reads like its meant for me to checkout, and it'll build a binary for me locally then [13:31:43 CEST] <nevcairiel> which would be fine license wise [13:31:56 CEST] <JEEB> I think they had repositories [13:31:59 CEST] <kierank> it's the buildscripts presumably that they use in their commercial product too [13:38:02 CEST] <nevcairiel> of course the lack of a decent aac encoder outside of fdk-aac screws with everyone [13:38:39 CEST] <JEEB> thankfully a lot of stuff can start moving towards opus [13:38:57 CEST] <nevcairiel> i was meaning to investigate that [13:39:13 CEST] <JEEB> of course it would mean including an opus decoder in your app on mobile depending on the OS [13:39:16 CEST] <nevcairiel> but since my streaming mostly targets android, i might need to ship libopus for compat [13:39:31 CEST] <nevcairiel> luckily, ExoPlayer already did the integration recently :D [13:39:38 CEST] <JEEB> :D [13:40:09 CEST] <JEEB> yeah, it really depends on how much time it'd take to support opus in addition, because some apps just use the system API calls, and thus including a new decoder would mean quite a bit of code [13:40:39 CEST] <nevcairiel> yeah we switched to ExoPlayer some time ago because the built-in mediaplayer is dumb as fuck for streaming, and even worse, behaves differently on every device [13:40:49 CEST] <nevcairiel> so a player we can ship and therefor control is just brilliant [13:41:21 CEST] <nevcairiel> and its format support grows every day, so yay [13:54:04 CEST] <BBB> j-b: poke [14:05:30 CEST] <wm4> * durandal_1707 waiting for s64 sample format <- I'd rather have a 24 bit sample format [14:05:38 CEST] <wm4> since these are actually used and important [14:05:50 CEST] <nevcairiel> i dont mind how it works today [14:06:03 CEST] <nevcairiel> 24-bit values are just annoying to handle [14:07:37 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:17cc35c76b92: avcodec/aactab: Fix rounding of elements in ff_aac_eld_window_512_fixed [14:07:38 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:f267d553f76f: avcodec/aactab: Add ff_aac_eld_window_480_fixed [14:08:50 CEST] <wm4> nevcairiel: so what they can just be converted to 32 bit if you really can't process them [14:09:23 CEST] <nevcairiel> sure, extra 0s dont hurt the audio [14:09:28 CEST] <nevcairiel> and if you want to handle it, you can [14:26:05 CEST] <nardev> guys, i need somebody to write me commands for converting videos i'm prepared to pay that. it should be able to process feew different formats and convert videos to a few most popular formats.. nothing which would not work out of box in ffmpeg.. and maybe some tweaking about processor usage and quality but that second part is less important. Anyone interested? [14:50:20 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:bbad3811cc86: avcodec/aacps: Fix ;; [14:57:33 CEST] <durandal_1707> kierank: you were mentioning smptebars the other day.. [14:57:49 CEST] <kierank> yeah I need to take some photos of the scope at work [15:07:16 CEST] <durandal_1707> you mean some values are out of range? [15:25:00 CEST] <kierank> some of the colours are too weak [15:25:05 CEST] <kierank> don't know why [15:25:13 CEST] <kierank> because the numbers are from the spec [15:28:24 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:38490e07247e: avcodec/dvdec: only attempt to conceal errors based on STA inconsistencies when error_concealment is set [17:48:47 CEST] <cone-201> ffmpeg 03Ivan Uskov 07master:6d0123f40e2a: avcodec: Add QSV MPEG-2 video decoder. [17:58:04 CEST] <durandal_1707> nlmeans is very good denoiser? [18:07:52 CEST] <ubitux> durandal_1707: i started implementing it [18:08:10 CEST] <ubitux> i'm sorry i havent made much progress lately [18:09:00 CEST] <durandal_1707> nlmeans? [18:09:06 CEST] <ubitux> yes [18:09:45 CEST] <durandal_1707> there are already gpl implementations available... [18:10:07 CEST] <ubitux> i like the way it works [18:10:21 CEST] <ubitux> so i would like to implement it [18:10:41 CEST] <durandal_1707> then go ahead [18:11:00 CEST] <ubitux> sure, i'll go back to it as soon as i'm done with selectivecolor [18:11:29 CEST] <ubitux> i'm trying to figure out the logic behind the black spining adjustment [18:11:29 CEST] <durandal_1707> there is also this nl filter from pnm on tracker [18:12:40 CEST] <durandal_1707> I was planning to add simple subtitle support in lavfi with sshowinfo [18:13:46 CEST] <ubitux> how are you going to do that without having subtitles structure in libavutil? [18:14:18 CEST] <durandal_1707> why one would need that? [18:14:18 CEST] <ubitux> mangled avcodec dependency all over? [18:14:37 CEST] <ubitux> well, AVSubtitle structures are defined in libavcodec [18:15:25 CEST] <durandal_1707> Can just move them? [18:18:15 CEST] <ubitux> that's one solution [18:18:34 CEST] <ubitux> but not that simple [18:19:26 CEST] <ubitux> i think i had a patch submitted for that but i wasn't satisfied with it... [18:21:01 CEST] <durandal_1707> people still use -deinterlace, I hope michaelni will remove it on next major bump [18:22:16 CEST] <durandal_1707> ubitux: so how much % is selectivecolor complete? [18:22:58 CEST] <ubitux> technically, it could be made upstreamable with just the addition of the doc [18:23:09 CEST] <ubitux> but complete i don't know [18:23:28 CEST] <ubitux> the last bit that needs work is to figure out the way the black spinning works [18:24:03 CEST] <ubitux> i thought it was a simple addition to all the 3 other spinning parameters, but it seems it isn't [18:24:24 CEST] <ubitux> i need to do more graph & shit, but i find PS a PITA to use [18:26:36 CEST] <ubitux> so basically CMY adjustment seems to match PS implementation, but the K parameter is still kind of stronger or something [18:27:01 CEST] <durandal_1707> graphs? what kind of graphs? [18:27:45 CEST] <ubitux> graph of data i observe in photoshop while tweaking the filter parameters :p [18:30:31 CEST] <durandal_1707> What about writting GUI for ffmpeg filtergraph creation in python? [18:30:32 CEST] <ubitux> durandal_1707: shit like this http://ubitux.fr/pub/pics/_selectivecolor-data.png [18:30:43 CEST] <ubitux> durandal_1707: would be nice. [18:31:33 CEST] <ubitux> i'd love to see a GUI for ffmpeg filtergraph [18:32:40 CEST] <ubitux> you could even make it web [18:32:55 CEST] <ubitux> to share your graphs online ;) [18:44:41 CEST] <nardev> guys, i need somebody to write me commands for converting videos i'm prepared to pay that. it should be able to process feew different formats and convert videos to a few most popular formats.. nothing which would not work out of box in ffmpeg.. and maybe some tweaking about processor usage and quality but that second part is less important. Anyone interested? [19:07:44 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:7ec5115409ed: avformat/nutenc: Omit AV_PKT_DATA_QUALITY_STATS when storing side data. [19:15:13 CEST] <kierank> jamrial: I guess you didn't use avx 3-operand because it's float? [19:17:57 CEST] <jamrial> the aacps patch? no, i just wanted to put all movas together and then the add/sub/mul intructions, rather than having them interleaved like x86inc generates when it converts 3 operand into 2 operand [19:18:46 CEST] <jamrial> most of these are sse so they will be able to run on really old cpus, where order of instructions is more important [20:02:31 CEST] <jamrial> huh, there's a qsv mpeg2 decoder committed to libav that's almost the same as ours but with a different author [20:03:06 CEST] <jamrial> and neither says "based on work by" or such [20:03:29 CEST] <BtbN> Is it possible that two people actualy wrote them independendly? [20:06:01 CEST] <jamrial> could be [20:06:49 CEST] <JEEB> looking at the base file they're seemingly quite different-looking. the timing is quite perfect though :P [20:06:53 CEST] <jamrial> admitedly, libav's seems more complete now that i look them more closely [20:06:55 CEST] <jamrial> yeah [20:08:22 CEST] <jamrial> guess the name, avoptions and configure dependencies being the same gave me the impression they were more similar than they actually are [20:10:08 CEST] <cone-201> ffmpeg 03Ivan Uskov 07master:fb57bc6c34b9: avcodec: Add QSV VC-1 video decoder. [20:11:14 CEST] <BtbN> Now if only QSV was actualy a widely available package on distributions, ffmpeg would have quite decent and widely available hardware acceleration support. [20:12:48 CEST] <BBB> nardev: you might have more success emailing the list, irc is very fleeting in nature so it might not work very well to ask here [20:12:59 CEST] <BBB> (not sure I like the idea of everyone emailing ffmpeg-devel, but ohwell( [20:14:05 CEST] <nardev> BBB, i already sent an email there thank you [20:27:45 CEST] <kierank> ohthisoneagain [20:27:46 CEST] <kierank> https://news.ycombinator.com/item?id=9948041 [20:28:42 CEST] <nevcairiel> hacking up the bsf even more for this silly qsv decoder seems so stupid [20:28:45 CEST] <nevcairiel> but noone listens to me [20:33:16 CEST] <wm4> that gentoo post lol [20:33:25 CEST] <wm4> users really have no fucking clue and blame Libav for it [20:33:29 CEST] <JEEB> yes [20:33:46 CEST] <JEEB> I bet most of the "didn't serve my needs" people actually had an issue with the oldness of the packaged version [20:33:48 CEST] <wm4> so gentoo packages old age libav -> it causes problems -> install a recent ffmpeg -> it works -> blame libav! [20:34:06 CEST] <JEEB> same for debian-based systems where version 0.8 was there for a long time [20:34:15 CEST] <wm4> oh and someone linked my new troll writeup [20:34:25 CEST] <JEEB> and then they grabbed a static binary from one of the places linked @ #ffmpeg [20:34:30 CEST] <JEEB> and suddenly shit worked [20:35:09 CEST] <JEEB> thus=>libav rather than old libav was the issue (and we totally would've not had these issues with ffmpeg when gentoo/debian/whatever had to update the stuff it was packaging) [20:35:15 CEST] <JEEB> but what can you do, really :D [20:35:20 CEST] <JEEB> lolperception [20:35:29 CEST] <JEEB> of course then there are actual things that are only in ffmpeg [20:35:33 CEST] <JEEB> like the concat vf [20:35:35 CEST] <JEEB> etc [20:35:57 CEST] <wm4> yes, it does have some more features [20:36:11 CEST] <wm4> and libav has some "bugs" ffmpeg doesn't have [20:36:45 CEST] <wm4> (I'd claim many are issues with slightly broken or obscure files, where often a 1-line patch helps) [20:37:21 CEST] <jamrial> wm4: only ffmpeg versions gentoo users can use without unmasking "unstable" releases are the "equivalent" to whatever libav version is also available [20:37:44 CEST] <jamrial> so it's not like they replaced libav 0.8 for ffmpeg 2.6 and they magically got modern features [20:38:09 CEST] <JEEB> yeah, with gentoo that is true. unless they happened to do those two things. you get around 1.0 of ffmpeg, right? [20:38:15 CEST] <JEEB> for libav 9 or whatever it was [20:38:32 CEST] <JEEB> but with debian it usually went so that the person grabbed a new static build [20:38:41 CEST] <jamrial> 1.2 i think [20:39:10 CEST] <BtbN> current stable is ffmpeg 2.6.3 and libav 11.3 [20:39:36 CEST] <JEEB> yeah, but there were a lot of birth pains [20:39:42 CEST] <jamrial> yeah. it was 2.2 until recenty, and 1.2 before that [20:39:52 CEST] <JEEB> due to packages that nobody wanted to update to newer APIs and all [20:40:10 CEST] <jamrial> on libav side of things, 9 and 0.8 respectively [20:40:11 CEST] <JEEB> I think I got one of the gentoo maintainers doing the work to ragequit a channel once when asking about these things (why is the thing so old) [20:40:49 CEST] <JEEB> I know at least gentoo did a whole lot of work trying to update everything, even if the usage rate of that thing wasn't too high [20:40:52 CEST] <JEEB> uhh [20:40:56 CEST] <JEEB> I meant debian there [20:41:03 CEST] <JEEB> although gentoo probably had some work to be done [20:41:05 CEST] <jahnyarrow> hi - i saw the news item about needing a new host - do you still need this? [20:41:05 CEST] <JEEB> as well [20:41:28 CEST] <JEEB> jahnyarrow: join the mailing list discussion, I think various alternatives are still being discussed there [21:57:33 CEST] <philipl> nevcairiel: Why aren't the qsv decoders done as proper hwaccels? [21:57:50 CEST] <nevcairiel> because [21:57:52 CEST] <philipl> What do the stub hwaccel entries with no methods even mean? [21:58:02 CEST] <BtbN> because that would mean touching libva. [21:58:09 CEST] <philipl> would it? [21:58:13 CEST] <philipl> I guess so. [21:58:20 CEST] <BtbN> at least on linux [21:58:41 CEST] <philipl> __gb__ didn't answer when I asked if anyone from Intel would be adding new vaapi hwaccels for the missing formats [21:58:51 CEST] <philipl> I suppose this QSV work is the answer. [21:58:53 CEST] <nevcairiel> the main reason is that because ffmpeg.c and other lazy applications couldnt use it easily [21:58:59 CEST] <nevcairiel> hwaccels require a tad bit of effort [21:59:08 CEST] <nevcairiel> easier to provide dumb codec wrappers with reduced functionality [21:59:10 CEST] <philipl> Yeah. Someone could write a vaapi adapter for ffmpeg.c [21:59:18 CEST] <BtbN> i tried that already [21:59:23 CEST] <philipl> Heh. [21:59:26 CEST] <nevcairiel> we have vdpau and dxva2 [21:59:31 CEST] <nevcairiel> not sure why noone finished a vaapi thing [21:59:33 CEST] <philipl> BtbN: brave but futile. [21:59:36 CEST] <philipl> ? [21:59:46 CEST] <BtbN> because libva is... libva [22:00:00 CEST] <philipl> It's a special library. [22:00:37 CEST] <BtbN> The fact alone that for propper, fast, read-back of the frames you need a large bunch of assembly... [22:00:44 CEST] <BtbN> a simple memcpy from its uswc buffers will be dead slow [22:00:56 CEST] <BtbN> not even fast enough for 30fps, just because of the copy [22:01:00 CEST] <nevcairiel> someone was working on a patch for that [22:01:03 CEST] <philipl> As for vdpau: The hevc fix has been made inside nvidia and will appear in another month or so, I think. [22:01:25 CEST] <nevcairiel> but he made it hard on himself by making the function too generic [22:01:29 CEST] <nevcairiel> so it never got finished =p [22:01:30 CEST] <philipl> There was that Kodi announcement about the non-bullshit EGL way to use libva. Of course, that might not help with read-back [22:01:41 CEST] <BtbN> nevcairiel, vlc already has such a function [22:01:54 CEST] <nevcairiel> the vlc function is like 200% too complex [22:02:09 CEST] <BtbN> For some reason vlc allways uses it instead of memcpy. According to my tests it's slower than a plain memcpy for normal memory though. [22:03:31 CEST] <philipl> Speaking of ass-slow - we still need to work out why the nvenc sample app is so much faster than ffmpeg. [22:05:31 CEST] <BtbN> I guess because of less copying the frames around [22:05:34 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:e171309756cb: avutil/softfloat: Add a test for av_sincos_sf() [22:05:46 CEST] <philipl> But which are the extra copies. [22:06:11 CEST] <philipl> The sample app writes output frames straight to disk, so it skips a copy there, but when I modified the app to copy to a memory buffer first, it made no significant impact. [22:06:17 CEST] <BtbN> Was it a fair comparison, yuv without the need to convert the format? [22:06:32 CEST] <philipl> Yeah, I used the same raw yuv input and tried to normalise as much as possible [22:06:50 CEST] <BtbN> and the sample was running in CUDA mode, not DirectX? [22:06:59 CEST] <philipl> and we've got a step up because we use the yv12 input format and the sample app uses nv12 with conversion. [22:07:04 CEST] <rcombs> I've had similar experiences experimenting with VAAPI encoding [22:07:07 CEST] <philipl> This was all on linux, so cuda, yeah [22:07:18 CEST] <BtbN> hmm [22:07:31 CEST] <BtbN> There shouldn't be anything super inefficient in the ffmpeg encoder [22:08:13 CEST] <philipl> There shouldn't be, but it ends up 2x slower in the fastest configurations [22:08:33 CEST] <philipl> When you go all the way to the hardware limits (4k hevc), then the speeds are comparable, unsurprisingly [22:10:38 CEST] <BtbN> strange [22:11:19 CEST] <philipl> indeed. [22:11:43 CEST] <philipl> I tried using gprof and oprofile at various times but I'm clearly not capable of getting anything useful out of them [22:11:57 CEST] <BtbN> The sample app is using nvenc in some async mode if i remember right? [22:12:54 CEST] <philipl> The docs say async mode isn't supported on linux. [22:12:57 CEST] <philipl> but that might be a lie [22:13:03 CEST] <BtbN> ah, right [22:13:09 CEST] <BtbN> no, definitely not a lie [22:13:13 CEST] <BtbN> it uses WinAPI stuff for that [22:14:22 CEST] <philipl> Yeah, it does use it on windows but that's disabled when built on linux [22:14:25 CEST] <philipl> so that doesn't explain it [22:14:42 CEST] <BtbN> that would have been a good explanation though [22:21:33 CEST] <wm4> <philipl> As for vdpau: The hevc fix has been made inside nvidia and will appear in another month or so, I think. <- oh nice [22:21:43 CEST] <wm4> <philipl> There was that Kodi announcement about the non-bullshit EGL way to use libva. Of course, that might not help with read-back <- I didn't find any information about that [22:21:50 CEST] <wm4> may as well not exist [22:22:05 CEST] <philipl> wm4: Yep. I'm also nudging the nvidia guy towards exposing vp8/9 decode next. [22:22:12 CEST] <BtbN> It's only usefull for directly rendering decoded stuff [22:22:17 CEST] <BtbN> not for getting the decoded data [22:22:52 CEST] <philipl> BtbN: People might start thinking vaapi was usable if they did that. [22:23:46 CEST] <BtbN> Non-Dev people already think that, and praise intels great open source. [22:23:51 CEST] <wm4> lol [22:24:03 CEST] <wm4> I've had so many problems with vaapi [22:24:15 CEST] <philipl> wm4: That's its job. [22:24:16 CEST] <wm4> vdpau just works [22:24:48 CEST] <BtbN> everything from nvidia just works. I don't care anymore that their drivers are closed source. They do what they are supposed to. [22:25:03 CEST] <philipl> Yeah, working is a feature. [22:25:42 CEST] <nevcairiel> linux, working is a feature [22:26:51 CEST] <rcombs> anime, Working!! is a feature [22:27:15 CEST] <philipl> anime: hardware-incompatible encoding is a feature. :-) [22:35:29 CEST] <BtbN> philipl, the most notable difference i see is that the sample encoder does not start outputting before it runs out of buffers [22:36:01 CEST] <BtbN> the ffmpeg encoder start as soon as possible [22:36:24 CEST] <nevcairiel> introducing delay can help hardware substantially in performance [22:36:30 CEST] <nevcairiel> as it increases parallelism [22:37:34 CEST] <BtbN> When the bufferQueue does not have any available encode buffers anymore, it will output one frame from it, so it has a buffer available again. [22:41:57 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:c105e0f077fb: avcodec/aacps_fixed_tablegen: change f_center to 64bit to avoid overflow [22:44:33 CEST] <jamrial> philipl: h264 high10 profile anime encoding, also known as how to annoy mobile users [22:48:12 CEST] <philipl> jamrial: pretty much. [22:48:35 CEST] <philipl> BtbN: that's a good point. They do say that maximum performance requires filling the queue all the way [22:48:39 CEST] <wm4> so, uh, does anyone happen to know anything about that new libav EGL interop? [22:49:04 CEST] <BtbN> libav? libva? [22:49:14 CEST] <philipl> wm4: No. The libva history doesn't call it out and the Kodi side didn't actually explain what was what. [22:49:28 CEST] <philipl> I guess look at the relevant kodi dev branch for commits that look related [22:50:55 CEST] <philipl> https://github.com/FernetMenta/xbmc/commits/EGL [22:51:00 CEST] <wm4> BtbN: sorry, libva [22:51:21 CEST] <wm4> github doesn't let you search the commit history [22:51:28 CEST] <philipl> Probably starting from the change on 31st may [22:51:39 CEST] <philipl> That is the EGl dev branch by the look of things [22:52:04 CEST] <BtbN> Basicaly it lets you bind an libva output thing to an OpenGL texture [22:52:16 CEST] <BtbN> so what vdpau has actual OpenGL extensions for [22:52:52 CEST] <philipl> Yeah, I think so - except they did it through egl for whatever reason [22:53:08 CEST] <BtbN> because they failed at doing it for GLX [22:53:14 CEST] <BtbN> and intel only assigns developers to new stuff [22:53:29 CEST] <philipl> Tried it once, didn't work, better write a wrapper. [22:53:45 CEST] <BtbN> The GLX "interop" is a bad joke [22:54:19 CEST] <philipl> wm4: so presumably, you can take the existing mpv egl support and wire it in relatively easily. [22:54:30 CEST] <philipl> "relatively" [22:54:55 CEST] <wm4> can't find it in the xbmc history [22:55:02 CEST] <wm4> I don't see a branch either [22:55:06 CEST] <philipl> https://github.com/FernetMenta/xbmc/commit/3588cddce89c50854f548981902f45c1… [22:55:26 CEST] <philipl> The first link I pasted is the history for the dev branch [22:55:57 CEST] <wm4> oh [22:56:00 CEST] <wm4> better [22:56:05 CEST] <wm4> it's not in the main repo [22:56:11 CEST] <philipl> right. not there yet. [22:57:25 CEST] <wm4> this is what the xbmc git history looks like https://0x0.st/scS.png [22:57:37 CEST] <philipl> hehe. [22:58:20 CEST] <wm4> reminds me of early mplayer, except with github instead of cvs [23:00:38 CEST] <philipl> So looks like the core of the change is in VAAPI.cpp in that diff [23:00:43 CEST] <philipl> all the EGL image stuff [23:02:28 CEST] <unknown23> hi, I am writing a demuxer for a new file format, in my "read_packet" function, I want to modify the underlying data a bit, so I really can't use "av_get_packet". how do I get my modified data to fit in "pkt"? [23:05:39 CEST] <wm4> unknown23: there are various methods to create (or set) AVPackets [23:06:22 CEST] <wm4> you could e.g. read the data from the file into memory, manipulate that, and then copy that to the AVPacket etc. [23:06:41 CEST] <philipl> ubitux: Can you look at Niklesh's latest diff [23:06:47 CEST] <unknown23> wm4, I see. searching for a suitable example then :) [23:07:00 CEST] <philipl> ubitux: is it ok to use an ASS reset tag and not explicitly close style tags? [23:07:43 CEST] <wm4> philipl: should be ok [23:07:54 CEST] <wm4> maybe it's actually even better [23:08:04 CEST] <wm4> because the user style could have some formatting set [23:08:18 CEST] <wm4> (didn't look at the patch) [23:09:10 CEST] <philipl> The patch is primarily adding support for font size overrides but he didn't want the reset block to get longer and longer by explicitly closing each separate part of the style. [23:09:23 CEST] <ubitux> philipl: you don't need to "close" tags in ass between events; tags just replace current markup [23:09:48 CEST] <ubitux> philipl: the reset will reset them all, is that what you want? [23:10:13 CEST] <philipl> ubitux: I believe so. The movtext semantic is that you have a style entry which fully describes all text properties and it has a start/end cannot overlap with other entries [23:10:21 CEST] <philipl> so when an entry ends, you should return to the default style [23:10:36 CEST] <ubitux> doesn't that mean you start a new Dialogue? [23:10:39 CEST] <philipl> When you make a partial change to a style, you still two fully described entries [23:11:03 CEST] <philipl> Well, in movtext you have a single sample line and a set of entries that describe it [23:11:19 CEST] <philipl> So if I want part of the line to be bold, then I have an entry with bold turned on that starts and ends appropriately [23:11:39 CEST] <unknown23> wm4, https://www.ffmpeg.org/doxygen/2.7/group__lavc__packet.html doesn't seem to say which function can copy a buffer into a packet? any hints? [23:12:05 CEST] <philipl> But as entries cannot overlap, each entry must fully describe all properties even if you don't change them at the same points. [23:12:11 CEST] <philipl> I don't think this is equivalent to dialogues. [23:12:28 CEST] <unknown23> just memcpy to pkt->buf? [23:12:30 CEST] <wm4> unknown23: if you want to copy, use memcpy [23:12:33 CEST] <wm4> yeah [23:12:41 CEST] <wm4> except pkt->buf is something else [23:12:42 CEST] <unknown23> ah ok ;) this is C after all :D [23:12:44 CEST] <ubitux> philipl: ok ok; well i'll try to have a look later, i have to go now [23:12:51 CEST] <philipl> ubitux: sure. Thanks. [23:12:57 CEST] <ubitux> but using \r is probably fine [23:13:07 CEST] <philipl> I think it's fine too. [23:13:09 CEST] <ubitux> as long as it renders correctly with libass,.. ;) [23:13:25 CEST] <philipl> I have been assuming he's been checking visually. :-) [23:14:47 CEST] <wm4> does ffmpeg output valid ASS yet [23:14:56 CEST] <wm4> instead of just ASS that only libass understands? [23:15:05 CEST] <philipl> is there any other kind? :-) [23:15:25 CEST] <wm4> philipl: I tried looking at that xmbc branch, but I have no idea what it does... maybe the VA interaction is still missing? [23:15:49 CEST] <wm4> yes, ASS is "defined" by vsfilter [23:16:47 CEST] <philipl> wm4: It looks like it's using eglCreateImageKHR and then using that with vaSyncSurface, roughly speaking [23:17:18 CEST] <philipl> but I'm no egl or va expert. [23:17:27 CEST] <cone-201> ffmpeg 03Anton Khirnov 07master:aa9d15d89bb4: qsvdec: avoid an infinite loop with no consumed data and no output [23:17:28 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:ef2a85ac5321: Merge commit 'aa9d15d89bb4ee8a31607d3db1b8c5334eb88d2d' [23:18:05 CEST] <wm4> ah [23:18:51 CEST] <wm4> oh, the vaapi headers even have docs for this [23:21:21 CEST] <philipl> where? [23:21:41 CEST] <wm4> va.h [23:21:45 CEST] <wm4> vaSyncSurface [23:23:51 CEST] <philipl> Well, I hope it makes sense to you :-) [23:25:38 CEST] <wm4> so apparently this works over VA_SURFACE_ATTRIB_MEM_TYPE_DRM_PRIME [23:26:03 CEST] <philipl> Ah. PRIME shows its head [23:26:07 CEST] <wm4> and then you need indeed to use eglCreateImageKHR to create a complicated mapping [23:26:49 CEST] <philipl> I see it now. [23:26:50 CEST] <philipl> Lovely [23:29:31 CEST] <wm4> looks pretty annoying [23:29:47 CEST] <philipl> But you couldn't expect any less [23:29:48 CEST] <wm4> I hope the VA surfaces don't need to be pre-mapped on allocation or so [23:30:13 CEST] <wm4> (that would be slightly annoying) [23:30:35 CEST] <philipl> Well, if you have two ways to do something and one is more annoying... [23:36:34 CEST] <BtbN> philipl, https://github.com/BtbN/FFmpeg/commit/cbf3d34a9a67ad9900946a5b9a160d2122deb… [23:36:54 CEST] <nevcairiel> does that actually make it faster? [23:37:00 CEST] <BtbN> Not tested yet [23:37:03 CEST] <cone-201> ffmpeg 03Anton Khirnov 07master:22522d9c2c69: qsvdec: fix a memleak of async_fifo [23:37:04 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:c3413a712ac2: Merge commit '22522d9c2c69624fe4d81d61ee65a56610f22f1d' [23:37:08 CEST] <BtbN> but it basicaly does what the nvenc samples do [23:37:18 CEST] <BtbN> fill all buffers before starting to read frames [23:37:18 CEST] <philipl> BtbN: will try it out [23:37:28 CEST] <nevcairiel> i wouldnt be that surpised if it does, as you would be best advised to do the same thing when decoding [23:37:30 CEST] <BtbN> Not tested yet _at all_ [23:37:37 CEST] <BtbN> i might have missed something and this breaks stuff [23:38:04 CEST] <BtbN> Might be off by one for the check [23:38:10 CEST] <nevcairiel> if you force the gpu to sync up while its still working, it might cut into performance, if you give it time to finish a few frames ahead, it might run faster [23:38:29 CEST] <BtbN> i will put that behind an option though [23:38:41 CEST] <BtbN> for some usecases the lower latency is more desireable than the enhanced speed [23:38:53 CEST] <nevcairiel> but should make the fast mode the default [23:38:57 CEST] <BtbN> yes [23:41:32 CEST] <philipl> BtbN: big big speed up [23:41:34 CEST] <wm4> btw. would you agree that for a frame pool (feeding get_buffer2), software decoding would best return the most recently used frame (for better cache use), while hardware decoding would return the oldest frame (to avoid any form of async still-in-use effects)? [23:42:44 CEST] <BtbN> Also found something else while digging through stuff [23:43:34 CEST] <BtbN> unrelated, but weird [23:43:42 CEST] <philipl> oh yeah? [23:44:10 CEST] <cone-201> ffmpeg 03Timo Rothenpieler 07master:15cd2f8ea9f6: avcodec/nvenc: Remove unused parameter [23:44:21 CEST] <BtbN> that [23:45:01 CEST] <philipl> I saw that warning. [23:45:13 CEST] <philipl> huh [23:46:07 CEST] <BtbN> the function accesses coded_frame through avctx itself [23:46:12 CEST] <BtbN> the parameter has allways been unused [23:47:09 CEST] <philipl> BtbN: for a dvd encode. 390 vs 1000 fps [23:47:15 CEST] <BtbN> what, wow [23:48:13 CEST] <nevcairiel> well thats that then [23:48:42 CEST] <cone-201> ffmpeg 03Anton Khirnov 07master:fa85fcf2b7d1: qsvenc_hevc: fix enum declaration [23:48:43 CEST] <cone-201> ffmpeg 03Michael Niedermayer 07master:af08d8bfbbed: Merge commit 'fa85fcf2b7d1ab822a59245077b8bb855406d3e9' [23:50:07 CEST] <BtbN> philipl, try increasing the 48 to 64 or even more, in line 693 [23:50:19 CEST] <BtbN> The 48 is a bit out of date [23:50:29 CEST] <nevcairiel> doubt that'll help with speed much [23:50:32 CEST] <BtbN> at least for maxwell cards with tons of ram [23:50:41 CEST] <nevcairiel> you probably only need ~10 or so for max speed [23:51:32 CEST] <BtbN> hm, could make it an int parameter for the amount of wait frames [23:51:42 CEST] <BtbN> so users can tune as they like [23:51:46 CEST] <nevcairiel> screw that [23:52:16 CEST] <BtbN> would be easier than a bool, most users would only ever set it from default to 0 [23:58:37 CEST] <philipl> BtbN: 4k h264 is now comfortably above 60fps [23:58:47 CEST] <philipl> hevc is still in the 50s. That's the hardware - can't do it [23:58:59 CEST] <philipl> the 48 -> 64 didn't do anything [23:59:20 CEST] <BtbN> ok, not going to touch that then [23:59:39 CEST] <BtbN> Interesting though, i expected something more convoluted [23:59:58 CEST] <philipl> We're owed something simple once in a while. [00:00:00 CEST] --- Sun Jul 26 2015

1 0

ffmpeg.log.20150725
by burek 26 Jul '15

26 Jul '15

[00:19:39 CEST] <UKCoder> Hi all, I was wondering if there was anyone who could tell me why when I run a 30s M2TS file through ffmpeg with the flags -r 30 -g 15 (trying to set frame rate to 30fps and GOP size to 15 frames) I end up with 67 GOPs instead of 60? [00:24:43 CEST] <c_14> Some encoders (x264 for example) also insert i-frames on scene changes. [00:28:30 CEST] <UKCoder> c_14 : do you know if that's the case with ffmpeg and MPEG2 video and if so, is there a way to suppress that type of behavior in ffmpeg? [00:30:20 CEST] <UKCoder> (btw - there's no scene change in the asset I'm using [00:30:29 CEST] <c_14> try -mpv_flags strict_gop [00:32:15 CEST] <UKCoder> c_14 : thanks for the suggestion, gave it a try, still getting 67 GOPs according to ffprobe [00:33:46 CEST] <UKCoder> even -keying_min 15 doesn't seem to affect it either :/ [00:33:53 CEST] <UKCoder> oops -keyint_min [00:35:38 CEST] <c_14> I'm getting exactly 60 with `ffmpeg -f lavfi -i color=black:r=30 -t 30 -c:v mpeg2video -g 15 out.mkv' [00:37:54 CEST] <c_14> Ditto when using mandelbrot [00:38:38 CEST] <c_14> Can you reproduce when using one of those as source? Because if you get != 67 i-frames with those sources something might be wrong with your ffmpeg version [00:39:36 CEST] <UKCoder> c_14 : What are you using to tell you the number of GOPs? I'm using ffprobe -show_frames myfile.ts | grep 'pict_type=I' | wc -l [00:40:14 CEST] Action: c_14 was using key_frame=1, but that should work. [00:40:22 CEST] <c_14> Do you maybe have an audio stream in there as well? [00:41:05 CEST] <UKCoder> c_14 : I do, but greping on pict_type=I should ignore audio right? [00:42:02 CEST] <c_14> Should, you might want to add '-select_streams v' anyway though. [00:45:03 CEST] <UKCoder> c_14 : in ffprobe? I'm getting an unrecognized option for select_streams [00:45:24 CEST] <c_14> in ffprobe [00:47:53 CEST] <Nolski> Whenever I change my encoding from QTmuxingAppLibWebM-0.0.1 to anything else, all of the sudden none of my ffmpeg commands work T_T [00:48:05 CEST] <Nolski> This stuff makes my brain hurt so bad [00:54:13 CEST] <UKCoder> c_14 : bizarre, I just used squeeze to transcode and confirmed that it was generating 15 fps GOPs with 30fps video using ffprobe then ran it through ffmpeg and ffprobe shows the resultant file with 67 GOPs [00:54:32 CEST] <UKCoder> instead of 60 (for 29.99s worth of content) [00:56:12 CEST] <UKCoder> Mind you, I am using a very old version of ffmpeg... going to download the latest and see if it still exhibits the same behavior [01:06:42 CEST] <UKCoder> c_14 : yup, still with the latest version of ffmpeg it's generating 67 GOPs instead of 60 [01:48:42 CEST] <c_14> Hmm, weird. No clue, sorry. [04:37:19 CEST] <pothibo> Anyone here? [04:39:08 CEST] <klaxa> no? (yes) [04:39:46 CEST] <pothibo> :D [04:40:30 CEST] <pothibo> Im wondering if its possible to get a stream from UDP (expecting h264 but it might need transcoding, not sure yet) and stream that feed to HLS within 1s latency with ffserver [04:41:10 CEST] <pothibo> Im at the initial tryout and I got a ~10sec latency behind a RTSP feed [04:41:43 CEST] <klaxa> i don't know about ffserver, but you can try using ffmpeg and 1 second segments [04:42:00 CEST] <pothibo> well, ffserver receives a feed from ffmpeg [04:42:03 CEST] <klaxa> i'm not sure how well that will work [04:42:24 CEST] <pothibo> Thats the thing. I have no idea where that 10s comes from.. [04:42:43 CEST] <pothibo> Its all bound localhost.. so there is a treshold somewhere [04:43:02 CEST] <pothibo> Im wondering if its not some issues with some keyframes or something [04:43:37 CEST] <klaxa> oh that could indeed be [04:44:32 CEST] <QQ> Hello. I'm attempting to concatenate (without reencoding) three aac files with VBR, using the concat demuxer (ffmpeg -f concat -i input.txt -c copy destination.aac). Everything works fine, except that the resulting file (destination.aac) reads the wrong duration both in Winamp and if I use ffprobe (or ffmpeg -i) on it. I even specify inside of input.txt the duration for each one of the files I'm merging, nothing changes. [04:44:39 CEST] <QQ> For what I've seen after searching a good while on google, it seems it's a pretty common problem. Is there any way to fix that and create a file that doesn't reencode AND writes the appropriate duration to the headers at the same time? Thanks a lot for any help provided. [07:15:13 CEST] <well0ne> hi, is it possible to run ffmpeg in daemon/background modus? [07:15:19 CEST] <well0ne> without nohup/screen [07:55:55 CEST] <well0ne> anyone? [07:56:34 CEST] <Canar> well0ne: what do you expect to do with it running as a daemon [07:56:49 CEST] <Canar> you just mean continue to encode? [07:57:20 CEST] <well0ne> nothing special, it doesnt have to accept commands, it should just launch in background without external tools like nohup or screen [07:57:42 CEST] <chungy> what are you trying to accomplish? [07:57:55 CEST] <well0ne> i already told you lol [07:57:58 CEST] <well0ne> running in background [07:58:00 CEST] <Canar> ffmpeg -i input.mp4 output.mkv & [07:58:10 CEST] <well0ne> no i mean real background [07:58:25 CEST] <well0ne> if i close the session the command is dead [07:58:29 CEST] <chungy> Can you explain more? [07:58:39 CEST] <well0ne> well there is not mutch to explain [07:58:49 CEST] <well0ne> i'm on a system where i cant use nohup/screen [07:59:02 CEST] <well0ne> and i want to pipe the ffmpeg process into background [07:59:23 CEST] <chungy> That's pretty much the way to do it. [07:59:27 CEST] <Canar> bash disown? [08:00:06 CEST] <Canar> https://en.wikipedia.org/wiki/Disown_%28Unix%29 [08:00:37 CEST] <well0ne> wont work [08:00:52 CEST] <Canar> if disown won't work, nothing else will fare better [08:01:16 CEST] <Canar> you're dealing with some bofh sysadmins [08:01:22 CEST] <chungy> if you can give more details about your system it might help [08:01:25 CEST] <well0ne> well i just asked if its possible to mod ffmpeg that way or if there is a existent patch [08:01:32 CEST] <Canar> yeah [08:01:38 CEST] <Canar> but that cannot be fixed on the ffmpeg side [08:01:45 CEST] <well0ne> aha? [08:01:46 CEST] <chungy> like... it's possible to use systemd as a user instance and you could run ffmpeg through that. It's convoluted but it'd work xD [08:01:54 CEST] <Canar> if disown won't solve the problem, there is no change that can do better [08:02:14 CEST] <well0ne> why so? [08:02:24 CEST] <well0ne> dont make sense to me [08:02:32 CEST] <Canar> because your sysadmin has something set up to remove processes of the logged out user [08:02:43 CEST] <Canar> ie. the system does not operate as a normal linux system [08:03:03 CEST] <well0ne> i dont have the tools, and i simply dont want them too [08:03:21 CEST] <well0ne> for example tell me how to start a ffmpeg command over php [08:03:27 CEST] <well0ne> without screen, or any other [08:03:32 CEST] <well0ne> which will run in background [08:03:41 CEST] <well0ne> without killing the process/timeout for php [08:03:41 CEST] <Canar> ugh [08:03:44 CEST] <Canar> you don't get it [08:03:47 CEST] <well0ne> i get it [08:03:50 CEST] <well0ne> really [08:03:52 CEST] <Canar> "run in backgroun" is meaningless [08:04:01 CEST] <Canar> ffmpeg does something [08:04:05 CEST] <Canar> what something do you want [08:04:10 CEST] <well0ne> other tools have daemon modus too [08:04:15 CEST] <well0ne> why ffmpeg not [08:04:24 CEST] <well0ne> thats just my question [08:04:28 CEST] <well0ne> i really get your point [08:04:41 CEST] <Canar> you cannot get my question given your stated interpretations [08:04:43 CEST] <Canar> cannot possibly [08:04:53 CEST] <Canar> "daemon modus" does something [08:05:06 CEST] <well0ne> yeah, as far as i know the most are waiting for commands [08:05:14 CEST] <well0ne> but thats not what i want to accomplish [08:05:21 CEST] <Canar> waiting for commands received how though [08:05:28 CEST] <Canar> you talk in meaningless generalizations [08:05:38 CEST] <Canar> no wonder you can't find answers [08:05:46 CEST] <well0ne> huh? [08:06:02 CEST] <Canar> your problem statement is ill-defined [08:06:24 CEST] <well0ne> if you mean [08:06:50 CEST] <chungy> ffmpeg runs with the operations you tell it, and then exits. It doesn't have a "daemon mode" [08:07:33 CEST] <Canar> ^^^ [08:07:44 CEST] <well0ne> my command never ends [08:07:47 CEST] <well0ne> its a loop [08:07:52 CEST] <well0ne> its streaming my files 24/7 [08:07:56 CEST] <Canar> good for your command [08:08:01 CEST] <well0ne> yeah i know [08:08:15 CEST] <Canar> i thought you were talking about ffmpeg though [08:08:21 CEST] <Canar> don't know why you bring your command up [08:08:28 CEST] <well0ne> lol [08:08:35 CEST] <well0ne> it was based on your previous talking [08:08:42 CEST] <well0ne> it doesnt exit [08:08:45 CEST] <well0ne> that was my point [08:09:08 CEST] <Canar> cool story bro [08:09:15 CEST] <well0ne> yah [08:13:03 CEST] <Canar> so i mean like [08:13:09 CEST] <Canar> you've got your command [08:13:15 CEST] <Canar> it transcodes one stream to another [08:13:20 CEST] <Canar> what's the input? what's the output? [08:13:51 CEST] <well0ne> thats not really necessary, [08:14:09 CEST] <well0ne> i just asked if there is a background option, a simple "no" would have been okay [08:14:18 CEST] <Canar> k [08:14:50 CEST] <Canar> ...but this "background option" you speak of isn't provided by processes, generally, it's provided by the os/shell [08:15:47 CEST] <well0ne> ok [08:16:11 CEST] <chungy> A program can double-fork and force itself to be disowned by the shell, but such an operation still doesn't make sense with ffmpeg given what the program does [08:16:55 CEST] <well0ne> i see [08:17:20 CEST] <chungy> hypothetically there could be a command that receives json or whatever of stuff to do and call off all the same functions that ffmpeg does, but without exiting. Doesn't exist (upstream) yet. [08:18:24 CEST] <well0ne> ah, cool, thanks [08:18:29 CEST] <well0ne> didnt know that [08:18:41 CEST] <Canar> if this kind of feature is useful to you, it's about an hour or two's worth of shellscript hacking to have a working prototype [08:18:49 CEST] <Canar> maybe quicker if you're eliter [08:19:13 CEST] <Canar> ...but then you need details [08:19:47 CEST] <chungy> also if your sysadmin doesn't let you keep processes running after you log out, such a command wouldn't be helpful to you anyway [08:24:23 CEST] <well0ne> well i can try to taslk to them [08:24:32 CEST] <well0ne> do you think disown would work in php too? [08:25:36 CEST] <Canar> if it won't work in your shell, i can't see why php would be different, but perhaps the policies are lax enough to let that sneak through [08:25:37 CEST] <chungy> php isn't bash. no [08:25:40 CEST] <Canar> you'd have to run through bash [08:26:08 CEST] <well0ne> ok, thanks for your kindness [08:26:33 CEST] <Canar> sorry if i can be a bit autistic regarding support details heh [08:26:57 CEST] <well0ne> its okay, my english is not the best, i have problems to explain what are my points [08:27:56 CEST] <well0ne> what my points are ? whats right? -.- its a long time ago... [08:31:01 CEST] <Canar> what your points are [08:31:14 CEST] <Canar> i barely notice, chat with people with broken english all the time lol [08:31:52 CEST] <Canar> est-ce que vous etes francais? [08:32:08 CEST] <Canar> your grammatical mistakes remind me of francophones [08:33:13 CEST] <well0ne> nope, i'm german [08:33:31 CEST] <well0ne> my english is quite well sometimes [08:33:43 CEST] <well0ne> but if i don write / speak it for some time [08:33:50 CEST] <well0ne> i have problems with gramar [08:34:22 CEST] <Canar> hahaha your english is still better than my french [08:34:31 CEST] <well0ne> possible^^ [08:34:39 CEST] <Canar> and that's the only other language i can even remotely speak [08:35:10 CEST] <well0ne> well, i think your french is better than my italian [09:00:28 CEST] <well0ne> Canar is it possible to tell a process to disown ? [09:00:40 CEST] <well0ne> i mean directly from itself [09:01:55 CEST] <well0ne> i think i already know the answer [09:36:51 CEST] <Canar> uh what [09:37:01 CEST] <Canar> disown is a bash command that does specific thingsw [09:37:48 CEST] <Canar> well0ne: in C, fork() [09:39:08 CEST] <Canar> http://linux.die.net/man/2/setsid ? [09:39:30 CEST] <Canar> http://linux.die.net/man/1/daemonize [09:42:07 CEST] <kahtahs> durandal_1707: select='gt(scene\,0.4)' <- what exaclty is that doing? doc says something about 'first scenes', but i would want all [09:43:10 CEST] <durandal_1707> selects depending on scene change score [09:45:47 CEST] <kahtahs> could you enforce some min/max numbers of frames? [09:48:55 CEST] <durandal_1707> maybe with some fu in equatition [14:26:02 CEST] <nardev> guys, i need somebody to write me commands for converting videos i'm prepared to pay that. it should be able to process feew different formats and convert videos to a few most popular formats.. nothing which would not work out of box in ffmpeg.. and maybe some tweaking about processor usage and quality but that second part is less important. Anyone interested? [17:06:57 CEST] <donguston> Anyone know of a good ffmpeg wrapper for C#? Needs to have progress output. [18:31:16 CEST] <TikityTik> is there a windows build that has --enable-libass? [18:32:11 CEST] <c_14> I think the zeranoe builds have it. [18:32:29 CEST] <JEEB> or you just use mpv's windows builds if you want to encode stuff with subtitles rendered on top of video [18:33:18 CEST] <TikityTik> JEEB: that's what i want [18:33:20 CEST] <TikityTik> mpv? [18:33:49 CEST] <JEEB> the video player which also has encoding support [18:33:52 CEST] <TikityTik> also why isn't --enable-libass default? [18:34:26 CEST] <JEEB> because FFmpeg has always been like that for most dependencies (zlib and bzip2 are the two exceptions I can think of) [18:34:28 CEST] <q3cpma> Hello, I have a dvd with a pixel aspect ratio different than 1. How can I expand the video before encoding it, and get the final viewing resolution? [18:36:02 CEST] <chungy> DVDs always have a SAR 8:9. Use the -aspect option to change the DAR (eg, -aspect 16:9 or -aspect 4:3) [18:36:12 CEST] <TikityTik> oh i did have zeranoe's build [18:36:20 CEST] <TikityTik> i just applied the syntax poorly [18:36:47 CEST] <JEEB> chungy: if you are using SAR in the wikipedia definition just stop that. no-one uses SAR for that [18:37:00 CEST] <q3cpma> i tried, but I still get 720x480 => 853x480 in the end [18:37:01 CEST] <JEEB> also DVDs have the aspect ratio correctly flagged in the video stream [18:37:17 CEST] <chungy> JEEB: I'm using SAR in the ffmpeg definition [18:37:24 CEST] <q3cpma> I also tried -vf setsar=1 [18:37:29 CEST] <chungy> q3cpma: yes, because the DAR is 16:9 [18:37:46 CEST] <JEEB> chungy: most definitely not [18:37:53 CEST] <JEEB> q3cpma: just go look at the scaling examples in either ffmpeg-all.html or the trac wiki [18:38:09 CEST] <chungy> I didn't even look at Wikipedia, only ffmpeg. [18:38:25 CEST] <JEEB> then you have no idea what SAR means if you say that DVDs have a SAR of 8:9 [18:38:25 CEST] <q3cpma> JEEB: I know how to scale, but I thought PAR was a different issue [18:38:30 CEST] <chungy> NTSC DVDs always have the source pixels at 720x480, it'll scale to 640x480 or 853x480 depending on 4:3 or 16:9 [18:39:11 CEST] <JEEB> q3cpma: I thought your issue was getting the resolution not being a modulo of your chroma subsampling [18:39:53 CEST] <q3cpma> No. I have a 720x480 video expanding to 853x480 when played, but I'd like a 853x480 video after conversion [18:40:18 CEST] <JEEB> which is what scaling filter does... except you don't want 853x480 with 4:2:0 YCbCr [18:40:23 CEST] <JEEB> because it's not mod2 [18:40:32 CEST] <CandyGumdrop> How do I use avdevice_list_devices in libavdevice? The second parameter is AVDeviceInfoList **---what does this mean I need to provide / allocate? [18:41:49 CEST] <JEEB> chungy: OK so 8:9 was to make 720x480 4:3. I stand corrected there. You do seem to know what you're talking about on some level. [18:41:55 CEST] <q3cpma> JEEB: Sure, I would crop [18:42:59 CEST] <anaraug> I just typed out a paragraph long question and then right before hitting enter figured out the answer for myself. [18:43:08 CEST] <anaraug> Just letting you guys know. [18:43:19 CEST] <anaraug> It was SO dumb. [18:43:43 CEST] <JEEB> chungy: btw if you want to know some scary stories about what DVDs really mean with the aspect ratio flag you should know that in reality the aspect ratio means the "analogue" (BT.601) derived ones. [18:43:52 CEST] <JEEB> This was later confirmed by the blu-ray spec [18:44:05 CEST] <JEEB> which sets those values specifically for SD content [18:44:12 CEST] <anaraug> I followed the instructions here https://trac.ffmpeg.org/wiki/Encode/YouTube for turning an audio file into a video with spectrograph. And was about to ask why on youtube it was in SD not HD. The answer was that youtube wasn't done processing yet. :( [18:44:19 CEST] <q3cpma> Bluray may be shit, but at least, it doesn't have all this bullshit [18:44:22 CEST] <kepstin> CandyGumdrop, I would expect that means you want to have a "AVDeviceInfoList *mylist;" then call avdevice_list_devices with "&mylist;" [18:44:35 CEST] <kepstin> modulo an extra ; that snuck in there [18:44:37 CEST] <nardev> guys, i need somebody to write me commands for converting videos i'm prepared to pay that. it should be able to process feew different formats and convert videos to a few most popular formats.. nothing which would not work out of box in ffmpeg.. and maybe some tweaking about processor usage and quality but that second part is less important. Anyone interested? [18:44:42 CEST] <JEEB> q3cpma: it does have non-1:1 SAR just for your information :P [18:44:50 CEST] <chungy> Blu-ray has backwards compatibility with DVD streams, heh. and likewise, DVD has backwards compatibility with VCD... [18:44:56 CEST] <chungy> 240p videos on a Blu-ray Disc anyone? :P [18:45:00 CEST] <JEEB> lol [18:45:03 CEST] <JEEB> I don't think that's in the spec [18:45:08 CEST] <JEEB> at least I don't remember seeing it [18:45:15 CEST] <JEEB> only down to 480i/480p [18:45:41 CEST] <kepstin> I wouldn't be surprised if 240p videos work on most bd players, even if it is out of spec :/ [18:45:52 CEST] <chungy> I'm not sure if BD allows for VCD streams, haha [18:45:57 CEST] <q3cpma> JEEB: Well, when I play the mt2s files, ffplay doesn't tell me about any scaling [18:46:15 CEST] <chungy> the old DVD codecs makes sense, sometimes it's easier to transfer special features straight from DVD [18:46:42 CEST] <TikityTik> I can't seem to get the subtitles filter working [18:46:42 CEST] <JEEB> q3cpma: that just means that you haven't stumbled on any such content yet :P [18:46:46 CEST] <q3cpma> JEEB: scaling doesn't work. It still tells me 832x480 => 858x480. [18:46:58 CEST] <JEEB> q3cpma: then you are using it wrong. also wtf is 832x480 [18:47:04 CEST] <q3cpma> (I cropped it to 832 width) [18:47:08 CEST] <TikityTik> this is what i'm inputting. ffmpeg -i out.mkv -c:v libvpx -crf 23 -c:a libvorbis -ac 1 -ar 44000 -b:a 96k -vf subtitles=out.mkv out.webm [18:47:27 CEST] <JEEB> q3cpma: also if you are very close to the SAR value but not exact, you may want to force the sar with setsar [18:47:42 CEST] <JEEB> also man that's a whole lot of cropping :P [18:48:12 CEST] <q3cpma> JEEB: I'll try. By the way, how do I set two variables in filters? I usually use "sws_flags=lanczos; scale=..." [18:48:17 CEST] <CandyGumdrop> kepstin: The function seems to be returning a negative value indicating an error. On the ML someone on a similar question suggests to use int avdevice_list_input_sources instead, and then says "But, dshow doesn't implement this API yet, so you will get an error." Does this mean there's no way of finding a list of inputs available with dshow? [18:48:39 CEST] <kepstin> hey, most analogue content on dvd is designed to have the edges not displayed, because analog->digital sampling is awesome... [18:48:54 CEST] <kepstin> so it often makes sense to crop a good chunk off the horizontal [18:49:28 CEST] <q3cpma> JEEB: forget that variable question, it's dumb [18:50:17 CEST] <chungy> scale=853:480:flags=lanczos+another_one [18:50:20 CEST] <JEEB> kepstin: I'm not sure about "not displayed" but yes, if you want to do it correctly and show it fullscreen on a 16:9 screen you crop 8px from both sides [18:50:57 CEST] <JEEB> (or you just show the full picture and enjoy the black bars on top and bottom because it's wider than 16:9) [18:51:03 CEST] <q3cpma> chungy: Thanks [18:51:22 CEST] <JEEB> DVDs are not really supposed to have non-active content otherwise though [18:51:40 CEST] <JEEB> also that 8px number comes from 480i DVDs [18:52:32 CEST] <JEEB> or I'm not sure if DVDs are supposed to have any non-active content at all. [18:53:20 CEST] <q3cpma> I tried "setsar=1, scale=853:480, crop=832:480:12:0" and it still doesn't work [18:53:28 CEST] <kepstin> q3cpma, wrong order [18:53:41 CEST] <kepstin> do scale=853:480, setsar=1, crop=832:480:12:0 [18:53:43 CEST] <q3cpma> Heh, I get it [18:54:04 CEST] <kepstin> scale updates the sar so the displayed video stays the same size [18:54:27 CEST] <kepstin> er, same aspect ratio anyways [18:54:31 CEST] <q3cpma> Now, is there a point in doing this? (Hehe) [18:54:53 CEST] <JEEB> only if your playback device can't handle aspect ratio correction [18:54:54 CEST] <q3cpma> kepstin: still gets me " 832x480 => 853x480 yuv420p" [18:55:25 CEST] <JEEB> in general when you re-encode DVDs you just crop and leave the aspect ratio to be handled by the player [18:55:51 CEST] <kepstin> unless you're encoding for really piece of junk hardware players or something, yeah [18:55:56 CEST] <q3cpma> JEEB: That's what I'll do in fine. Especially since I can crop to modulo 16 [18:56:12 CEST] <JEEB> mod16 cropping isn't really a thing any more [18:56:31 CEST] <q3cpma> x264 doesn't benefit anymore from it? [18:56:34 CEST] <JEEB> AVC and HEVC fixed the inefficiency to be so small that you just don't give a fuck. [18:56:40 CEST] <q3cpma> I see. [18:56:54 CEST] <q3cpma> Well, I won't use these anyway. [18:56:57 CEST] <q3cpma> Waiting for Daala [18:57:08 CEST] <kepstin> I mean, they had to fix it for AVC anyways because 1080p video. [18:57:23 CEST] <q3cpma> Even mod8 isn't helpful? [18:57:28 CEST] <JEEB> nope [18:57:53 CEST] <JEEB> if you start limiting your mod it's usually to fix playback on some shitty device or player you're supporting [18:57:54 CEST] <q3cpma> What about VPx? (not that I use them) [18:58:25 CEST] <JEEB> no idea about the possible compression loss there, but I'd guess that they really don't give a damn either in compression efficiency [18:59:49 CEST] <TikityTik> I have my ffmpeg complaining about fontconfig when I use the subtitles filter, any help? [19:00:16 CEST] <q3cpma> TikityTik: you should try to extract your subtitle with mkvextract. [19:00:57 CEST] <q3cpma> TikityTik: if I remember well, the subtitle filter only works with srt. You need the ass filter for, well, ssa/ass [19:01:46 CEST] <kepstin> q3cpma, nah, that's wrong. the 'subtitles' filter loads any file format ffmpeg can read with subtitles, and renders with libass. [19:02:22 CEST] <kepstin> in particular, it's the only way in ffmpeg to render ass subtitles with embedded fonts, which is probably where the fontconfig complaints are coming from. [19:02:26 CEST] <q3cpma> kepstin: well, I remembered badly. Can choose the stream with this filter [19:09:51 CEST] <TikityTik> q3cpma: I'm using the assfilter but i'm not seeing anything overlaying [19:11:06 CEST] <q3cpma> TikityTik: No errors from ffmpeg? [19:11:25 CEST] <TikityTik> it's giving me: Cannot load default config file [19:11:28 CEST] <TikityTik> for fontconfig [19:12:03 CEST] <q3cpma> TikityTik: That's strange. Could you try to edit the ass file (it's plaintext) and use a font you're sure to have installed? [19:24:45 CEST] <TikityTik> I'm just trying to figure out why it's complaining about my fontconfig [19:25:00 CEST] <q3cpma> TikityTik: You're using Windows? [19:25:08 CEST] <TikityTik> yes [19:25:25 CEST] <q3cpma> TikityTik: There are some forum post on the web about your error [19:25:33 CEST] <TikityTik> yeah i've been reading [19:25:50 CEST] <TikityTik> i tried making this fontconfig and setting the environment variables to point to it, but no luck [19:25:54 CEST] <nardev> anyone? [19:26:06 CEST] <nardev> i posted my question here: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2238 [19:27:29 CEST] <q3cpma> nardev: well, what don't you understand about ffmpeg that impeach you from doing this? [19:27:55 CEST] <nardev> q3cpma, i just want to save some time, i'm not into that.. [19:28:03 CEST] <nardev> my requirements are not complex [19:28:26 CEST] <nardev> i have some clips up to 100MB and under 1 minute [19:28:32 CEST] <nardev> which i want to convert to [19:28:44 CEST] <nardev> appropriate formats to be able to play them anywhere [19:28:48 CEST] <nardev> the server is Debian [19:29:03 CEST] <nardev> with multimedia repos where i gor ffmpeg [19:29:06 CEST] <nardev> *got [19:29:15 CEST] <BtbN> you could have already solved it yourself by now [19:29:25 CEST] <nardev> BtbN, please help [19:29:28 CEST] <nardev> i'll pay you for that [19:29:38 CEST] <nardev> i'm not into that.. [19:30:43 CEST] <q3cpma> nardev: Well, except if one of those input formats isn't supported by ffmpeg, you don't care about it. [19:31:17 CEST] <q3cpma> nardev: webm is a container. I suppose you mean vp8. Same for ogg (ogv), theora? [19:31:18 CEST] <nardev> q3cpma, ok.. so ignore that part, just the [19:31:21 CEST] <nardev> export [19:31:28 CEST] <nardev> ogg [19:31:39 CEST] <nardev> something which would play anywhere [19:31:59 CEST] <nardev> so my goal.. [19:32:00 CEST] <q3cpma> I don't know if theora can be played everywhere [19:32:01 CEST] <nardev> 4-5 [19:32:09 CEST] <nardev> q3cpma, i mean in total [19:32:24 CEST] <nardev> i want to get 4-5 different versions of same clip [19:32:35 CEST] <nardev> and that one of those versions is possible to play [19:32:43 CEST] <nardev> where ever i need that clip [19:32:46 CEST] <q3cpma> nardev: You just need an h264 profile/level compatibility table [19:33:18 CEST] <nardev> q3cpma, .. ok.. would you be able to provide me with a command and the rest? [19:33:44 CEST] <nardev> and tell me in private how much would that cost me [19:33:44 CEST] <q3cpma> nardev: have a you a bitrate/filesize limit? [19:34:01 CEST] <nardev> q3cpma, well if bitrate can affect playing on an iphone [19:34:04 CEST] <q3cpma> Man, I don't want money for such a thing. It'd be shamefule for me (and for you) [19:34:06 CEST] <nardev> yes.. [19:34:22 CEST] <nardev> q3cpma, i would be honored to pay you if it works for me.. [19:34:26 CEST] <nardev> nothing to be ashamed [19:34:42 CEST] <nardev> you do what you are an expert for i do what i do.. [19:34:49 CEST] <q3cpma> https://developer.apple.com/library/ios/documentation/NetworkingInternet/Co… [19:35:54 CEST] <q3cpma> Apparently, high(a)4.1 is supported. You need baseline(a)3.1 if you want to support the ipad, the iphone 3G and the ipod touch [19:36:44 CEST] <nardev> q3cpma, oki.. [19:37:08 CEST] <nardev> so it's double video clip version for ipphone? [19:37:12 CEST] <nardev> it is ok.. [19:37:33 CEST] <q3cpma> double video clip? what? [19:37:37 CEST] <nardev> is it reverse compatible if i get baseline(a)3.1 would it play on newer iphone? [19:37:49 CEST] <q3cpma> Yeah, of course. [19:38:12 CEST] <nardev> than would only need that version with baseline(a)3.1? [19:39:19 CEST] <nardev> q3cpma, could i expect you to do it for me? [19:40:23 CEST] <q3cpma> I don't know about your resolution or bitrate, but I'd try something like "ffmpeg -i input -c:v libx264 -profile:v baseline -level:v 3.1 -preset veryslow -crf 20 -c:a libfdk_aac -b:a -profile:a aac_he -b:a 64K -sn output.mp4é [19:40:30 CEST] <coops> hey folks. I'm trying to stream audio to ffserver, but when I start the ffmpeg client it says: "Missing video stream which is required by this ffm". Can I either configure ffserver to allow audio-only streams, or add dummy video to the stream? [19:40:47 CEST] <q3cpma> You'll probably need a -b:v to restrain the video bitrate, but i don't know what quality nor resolution you want and have. [19:41:02 CEST] <q3cpma> I'd advise you to experiment with different birates [19:41:29 CEST] <q3cpma> Oups, I mean -maxrate [19:41:35 CEST] <q3cpma> Not -b:v [19:42:36 CEST] <q3cpma> nardev: Also, you'll probably have to get libfdk_aac somewhere and recompile ffmpeg [19:47:47 CEST] <nardev> q3cpma, what is libfdk_aac [19:48:11 CEST] <q3cpma> A library to encode he-aac [19:48:45 CEST] <nardev> which i would need why? [19:49:25 CEST] <q3cpma> The native ffmpeg aac encoder is shit (well, it was when I tried it), and you need a low bitrate for streaming shit. [19:49:40 CEST] <q3cpma> cf https://trac.ffmpeg.org/wiki/Encode/AAC [19:50:04 CEST] <nardev> q3cpma, aha.. so that is only in case i need to lower bitrate so that i don't loose a lot on video quality? [19:50:19 CEST] <q3cpma> nardev: basically [19:50:37 CEST] <BtbN> It's not shit, it's already quite ok if you don't want to use ultra low bitrates [19:50:50 CEST] <nardev> q3cpma, so in case my source videos are not over HD i would not have to worry about that? [19:50:55 CEST] <BtbN> It's only AAC LC [19:51:04 CEST] <nardev> ok [19:51:21 CEST] Last message repeated 1 time(s). [19:51:21 CEST] <q3cpma> nardev: it depends on what compromise you're ready to make [19:51:27 CEST] <nardev> q3cpma, i will try without that.. and i will write down that a s i tip.. [19:51:45 CEST] <q3cpma> Ok. Maybe you'll want to scale your videos [19:51:46 CEST] <nardev> q3cpma, right now.. it will be i ok i hope.. [19:52:06 CEST] <nardev> q3cpma, well i would scale them if ULTRA HD is [19:52:06 CEST] <q3cpma> But you should really learn how to use ffmpeg, it's really not hard [19:52:07 CEST] <nardev> ? [19:52:32 CEST] <q3cpma> I mean downscale them, to reduce the bitrate requirements (then again, what the source resolution) [19:52:41 CEST] <nardev> q3cpma, i'm into bash for many years but i don't know anything about codecs bitrates, formats etc.. [19:53:10 CEST] <q3cpma> nardev: Why don't you use a GUI if you're not into this? Handbrake is good for beginners [19:53:11 CEST] <nardev> source resolution is what you can get from most mobile phones.. [19:53:13 CEST] <nardev> about HD [19:53:29 CEST] <nardev> q3cpma, i have to make web interface :P [19:53:47 CEST] <q3cpma> nardev: Ah [19:53:52 CEST] <nardev> i already made that to work with vzaar and it's SHIT [19:53:59 CEST] <nardev> all other services are SO EXPENSIVE... [19:54:09 CEST] <q3cpma> What the resolution of your input file? [19:55:06 CEST] <TikityTik> is there a way I can use -ss with the subtitles overlay? [19:55:08 CEST] <nardev> q3cpma, usually from mobile phones [19:55:13 CEST] <nardev> it can be HD [19:55:17 CEST] <TikityTik> i mean with the subtitiles filter [19:55:37 CEST] <nardev> but i some people have that nasty phones with ultra HD [19:55:40 CEST] <q3cpma> nardev: Could you explain what service you're required to provide? It's pretty vague to me [19:56:29 CEST] <nardev> q3cpma, basically : upload video -> with videojs play in on web, should be possible to be played on android, iphone and chrome default web browsers [19:56:32 CEST] <nardev> that is about it.. [19:56:38 CEST] <nardev> no high traffic [19:56:42 CEST] <nardev> very strong server [19:56:48 CEST] <q3cpma> Users upload videos? [19:56:54 CEST] <nardev> q3cpma, yep [19:57:16 CEST] <q3cpma> They are to be streamed or downloaded after conversion? [19:57:29 CEST] <nardev> streamed [19:57:34 CEST] <nardev> just that [19:57:40 CEST] <nardev> html5 player ... [19:58:04 CEST] <nardev> q3cpma, so you said this would work what for me? : "ffmpeg -i input -c:v libx264 -profile:v baseline -level:v 3.1 -preset veryslow -crf 20 -c:a libfdk_aac -b:a -profile:a aac_he -b:a 64K -sn output.mp4" [19:58:13 CEST] <q3cpma> In this case, you'll really benefit from using libfdk_aac [19:58:27 CEST] <nardev> q3cpma, ok i'll recompole it.. [19:58:31 CEST] <nardev> no warries [19:58:43 CEST] <nardev> but how to get all those different formats that i got in requirements [19:58:44 CEST] <nardev> ? [19:59:03 CEST] <q3cpma> "ffmpeg -i input -c:v libx264 -profile:v baseline -level:v 3.1 -preset veryslow -crf 20 -maxrate 1000K -c:a libfdk_aac -b:a -profile:a aac_he -b:a 64K -sn output.mp4" [19:59:10 CEST] <q3cpma> And you need to scale it [19:59:17 CEST] <q3cpma> downscale it to a sane value [19:59:18 CEST] <nardev> q3cpma, no [19:59:22 CEST] <nardev> ig it is playable [19:59:26 CEST] <nardev> on ay device [19:59:33 CEST] <q3cpma> The thing is that you can't use maxrate with scaling it [19:59:34 CEST] <nardev> i'm afraid high resolutions would no t pley [19:59:56 CEST] <nardev> this was my guidelines.. (btw zencoder is great BUT SO EXPENSIVE!!!) [20:00:39 CEST] <q3cpma> Well, if you play a 4K video on a 840x480 screen, it'll be useless [20:01:36 CEST] <q3cpma> I'd use 1280x720 as output, personally [20:02:28 CEST] <nardev> q3cpma, for all videos [20:02:28 CEST] <nardev> ? [20:02:34 CEST] <nardev> i would too.. [20:02:43 CEST] <q3cpma> Yeah, so you can precise a maxrate [20:02:49 CEST] <nardev> i'm afraid smaller screens are on devices that would jam... [20:03:04 CEST] <nardev> so.. rescaling would be enough? [20:03:16 CEST] <nardev> is there a way to rescale only those that are bigger and to leave [20:03:19 CEST] <nardev> others as is [20:03:20 CEST] <nardev> ? [20:03:26 CEST] <q3cpma> The only thing you need for device compatibility is profile and level [20:03:37 CEST] <q3cpma> (if your device support h264 and aac, of course) [20:04:11 CEST] <q3cpma> Also, what's the framerate? [20:08:13 CEST] <nardev> hm :( [20:08:16 CEST] <nardev> q3 [20:08:26 CEST] <nardev> q3cpma, wait i have to investigate :( [20:08:56 CEST] <q3cpma> ffmpeg -i input -c:v libx264 -profile:v baseline -level:v 3.1 -preset veryslow -crf 20 -maxrate YOURRATE -r 24 -c:a libfdk_aac -b:a -profile:a aac_he -b:a 64K -sn output.mp4 [20:09:00 CEST] <q3cpma> Oups [20:09:22 CEST] <q3cpma> ffmpeg -i input -vf scale=-1:720 -c:v libx264 -profile:v baseline -level:v 3.1 -preset veryslow -crf 20 -maxrate YOURRATE -r 24 -c:a libfdk_aac -b:a -profile:a aac_he -b:a 64K -sn output.mp4 [20:09:27 CEST] <q3cpma> It's what I'd use [20:10:39 CEST] <nardev> q3cpma, just tell me one thing [20:10:51 CEST] <nardev> that ffmpeg command that you wrote [20:10:55 CEST] <nardev> would cover ll [20:10:57 CEST] <nardev> my [20:11:07 CEST] <nardev> all my output formats which i need ? [20:11:15 CEST] <nardev> for all my devices? [20:11:20 CEST] <q3cpma> Probably not. But the main platforms yes [20:11:21 CEST] <nardev> i don't need separate files? [20:11:53 CEST] <q3cpma> Smartphones, tablets and desktop [20:12:01 CEST] <q3cpma> I don't know about other devices [20:12:40 CEST] <TikityTik> for some reason when I use -t with ffmpeg, it makes the export freeze on the first frame a bit. [20:13:33 CEST] <q3cpma> TikityTik: strange. What's your command? [20:14:19 CEST] <nardev> q3cpma, :( phones are more important :( [20:14:20 CEST] <TikityTik> ffmpeg -ss 1:36:38 -i eva.mkv out.mkv [20:14:33 CEST] <TikityTik> q3cpma: I then stop it manually [20:15:13 CEST] <q3cpma> nardev: phones or smartphones? [20:15:19 CEST] <nardev> smartphones only [20:15:45 CEST] <nardev> iphone 3 and above and android phones not older than 3-4 years [20:15:47 CEST] <q3cpma> nardev: Well, I advised baseline(a)3.1 because it even works on old iphones [20:15:57 CEST] <q3cpma> TikityTik: You said you were using -t [20:16:05 CEST] <nardev> so this would work even for that? ffmpeg -i input -vf scale=-1:720 -c:v libx264 -profile:v baseline -level:v 3.1 -preset veryslow -crf 20 -maxrate YOURRATE -r 24 -c:a libfdk_aac -b:a -profile:a aac_he -b:a 64K -sn output.mp4 [20:16:18 CEST] <TikityTik> q3cpma: ffmpeg.exe -ss 1:36:38 -i eva.mkv -t 2:07 out.mkv [20:16:32 CEST] <TikityTik> it's freezing a bit on the first frame [20:16:46 CEST] <q3cpma> nardev: You should check the profile/level requirement for other OSes, too. [20:16:53 CEST] <q3cpma> TikityTik: Isn't the -t before the -i? [20:17:35 CEST] <TikityTik> I'll give it a try [20:17:38 CEST] <q3cpma> TikityTik: Also, you should always try to play your video with ffplay, to see if there are errors [20:17:46 CEST] <TikityTik> oh [20:18:29 CEST] <TikityTik> q3cpma: typing this didn't make it stop. ffmpeg.exe -ss 1:36:38 -t 2:07 -i eva.mkv out.mkv [20:18:38 CEST] <TikityTik> it kept on encoding [20:19:10 CEST] <q3cpma> TikityTik: Sorry, I'm dumb, the -ss after the -i, normally [20:19:45 CEST] <q3cpma> nardev: another interesting page for you https://developer.android.com/guide/appendix/media-formats.html [20:20:46 CEST] <TikityTik> q3cpma: I added -ss after -i and it seems to be stuck [20:21:01 CEST] <TikityTik> where it shows frame= 0 and fps= 0.0 [20:21:06 CEST] <nardev> q3cpma, that is exactly why i was looking to pay somebody [20:21:27 CEST] <nardev> so that i don't need to spend days going around and tuning this shit to work [20:21:49 CEST] <q3cpma> TikityTik: It may take some time, -ss after the -i is precise seeking [20:22:07 CEST] <TikityTik> arg, isn't there a faster way? [20:22:43 CEST] <q3cpma> nardev: I gave you the command, and you basically only have to search for "h264 profile level OSname" to see what's the maxium profile and level you can use [20:22:54 CEST] <q3cpma> It's only websurfing nox [20:23:53 CEST] <nardev> q3cpma, :) [20:24:11 CEST] <q3cpma> TikityTik: https://trac.ffmpeg.org/wiki/Seeking [20:24:44 CEST] <q3cpma> TikityTik: there's a thing called "combined seeking" apparently [20:26:24 CEST] <c_14> -ss before -i is both fast and accurate (with recent versions of ffmpeg) [20:27:01 CEST] <q3cpma> c_14: yeah, but he seemd to have some problems with it [20:27:15 CEST] <q3cpma> TikityTik: have you tried playing it with ffplay to look for errors? [20:27:33 CEST] <TikityTik> yeah there was an error [20:27:48 CEST] <q3cpma> TikityTik: which is? [20:28:49 CEST] <TikityTik> SDL_OpenAudio (6 channels, 48000 Hz): DirectSound CreateSoundBuffer: Invalid parameter [20:28:53 CEST] <TikityTik> SDL_OpenAudio (4 channels, 48000 Hz): DirectSound CreateSoundBuffer: Invalid parameter [20:29:22 CEST] <q3cpma> TikityTik: well, that's not it. Maybe you should try with more precise number? Like 01:36:38.000 [20:32:31 CEST] <TikityTik> q3cpma: I think that did the trick [20:32:36 CEST] <TikityTik> q3cpma: finally got what i wanted [20:32:47 CEST] <q3cpma> TikityTik: which one did fix it? [20:33:06 CEST] <TikityTik> making it more precise [20:33:10 CEST] <TikityTik> or it might've been luck lol [20:33:32 CEST] <q3cpma> TikityTik: Yeah, it seems strange to me [20:33:38 CEST] <TikityTik> the main problem was the freezing and when i would try to encode from the export, it would giveo ut an error [20:34:20 CEST] <q3cpma> TikityTik: What do you mean by "export"? [20:34:55 CEST] <TikityTik> q3cpma: I would encode an out.mkv [20:35:06 CEST] <TikityTik> then I would use the subtitles filter on out.mkv to make out2.mkv [20:35:19 CEST] <TikityTik> because the subtitles filter would break if i used -ss [20:35:19 CEST] <q3cpma> Wow, why would you do that? [20:35:38 CEST] <TikityTik> q3cpma: the filter would break on -ss [20:35:45 CEST] <TikityTik> for input seeking [20:36:06 CEST] <q3cpma> TikityTik: Break? Like give you an error or simply not work? [20:36:27 CEST] <TikityTik> q3cpma: it would render the subs from the beginning of the video instead of where I asked from -ss [20:36:40 CEST] <TikityTik> not sure if that's reported [20:36:51 CEST] <q3cpma> TikityTik: you put the vf after -ss and -t, right? [20:36:53 CEST] <nardev> q3cpma, Unable to find a suitable output format for 'aac_he' [20:36:56 CEST] <TikityTik> q3cpma: yeah [20:37:18 CEST] <q3cpma> nardev: repost the command, there must be a dumb mistake somewhere [20:37:31 CEST] <q3cpma> TikityTik: Well, that's strange [20:38:00 CEST] <TikityTik> i would repost it but my computer crashed, as i'm overclocking [20:38:39 CEST] <q3cpma> nardev: There's a "-b:a" before profile:a that has nothing to do here [20:38:39 CEST] <nardev> q3cpma, ffmpeg -i misijaweb9.mp4 -vf scale=-1:720 -c:v libx264 -profile:v baseline -level:v 3.1 -preset veryslow -crf 20 -r 24 -c:a libfdk_aac -b:a -profile:a aac_he -b:a 64K -sn output.mp4 [20:38:58 CEST] <q3cpma> Remove the first -b:a [20:39:07 CEST] <nardev> aha [20:39:10 CEST] <nardev> works now [20:39:37 CEST] <nardev> q3cpma, if you sent constant bitrate why to use max bitrate then? [20:39:51 CEST] <q3cpma> nardev: What? [20:40:19 CEST] <nardev> i see you sen -crf here and also you suggested -maxrate [20:40:49 CEST] <q3cpma> nardev: Ah. It's because you're streaming. You need a low bitrate. -crt + -maxrate is constrained mode. [20:41:08 CEST] <q3cpma> Try 2M, since that's what the Android page recommends [20:41:29 CEST] <TikityTik> q3cpma: is it recommended to use both -crf and -b:v if I'm encoding in libvpx? [20:42:03 CEST] <q3cpma> TikityTik: It's mandatory, otherwise it defaults to 250K if I remember well. I hate libvpx with passion. [20:42:29 CEST] <TikityTik> q3cpma: why do you hate it? [20:42:54 CEST] <q3cpma> TikityTik: Slow, stupid option (like you can see), not properly threaded and Googleware [20:43:21 CEST] <nardev> q3cpma, i have 16 cores xeon can i tune this to get more resources? [20:43:27 CEST] <q3cpma> libvpx-vp9 encode under 0.1FPS on my 8350, to say how it's slow [20:43:39 CEST] <pzich> are you using -threads 16? :D [20:43:48 CEST] <nardev> pzich, no [20:43:51 CEST] <nardev> ok i'll try [20:43:56 CEST] <TikityTik> q3cpma: How should I go about adjusting crf and video bitrate? Do I adjust one before the other until satisfied, and use the other to make slight adjustments? [20:44:00 CEST] <q3cpma> nardev: normally, you have nothing to do with libx264. ffmpeg automatically uses nproc threads [20:44:13 CEST] <JEEB> note1: you should never use both crf and b:v together [20:44:30 CEST] <nardev> q3cpma, so -threads would not do anything? [20:44:32 CEST] <JEEB> note2: libvpx finally got in-picture threading in the encoder a few months back [20:44:37 CEST] <q3cpma> TikityTik: You usually use a mix of crf, qmin/qmax and b:v [20:44:53 CEST] <q3cpma> JEEB: I tried it, but it was still going at 0.1FPS for me [20:44:54 CEST] <TikityTik> i never used qmin and qmax, i read that qmax doesn't do much. [20:45:05 CEST] <JEEB> q3cpma: yeh - it's slow [20:45:09 CEST] <q3cpma> Well, with libvpx it's really useful [20:45:25 CEST] <q3cpma> Because it has poor movement encoding [20:45:47 CEST] <JEEB> also I would never use qmin/max together with anything :V I would only use a rate control mode (either crf or ABR) and maxrate+bufsize for keeping the stream within bandwidth constraints [20:46:38 CEST] <q3cpma> JEEB: libvpx behaves strangely [20:47:23 CEST] <q3cpma> TikityTik: use this page http://www.webmproject.org/docs/encoder-parameters/ [20:49:52 CEST] <TikityTik> q3cpma: where do i input --passes? [20:49:57 CEST] <TikityTik> it's telling me option not found [20:50:40 CEST] <q3cpma> TikityTik: It's for vpxenc, not ffmpeg [20:50:45 CEST] <TikityTik> ah [20:50:54 CEST] <q3cpma> For ffmpeg, it's -pass 1 or -pass 2 [20:52:18 CEST] <q3cpma> TikityTik: What's your goal? It's to post on an imageboard or something like this? [20:52:29 CEST] <TikityTik> yeah [20:52:33 CEST] <nardev> q3cpma, that command doesn't even work on android :( [20:52:39 CEST] <TikityTik> 2 pass makes the file size larger? [20:52:54 CEST] <q3cpma> nardev: Well, then the android page has problems. [20:53:01 CEST] <nardev> :D [20:53:06 CEST] <nardev> q3cpma, https://www.footgoll.com/videojs/demo.html [20:53:32 CEST] <q3cpma> TikityTik: No, it makes it better for the same size when being limited [20:53:44 CEST] <q3cpma> nardev: Why do you use HTML5 if you don't need to embed? [20:54:03 CEST] <nardev> q3cpma, i need to embed ... [20:54:08 CEST] <nardev> i play in browsers [20:54:10 CEST] <TikityTik> q3cpma: this is what i got so far, trying to lower the filesize. ffmpeg -i out2.mkv -c:v libvpx -c:a libvorbis -pass 2 -crf 35 -b:v 1000k -ar 22050 -ac 1 -b:a 64k -vf scale=-1:640 out.webm [20:54:18 CEST] <nardev> that is what i said at first place :( [20:55:53 CEST] <q3cpma> nardev: Well, in this case, I don't know. [20:56:03 CEST] <TikityTik> q3cpma: is it trial and error trying to get something to a certain file size? [20:56:13 CEST] <nardev> :)_ [20:56:21 CEST] <nardev> q3cpma, thank you anyway [20:56:24 CEST] <q3cpma> TikityTik: No, you just habe to do some math [20:56:31 CEST] <nardev> i'm in deep shits [20:56:40 CEST] <TikityTik> q3cpma: is it simple math at least? [20:57:27 CEST] <q3cpma> TikityTik: Just do a script. It's easy to find the maxrate or -b:v [20:57:57 CEST] <TikityTik> q3cpma: a script? [20:58:23 CEST] <q3cpma> TikityTik: I mean a shell script, to calculate the bitrate based on the duration [20:58:29 CEST] <TikityTik> ah [20:59:17 CEST] <q3cpma> TikityTik: "-b:v $(bc <<< "($MBLIMIT * 8192) / $RUNTIME - $ARATE")K" [21:00:26 CEST] <TikityTik> q3cpma: what about resolution? [21:00:55 CEST] <q3cpma> TikityTik: you should downscale [21:01:50 CEST] <TikityTik> q3cpma: I'm wondering why that calculation doesn't include resolution. [21:02:16 CEST] <q3cpma> TikityTik: Because it has nothing to do with it when you have a filesize limit. [21:02:47 CEST] <TikityTik> oh, I didn't realize resolution didn't affect file size [21:02:48 CEST] <q3cpma> TikityTik: You will have to make your video use this bitrate, whatever its resolution is [21:11:49 CEST] <TikityTik> q3cpma: does anything besides bitrate for audio affect file size? [21:12:08 CEST] <TikityTik> q3cpma: does sampling rate affect file size? [21:12:30 CEST] <c_14> Nothing besides bitrate affects filesize. [21:12:45 CEST] <q3cpma> TikityTik: Yes it does, but at 22KhZ, you only have 11KhZ of maxium frequency [21:13:04 CEST] <TikityTik> what? [21:13:22 CEST] <q3cpma> The maxium frequency representable is sampling rate / 2 [21:13:34 CEST] <TikityTik> does qmin and qmax help a lot? [21:13:45 CEST] <q3cpma> And the maximum (theorical) of the human ear is 19-20Khz [21:13:50 CEST] <TikityTik> and are there any guides on compressing and file size? [21:14:50 CEST] <pothibo> Is it possible to serve a crossdomain.xml file from ffserver? <Redirect> is not an option as any 301/302 request gets ignroed by Flash :( [21:14:57 CEST] <q3cpma> TikityTik: I usually using qmax 55 and qmin 5. Also, you should use Opus for audio [21:15:57 CEST] <TikityTik> why Opus? [21:16:10 CEST] <q3cpma> Because it's the best lossy codec? [21:16:42 CEST] <TikityTik> why? [21:16:48 CEST] <BtbN> because it is? [21:16:58 CEST] <BtbN> Doesn't have the best software and hardware support though [21:17:24 CEST] <brontosaurusrex> TikityTik: becouse it sounds best at moderate bitrates (under 128kbps), but . [21:17:29 CEST] <q3cpma> On the desktop it's okay. [21:17:55 CEST] <q3cpma> brontosaurusrex: Actually, it beats other lossy codecs (including vorbis and musepack sv8) at every bitrate [21:18:11 CEST] <q3cpma> The difference is just more noticeable near 32-64K [21:18:24 CEST] <brontosaurusrex> q3cpma: ok [21:18:47 CEST] <q3cpma> And it's libre. [21:19:05 CEST] <q3cpma> The only problem is its shitty handling of replay gain/r128 tags [21:19:36 CEST] <brontosaurusrex> rgain tags are generally shity idea [21:19:40 CEST] <q3cpma> Why? [21:20:10 CEST] <brontosaurusrex> becous you can have external file that says -13.2 [21:20:26 CEST] <q3cpma> What? [21:20:36 CEST] <brontosaurusrex> and not trying to force that into messy metadata standards [21:20:45 CEST] <brontosaurusrex> that [21:20:50 CEST] <q3cpma> ogg metadata is pretty clean [21:20:57 CEST] <brontosaurusrex> dont care [21:21:19 CEST] <q3cpma> Metadatas are here for that [21:21:43 CEST] <brontosaurusrex> no, that was just an idea, the reality is messy [21:22:10 CEST] <kepstin> i've found that most software that plays opus and supports replaygain will use vorbis-style replaygain tags if present [21:22:15 CEST] <kepstin> so i just do that :/ [21:22:37 CEST] <q3cpma> kepstin: Well, my problem is that Rockbox use r128 tags [21:22:53 CEST] <kepstin> hmm? rockbox does *not* support opus r128gain tags... [21:23:25 CEST] <q3cpma> Sure? [21:23:45 CEST] <kepstin> if you use foobar2000 to replaygain your opus files, it incorrectly puts the album gain into a special header field rather than an R128_ALBUM_GAIN tag. [21:23:46 CEST] <q3cpma> Ah yeah. [21:23:55 CEST] <kepstin> which is then applied by rockbox [21:24:33 CEST] <q3cpma> Another problem is the lack of r128 gain calculator [21:25:22 CEST] <kepstin> eh, most replaygain software right now (aside from the older command line tools on linux) actually uses the r128 algorithm, with an adjusted reference point to be replaygain compatible [21:25:26 CEST] <brontosaurusrex> q3cpma: ffmpeg got r128 for a while now [21:25:48 CEST] <kepstin> it's a 5dB offset between the R128 and replaygain reference levels, iirc [21:25:50 CEST] <q3cpma> brontosaurusrex: it got the scanning, but I still don't know how to put into the metadata [21:26:21 CEST] <brontosaurusrex> maybe ffmpeg people knew that was a bad idea [21:26:43 CEST] <q3cpma> Why? ebur128 is more efficient than replay gain algo [21:26:58 CEST] <q3cpma> The only problem is the default loudness [21:27:24 CEST] <kepstin> but yeah, I have a local patch hanging around that adds support for the R128_*_GAIN tags to rockbox - it works by adjusting the reference then treating it as replaygain [21:27:36 CEST] <kepstin> just wrote it last week, should submit the patch I guess. [21:27:39 CEST] <brontosaurusrex> i mean puting rg tags into meta [21:28:07 CEST] <q3cpma> Why would it be a bad idea? It's useless without this function [21:28:17 CEST] <kepstin> replaygain in meta makes perfect sense; i can just copy my music files to my rockbox ipod and they're volume leveled there without having to recalculate anything... [21:28:33 CEST] <brontosaurusrex> ffmpeg audio.some -calc r128 > .audio.some.rg [21:28:47 CEST] <brontosaurusrex> play audio.some with .audio.some.rg [21:28:49 CEST] <brontosaurusrex> done [21:28:51 CEST] <kepstin> then I have to worry about keeping hidden files around and in sync [21:29:05 CEST] <q3cpma> brontosaurusrex: This is really that? I thought is was -af ebur128 [21:29:12 CEST] <kepstin> same reason as why it's nice to mux subs into video files rather than have video.srt files alongside every video.mp4... [21:29:26 CEST] <brontosaurusrex> kepstin: they are not golden, if they got lost, just recalc them [21:29:38 CEST] <q3cpma> Also, how does this work for album gain? [21:29:42 CEST] <kepstin> but if i put some tiny metadata in the file, they won't get lost... [21:29:59 CEST] <JEEB> I think the only issue with ebu r128 stuff is that we need standardized places to put them, not having the data in random tags [21:30:03 CEST] <kepstin> q3cpma, you'd probably have to do it manually by running the filter over the concatenated audio [21:30:26 CEST] <kepstin> JEEB, http://wiki.hydrogenaud.io/index.php?title=ReplayGain_2.0_specification is what most software does now. [21:30:29 CEST] <brontosaurusrex> q3cpma: actually dont know what would be the best way to handle album rg [21:30:52 CEST] <JEEB> kepstin: yeah but I'd rather have specified stuff in containers for it [21:30:58 CEST] <JEEB> rather than have it in, say, bitstream itself [21:31:11 CEST] <JEEB> which is what id3v2 tags are f.ex. [21:31:24 CEST] <q3cpma> Well, just don't use mpleb3, in this case [21:32:00 CEST] <kepstin> imo, it's nice to have the spec for gain calculations separate from the container spec, so making use of whatever generic metadata the format supports seems like a good idea [21:32:15 CEST] <q3cpma> It is [21:32:16 CEST] <JEEB> yes, naturally [21:32:21 CEST] <JEEB> which is why you have ebu r128 and stuff [21:32:31 CEST] <TikityTik> q3cpma: is there a way to see the supported sampling rates for a codec? [21:32:34 CEST] <kepstin> pretty much the only container I know of which support replaygain at the container level rather than inside the bitstream is matroska, really. [21:32:34 CEST] <JEEB> it's just how you save that information in containers [21:32:47 CEST] <JEEB> kepstin: are you sure that is not in random tags [21:32:51 CEST] <kepstin> but that's just because matroska actually has container-level generic metadata [21:33:03 CEST] <q3cpma> TikityTik: Just use opus at 64 or 96K if you need size this much, mate [21:33:27 CEST] <JEEB> because I remember mosu adding various random information into tags instead of proper container entities [21:33:34 CEST] <JEEB> lately [21:34:01 CEST] <JEEB> I'd much rather have the EBU R128 stuff be in a per-stream proper container entity [21:34:25 CEST] <JEEB> maybe after they get the first version of the Matroska mindfuck spec done we can actually start properly improving the format [21:34:29 CEST] <JEEB> like adding proper timestamps [21:34:35 CEST] <JEEB> instead of the mess that it is right now [21:34:57 CEST] <kepstin> well, one of the points of replaygain was that it worked with existing files without breaking playback in non-updated players. [21:35:11 CEST] <JEEB> just adding new entities doesn't break anything [21:35:27 CEST] <JEEB> parsers are supposed to ignore shit that they don't know [21:36:54 CEST] <kepstin> amusingly, the original replaygain spec used a custom frame type (xing header iirc?) with binary data in mp3 rather than id3 tags, but nobody uses that any more. lame still writes them, though... [21:37:40 CEST] <q3cpma> Really lame [21:37:49 CEST] <JEEB> well the issue with mp3 was that it was fucking raw bitstream [21:38:07 CEST] <JEEB> It's really nice that IIRC no other popular audio format went for that being the mainstream way of distroing it [21:38:18 CEST] <JEEB> AAC is mostly in mpeg-4 part 12 derivatives [21:38:32 CEST] <JEEB> Vorbis is in the oggwutlol, same for Opus [21:38:46 CEST] <kepstin> and of course, ogg simply doesn't have any stream metadata at the container level, yeah [21:39:10 CEST] <JEEB> depends on what you mean as metadata [21:39:17 CEST] <q3cpma> What about musepack? [21:39:47 CEST] <kepstin> musepack uses APEv2 tags [21:40:01 CEST] <JEEB> also some formats just added a seeking list in them so even if it was a "raw bit stream" it would still be seekable [21:40:19 CEST] <JEEB> mp3 is basically one of the epitomes of how not to do shit [21:40:33 CEST] <JEEB> yet it was of course popular as hell [21:40:46 CEST] <kepstin> well, next time you go and make a codec and container that will take the world by storm, make sure you get it right. [21:40:51 CEST] <q3cpma> Pleb ruins everything [21:41:04 CEST] <kepstin> in the mean time, i'll keep putting my replaygain in tags because it works. [21:41:19 CEST] <q3cpma> Of course. [21:42:06 CEST] <JEEB> you kind of misunderstood my issue :P anyways, matroska should be fix'able after it gets a proper spec, and mp4 might already have stuff [21:42:20 CEST] <JEEB> I haven't looked into it in the spec or around mpeg-4 ra [21:42:52 CEST] <kepstin> as far as i know, nothing uses the existing replaygain tags in matroska, so it shouldn't really break anything to move them to a different place where they'll still be unused :/ [21:43:36 CEST] <kepstin> apple's volume-leveling stuff in itunes is done via a tag as well, containing a bunch of undescriptive opaque numbers :/ [21:43:41 CEST] <kepstin> in aac/mp4 [21:43:52 CEST] <JEEB> yeah, there are proprietary solutions almost surely [21:44:13 CEST] <JEEB> also I'm not sure if the thing you're speaking about is a proper entity in matroska or a tag [21:44:23 CEST] <kepstin> it's a tag [21:45:07 CEST] <kepstin> http://matroska.org/technical/specs/tagging/index.html ctrl-f "replaygain" [21:45:33 CEST] <JEEB> seems so, yes [21:46:56 CEST] <JEEB> anyways, I'm not into that stuff so my biggest beef would be to get proper timestamps into matroska (and that will break backwards compat and thus version will get one-up'd - unless the old information will be added as well) [21:47:48 CEST] <JEEB> currently if you put 24000/1001 content into matroska you can't have monotonically rising timestamps because of the current way timestamps are done in the spec [21:48:18 CEST] <kepstin> it doesn't support arbitrary fractional timebases? [21:48:21 CEST] <JEEB> nope [21:48:52 CEST] <JEEB> proper timestamps were once introduced into the changes that would have gone into matroska v4 [21:48:59 CEST] <JEEB> buut then mosu got the blues [21:53:37 CEST] <TikityTik> q3cpma: does keyframe interval help with size as well? [21:53:45 CEST] <TikityTik> I still can't get my clip to go under 4 MB [21:54:05 CEST] <TikityTik> my 00:02:09 clip [21:54:11 CEST] <kepstin> increasing keyframe interval with modern codecs generally reduces size, at the expense of making seeking slower. [21:54:57 CEST] <TikityTik> so how would I do it? just with -g? [21:55:29 CEST] <q3cpma> TikityTik: Did you calculate the maxrate? [21:55:59 CEST] <TikityTik> I ended up using this http://www.dr-lex.be/info-stuff/videocalc.html [21:56:19 CEST] <TikityTik> err 00:02:28 i meant [21:56:30 CEST] <nardev> q3cpma, http://diveintohtml5.info/video.html [21:56:36 CEST] <q3cpma> TikityTik: (8 * 8192 / 148) - 64 [21:56:38 CEST] <q3cpma> Not hard [21:58:14 CEST] <TikityTik> runtime? artame? [21:58:22 CEST] <TikityTik> err, arate* [21:58:31 CEST] <q3cpma> Yeah, 64 is the audio bitrate [21:58:37 CEST] <q3cpma> That gives 190 [21:58:56 CEST] <q3cpma> Whoops [21:58:59 CEST] <TikityTik> I also noticed I got smaller sizes i think without specifying -pass 2 [21:59:40 CEST] <q3cpma> The twopass system gives more accurate sizes, that's all [21:59:45 CEST] <q3cpma> Not bigger or smaller [22:00:00 CEST] <TikityTik> what's runtime? [22:00:09 CEST] <TikityTik> in that formula that you gave me [22:00:15 CEST] <q3cpma> duration [22:00:18 CEST] <TikityTik> oh [22:00:20 CEST] <q3cpma> in seconds [22:00:40 CEST] <q3cpma> It's not rocket science [22:01:24 CEST] <TikityTik> for me it is at the moment, as I'm falling asleep in my chair lol [22:02:00 CEST] <TikityTik> also vp8 is only available for pass 2 right? [22:02:13 CEST] <q3cpma> Huh? [22:02:26 CEST] <TikityTik> because i was having troubles testing pass 1 [22:02:43 CEST] <q3cpma> I don't understand anything you're saying [22:03:02 CEST] <TikityTik> yeah, the video doesn't work with pass 1 [22:03:17 CEST] <q3cpma> Do you know how to use passes? [22:03:32 CEST] <TikityTik> you just specify if you want pass 1 or pass 2? [22:03:42 CEST] <q3cpma> No [22:03:54 CEST] <q3cpma> Look for h264 [22:03:55 CEST] <q3cpma> https://trac.ffmpeg.org/wiki/Encode/H.264 [22:06:01 CEST] <TikityTik> so you shouldn't use -crf if you're using two-pass? [22:06:31 CEST] <q3cpma> You can use crf with maxrate [22:06:48 CEST] <TikityTik> what? [22:07:14 CEST] <q3cpma> https://trac.ffmpeg.org/wiki/Encode/VP8 [22:07:25 CEST] <q3cpma> crf and -b:v [22:08:00 CEST] <q3cpma> or just -minrate and -maxrate [22:08:25 CEST] <TikityTik> which is preferred? [22:08:43 CEST] <q3cpma> >You should use constant bitrate encoding if you need your video files to have a certain size, [22:09:59 CEST] <TikityTik> q3cpma: but i read that you can make up for quality if you use cfr because of motion and still images and how compression works. [22:10:09 CEST] <TikityTik> which i still don't understand [22:10:46 CEST] <q3cpma> > This doesn't mean that every frame is encoded with the same amount of bits [22:10:51 CEST] <kepstin> the radio of quality per overall bitrate is the same between a 2-pass and a crf encode, the difference is that with a 2-pass encode you set bitrate and it picks quality, with a crf encode you set quality and it picks bitrate [22:10:55 CEST] <kepstin> ratio [22:11:09 CEST] <q3cpma> It's basically like crf with qmin and qmax really near [22:13:23 CEST] <TikityTik> alright so i specified a maxrate and minrate equaling 190 [22:13:40 CEST] <TikityTik> but it says that the bitrate is 301, i'm guessing partly from the audio? [22:14:22 CEST] <q3cpma> 190 + 64 != 301 [22:14:27 CEST] <q3cpma> There's a problem [22:15:54 CEST] <TikityTik> is it because i need -bufsize? [22:16:08 CEST] <q3cpma> I don't know [22:16:11 CEST] <q3cpma> What's your command [22:16:30 CEST] <TikityTik> ffmpeg -i out2.mkv -c:v libvpx -c:a libopus -b:v 190k -minrate 190k -maxrate 190k -b:a 64k -ac 1 -vf scale=-1:480 -pass 2 out.webm [22:16:59 CEST] <q3cpma> And the first pass? [22:17:25 CEST] <TikityTik> it doesn't work, it it's not encoding the video [22:17:31 CEST] <TikityTik> if you mean -pass 1 [22:17:46 CEST] <TikityTik> video:0kB audio:929kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 4.801717% [22:18:52 CEST] <q3cpma> No, the command [22:18:56 CEST] <q3cpma> It's the same, right? [22:19:15 CEST] <TikityTik> for -pass 1? [22:19:39 CEST] <TikityTik> ffmpeg -i out2.mkv -c:v libvpx -c:a libopus -b:v 190k -minrate 190k -maxrate 190k -b:a 64k -ac 1 -vf scale=-1:480 -pass 1 out.webm [22:22:08 CEST] <q3cpma> And it doesn't work? [22:22:18 CEST] <TikityTik> no [22:22:26 CEST] <q3cpma> What's the error? [22:23:04 CEST] <TikityTik> there's no error, it just shows that nothign is being encoded for the video part [22:23:20 CEST] <TikityTik> i think vp8 doesn't support first pass [22:23:29 CEST] <q3cpma> Yes it does, I'm using it right now [22:23:46 CEST] <q3cpma> I usually put "-f webm /dev/null" instead of "out.webm" for the first pass [22:24:34 CEST] <TikityTik> i don't have /dev/null on windows [22:24:43 CEST] <q3cpma> It's NULL, I think [22:25:01 CEST] <TikityTik> yeah nothing for the video [22:25:19 CEST] <q3cpma> I'll give you my command, and you'll try with it [22:25:52 CEST] <TikityTik> k [22:25:57 CEST] <q3cpma> ffmpeg -ss 01:06:53.300 -i The\ End\ of\ Evangelion\ \[1004x574\]\ \[High10(a)L4.1\].mkv -t 120 -c:v libvpx -quality good -slices 4 -minrate 209K -maxrate 209K -b:v 209K -c:a libopus -ac 2 -b:a 64K -sn -pass 1 -f webm /dev/null [22:26:13 CEST] <q3cpma> Then ffmpeg -ss 01:06:53.300 -i The\ End\ of\ Evangelion\ \[1004x574\]\ \[High10(a)L4.1\].mkv -t 120 -c:v libvpx -quality good -slices 4 -minrate 209K -maxrate 209K -b:v 209K -c:a libopus -ac 2 -b:a 64K -sn -pass 2 -f webm out.webm [22:27:41 CEST] <TikityTik> i took out the -ss and -t and made it input out2.mkv, also had to change the /dev/null, there's still no video encoding [22:27:46 CEST] <TikityTik> for the first one [22:28:01 CEST] <q3cpma> That's normal [22:28:09 CEST] <q3cpma> The first pass only produce a log [22:29:12 CEST] <TikityTik> the second one is showing a bitrate of 236 [22:29:22 CEST] <TikityTik> and 270 [22:29:52 CEST] <q3cpma> During the encoding [22:29:57 CEST] <q3cpma> You'll see at the end [22:29:57 CEST] <TikityTik> yes [22:30:22 CEST] <q3cpma> Mine produce a 3.9MB file, so it's perfect [22:30:29 CEST] <q3cpma> Except that VP8 is shit, of course [22:31:05 CEST] <TikityTik> i think you mispelled the -slices option [22:31:13 CEST] <TikityTik> nvm [22:31:15 CEST] <q3cpma> It works for me [22:31:48 CEST] <TikityTik> the size ended up being 5109kB [22:31:56 CEST] <TikityTik> bitrate of 326 kbs [22:32:11 CEST] <q3cpma> That's kind of normal [22:32:16 CEST] <TikityTik> my duration is 128 though [22:32:17 CEST] <q3cpma> We don't use the same duration, right? [22:32:23 CEST] <TikityTik> 8 seconds difference [22:32:39 CEST] <q3cpma> Well, 326 is too high [22:32:54 CEST] <TikityTik> but why can I not achieve CBR? [22:32:59 CEST] <q3cpma> Hey, it's maybe because of -ac 1 [22:35:37 CEST] <TikityTik> ffmpeg -i out2.mkv -c:v libvpx -c:a libopus -b:v 190k -minrate 190k -maxrate 190k -b:a 64k -vf scale=-1:480 -bufsize 1835k -pass 2 out.webm [22:35:47 CEST] <TikityTik> frame= 3068 fps= 30 q=-0.0 Lsize= 3979kB time=00:02:08.03 bitrate= 254.6kbits/s [22:36:18 CEST] <q3cpma> Looks okay to me [22:36:32 CEST] <TikityTik> but why isn't CBR happening? [22:36:40 CEST] <q3cpma> 190 + 64 does equal 254 [22:36:47 CEST] <q3cpma> What do you mean? [22:36:55 CEST] <TikityTik> oh... [22:37:25 CEST] <TikityTik> i think -bufsize helped though [22:38:13 CEST] <q3cpma> Actually, you're right [22:38:19 CEST] <q3cpma> maxrate requires bufsize to be set [22:38:46 CEST] <TikityTik> how do you find the optimal bufsize? trial and error? [22:39:27 CEST] <q3cpma> The bigger the better (I think) [22:44:08 CEST] <TikityTik> I get better quality with mono [22:44:36 CEST] <q3cpma> Well, of course. You have 32K per channel in stereo [22:49:12 CEST] <q3cpma> TikityTik: If you have a good CPU, I suggest you use libvpx-vp9. [22:49:30 CEST] <TikityTik> I would, but the site doesn't allow vp9 [22:49:38 CEST] <q3cpma> What site? [22:49:52 CEST] <TikityTik> ughh... [22:50:10 CEST] <TikityTik> here's a hint https://i.4cdn.org/wsg/1437838366379.webm [22:50:16 CEST] <q3cpma> >cuckchan [22:50:40 CEST] <TikityTik> lol [22:50:44 CEST] <q3cpma> Well, anyway, now you can do webm. We have mp4 on real chans [22:51:05 CEST] <TikityTik> yeah vp8 hurts [22:51:21 CEST] <q3cpma> vp9+opus or mp4+aac on 8chan [22:51:29 CEST] <q3cpma> Maybe theora soon [22:52:32 CEST] <TikityTik> why can I not change the sampling rate for opus anyhow? [22:53:04 CEST] <q3cpma> Because that's how opus is made [22:59:02 CEST] <unknown23> hi, I am writing a demuxer for a new file format, in my "read_packet" function, I want to modify the underlying data a bit, so I really can't use "av_get_packet". how do I get my modified data to fit in "pkt"? [23:00:51 CEST] <TikityTik> q3cpma: so there isn't a way to increase quality for the scenes with motion and lower quality for the still scenes? [23:01:32 CEST] <TikityTik> that's what i got from this: http://slhck.info/articles/crf [23:02:08 CEST] <JEEB> unknown23: for development within lavc/lavf and friends, feel free to use #ffmpeg-devel ;) [23:02:21 CEST] <JEEB> which reminds me I should poke my WIP demuxer as well [23:02:37 CEST] <unknown23> JEEB, thanks, doing that now. [23:03:02 CEST] <q3cpma> TikityTik: Like it's said in the wiki, constrained != constant [23:03:16 CEST] <TikityTik> but it really helps with quality right? [23:05:26 CEST] <q3cpma> TikityTik: If you have abig enough bufsize, yeah. [23:07:10 CEST] <q3cpma> TikityTik: https://trac.ffmpeg.org/wiki/Encode/VP9 it's different with vp9 [23:07:32 CEST] <TikityTik> which i can't use [23:07:39 CEST] <q3cpma> TikityTik: Why? [23:07:49 CEST] <TikityTik> cuckchan doesn't allow vp9 [23:07:57 CEST] <q3cpma> Oh yeah, forgot [23:08:00 CEST] <q3cpma> Hehe [23:20:34 CEST] <TikityTik> q3cpma: https://i.4cdn.org/wsg/1437859220016.webm [23:25:16 CEST] <q3cpma> TikityTik: There's no sound? [23:26:14 CEST] <TikityTik> there is for me [23:27:33 CEST] <q3cpma> TikityTik: Strange, it doesn't for me in the browser [23:30:40 CEST] <h1113> Hello, did anyone ever try to use two TCP inputs at the same time? [23:30:53 CEST] <h1113> It seems like it completely ignores the second tcp input. [23:32:01 CEST] <h1113> ffmpeg -i tcp://localhost:15014 -i tcp://localhost:15015 out.mp4 [23:32:23 CEST] <TikityTik> q3cpma: apparently cuckchan doesn't accept opus either [23:32:27 CEST] <TikityTik> so i had to switch to vorbis [23:32:49 CEST] <q3cpma> TikityTik: I mean I download the webm and tried to open it in firefox and still no sound. [23:33:48 CEST] <TikityTik> you're missing vorbis? I'm not sure [23:34:42 CEST] <maletor> Is there a way to jump to the frame where a subtitle track is first used? [23:39:59 CEST] <h1113> Could anyone try on their end to use two TCP connections as input? [00:00:00 CEST] --- Sun Jul 26 2015

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ffmpeg-devel.log.20150724
by burek 25 Jul '15

25 Jul '15

[00:00:19 CEST] <durandal_1707> jamrial: from ducker, duck is also audacity effect [00:01:03 CEST] <jamrial> ah [00:01:08 CEST] <jamrial> well, if more people recognize what the filter does with that name then go for it :p [00:53:13 CEST] <cone-170> ffmpeg 03Steve Lhomme 07master:a54e720e0289: configure: force -nologo- when detecting MSVC [00:53:13 CEST] <cone-170> ffmpeg 03Michael Niedermayer 07master:da8b70b810bb: Merge commit 'a54e720e0289433d6bc3f7ba0a37fa5cabfaeea9' [04:41:19 CEST] <cone-562> ffmpeg 03Michael Niedermayer 07master:5da90d7ec3a3: avcodec/hapdec: Check that there is sufficient input data [10:03:35 CEST] <durandal_1707> if nobody minds I will push areverse and sidechaincompress filters [10:12:12 CEST] <nevcairiel> man the gst-ffmpeg code is so ugly [10:12:16 CEST] <nevcairiel> no wonder gstreamer sucks [10:15:29 CEST] <wm4> link? [10:15:38 CEST] <wm4> (not easy to track that down with all those plugins) [10:15:43 CEST] <nevcairiel> http://cgit.freedesktop.org/gstreamer/gst-ffmpeg/tree/ext/libav/gstavauddec… [10:15:48 CEST] <nevcairiel> they did use the api wr ong [10:15:53 CEST] <nevcairiel> but they fixed it a month ago [10:16:00 CEST] <nevcairiel> avframe on the stack, no free call, ever [10:16:20 CEST] <BtbN> Their vaapi code is also very "interesting". [10:16:38 CEST] <BtbN> I wasn't able to understand how it works within a full evening of reading through it. [10:17:07 CEST] <wm4> there's decidedly too much G in that code [10:18:28 CEST] <nevcairiel> i wish there was a way to have the compiler throw a big fat warning when avframe is used on the stack [10:18:57 CEST] <wm4> you could just add something at the end of AVFrame that makes it not stack allocatable [10:19:12 CEST] <wm4> but it'd have to be ifdeffed out for libavutil internal use [10:26:25 CEST] <nevcairiel> avutil doesnt use it on the stack either, does it? The closest it comes is using the deref operator to copy all its contents [10:29:08 CEST] <wm4> hm, my standard C idea doesn't work anyway (using flexible array members or what is was called) [10:33:28 CEST] <wm4> hm, you could add a trailing array that is SIZE_MAX/2 big [10:33:42 CEST] <wm4> gcc accepts this, but will error if you allocate that on the stack [10:34:07 CEST] <wm4> but clang won't [11:17:40 CEST] <rcombs> break API; add accessors; move struct private [11:18:18 CEST] <nevcairiel> accessors are so annoying [13:03:11 CEST] <cone-609> ffmpeg 03Ivan Uskov 07master:c90dbc67ed6b: libavcodec/qsvdec.c: The ff_qsv_decode() now guarantees the consumption of whole packet. [13:13:56 CEST] <cone-609> ffmpeg 03Ivan Uskov 07master:0b159e3b65d5: libavcodec/qsvdec_h264.c: packet buffering has been removed since qsvdec.c does maintain own data buffering now. [13:19:18 CEST] <nevcairiel> hm who was it that was working on the braindead get_key method in ffmpeg.c for vs2015 some time ago? [13:19:27 CEST] <nevcairiel> i should get better long term storage in my brain [13:22:44 CEST] <AlicanC> I need detailed information on filters. I think the online docs are generated from .texi files in doc/. How are .texi files are made? By hand or is there something that generates them from the source code? [13:23:47 CEST] <AlicanC> By "detailed information", I mean for programmatic usage. [13:25:59 CEST] <wm4> read the source code [13:26:13 CEST] <wm4> the texi files are manually edited and usually contain general information [13:32:22 CEST] <kierank> nevcairiel: "no codecs for which we're GUARANTEED to have better alternatives" [13:32:23 CEST] <kierank> wtf [13:32:32 CEST] <nevcairiel> wat [13:32:32 CEST] <kierank> in gstreamer [13:32:37 CEST] <nevcairiel> oh [13:33:07 CEST] <nevcairiel> i wonder if these dedicated decoding libs are actually better [13:33:10 CEST] <nevcairiel> or just gst nonsense [13:45:58 CEST] <wm4> maybe ask BBB [13:48:47 CEST] <iive> they have their own alternatives for every codec in ffmpeg?! [13:50:55 CEST] Action: J_Darnley suggests that they just ripped codecs out of libavcodec and call them "alternatives" [14:17:05 CEST] <durandal_1707> hmm, they have own vp9? [14:17:35 CEST] <cone-609> ffmpeg 03Paul B Mahol 07master:591741b51d4f: avfilter: add areverse filter [14:17:36 CEST] <cone-609> ffmpeg 03Paul B Mahol 07master:44fb00866f9b: avfilter: rename vf_reverse.c to f_reverse.c [14:18:58 CEST] <kierank> durandal_1707: libvpx i sassume [14:36:36 CEST] <BBB> wm4: ? [14:36:38 CEST] <BBB> what? [14:37:32 CEST] <wm4> see gstreamer discussion above [14:37:42 CEST] <wm4> <nevcairiel> i wonder if these dedicated decoding libs are actually better [14:37:42 CEST] <wm4> <nevcairiel> or just gst nonsense [14:38:06 CEST] <BBB> dedicated decoding libs? [14:38:08 CEST] <BBB> like what [14:38:18 CEST] <BBB> libvpx instead of ffvp9 and dehevc instead of ffhevc? [14:38:38 CEST] <ubitux> huge flood of "Past duration 0.673149 too large" (growing) [14:38:38 CEST] <BBB> libabcxyz instead of ffabcxyz? [14:38:46 CEST] <ubitux> when transcoding from mp4 to webm [14:38:49 CEST] <ubitux> known prb? [14:39:01 CEST] <ubitux> after 1 sec it disappears [14:39:36 CEST] <BBB> wm4: got more info? I would have to look into it, right now I can only guess [14:40:23 CEST] <wm4> we were guessing based on the contents of this file http://cgit.freedesktop.org/gstreamer/gst-ffmpeg/tree/ext/libav/gstavauddec… [14:40:32 CEST] <wm4> it says "no codecs for which we're GUARANTEED to have better alternatives" somewhere [14:40:46 CEST] <nevcairiel> There must be some reason they somehow prefer all these external libs over avcodec [14:40:58 CEST] <nevcairiel> Although stupidity is a reason I would accept [14:41:19 CEST] <BBB> ah I see [14:41:28 CEST] <BBB> so, they have this thing called gst-plugins-xyz [14:41:35 CEST] <BBB> where xyz is base, good, bad, ugly, ffmpeg, etc. [14:41:44 CEST] <BBB> youre assumed to have base and good installed [14:41:48 CEST] <BBB> without them, nothing works [14:42:08 CEST] <BBB> they guarantee that good/base are patent-unencumbered, free (lgpl or better) and high quality [14:42:13 CEST] <BBB> whatever that means [14:42:25 CEST] <nevcairiel> Like any of these codecs is patent free [14:42:32 CEST] <BBB> so, they apparently claim that the performance of these decoders is also better than anything ffmpeg/libav could ever provide [14:43:02 CEST] <BBB> (I dont know if they measured it, likely they didn't) [14:43:08 CEST] <BBB> (but maybe they did) [14:43:19 CEST] <BBB> so they likely claim libvorbis > ffvorbis [14:43:23 CEST] <BBB> etc. [14:43:54 CEST] <BBB> if you disagree, you could consider putting out decoder performance stats that show otherwise, I dont know how our audio codecs compare to theirs [14:44:14 CEST] <kierank> opus should be faster in ffmpeg [14:44:16 CEST] <BBB> for subtitles, they likely wrote their own which actually are integrated in the rest of the framework [14:44:16 CEST] <nevcairiel> Clearly for encoding that's true, but decoding the ff codecs tend to beat the external libs eventually [14:44:25 CEST] <BBB> kierank: opus is not in that list [14:44:38 CEST] <BBB> nevcairiel: wed have to show numbers, blind claims dont help anyone [14:45:04 CEST] <BBB> we showed ffvp9>libvpx and ffvp8>libvpx [14:45:12 CEST] <BBB> wed have to show that ffvorbis>libvorbis [14:45:56 CEST] <BBB> for subtitles I have no idea, it tends to not be limiting, so they might just want to use base plugins and not do effort integrating ext plugins, if you see what I mean [14:46:09 CEST] <BBB> since theyre unlikely to be patent-encumbered and performance doesnt matter much [14:47:14 CEST] <BBB> but from the ranking it looks like they prefer ext libs anyway [14:47:21 CEST] <BBB> default: [14:47:22 CEST] <BBB> rank = GST_RANK_MARGINAL; [14:47:45 CEST] <BBB> that seems like in most cases itd use ext libs if installed (e.g. libvpx over ffvp9, etc.), which is not good for users [14:47:47 CEST] <BBB> but what do I know [14:50:13 CEST] <BBB> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/ext/vpx/plugin.c -> all libvpx plugins are primary [15:00:28 CEST] <wm4> did they make the same mistake as dshow, and allow plugins to "take over"? [15:00:38 CEST] <wm4> because they specify their own prioritx [15:00:39 CEST] <wm4> *y [15:05:50 CEST] <mateo`> BBB: maybe this is something that could discussed with them, i've been working on the project quite a bit and i've never heard such claim [15:16:54 CEST] <BBB> wm4: yes [15:17:12 CEST] <BBB> wm4: although in this case they create the plugins themselves so its not as much of a viral-y issue [15:17:21 CEST] <BBB> mateo`: what kind of claim? [15:37:30 CEST] <BBB> mateo`: also note that they use libav [15:39:26 CEST] <mateo`> BBB: that external libs are > lavc [15:40:20 CEST] <Daemon404> for which? [15:40:21 CEST] <mateo`> BBB: they use libav when you build locally gst-plugins-libav, but you can make it build with the system libs which are usually ffmpeg on most distros [15:40:39 CEST] <Daemon404> i bet they still claim shit like libdca is better [15:40:40 CEST] <Daemon404> or libmad [15:41:07 CEST] <mateo`> Daemon404: what i said is that i've never heard such claim [15:41:12 CEST] <BBB> Daemon404: they do [15:41:18 CEST] <Daemon404> mateo`, o [15:41:18 CEST] <BBB> mateo`: the plugin ranking is higher [15:41:36 CEST] <BBB> that defines which resource will be used for automatic decoding, which is 99% of all cases (e.g. decodebin, playbin, etc.) [15:41:46 CEST] <mateo`> BBB: i know [15:41:46 CEST] <BBB> nobody manually changes these things [15:41:55 CEST] <Daemon404> anyway, afaict the only reason gst exists these days is so big corps and consultants can do stuff with it [15:41:58 CEST] <BBB> so if mad priority > ff priority, it will use mad for decoding [15:41:59 CEST] <BBB> mad is primary [15:42:01 CEST] <Daemon404> and have proprietary (terrible) plugins [15:42:01 CEST] <BBB> ff is marginal [15:42:05 CEST] <BBB> so it uses mad, not ff [15:42:10 CEST] <mateo`> BBB: all i say is that this is something that can be discussed [15:42:13 CEST] <BBB> which is bad for users if ff is actually better [15:42:37 CEST] <BBB> mateo`: ok& so who do we discuss this with? I mean, Im not a political dude, I just code [15:43:59 CEST] <mateo`> BBB: maybe i can (i'm not a political dude either), i can at least ask why it is done like that [15:44:56 CEST] <Daemon404> $5 says you get an unsubstantiated claim that external lib is better [15:45:09 CEST] <BBB> mateo`: sgtm, tnx [15:45:36 CEST] <BBB> mateo`: Id give an example of ffvp8 or ffvp9 being substantially faster than libvpx, yet their priority being lower [15:46:05 CEST] <BBB> (since for that we have sufficiently much evidence) [16:09:14 CEST] <cone-609> ffmpeg 03Paul B Mahol 07master:f0489a35c034: avutil: add ayuv64le and ayuv64be packed pixel format [16:09:15 CEST] <cone-609> ffmpeg 03Paul B Mahol 07master:052f64ecb2f1: swscale: ayuv64le input support [16:09:16 CEST] <cone-609> ffmpeg 03Paul B Mahol 07master:3cb8eee6f7bf: swscale: ayuv64le output support [17:00:39 CEST] <ubitux> is it safe to change avctx->skip_frame in the middle of the decoding? [17:00:49 CEST] <ubitux> i mean is it supposed to be supported? [17:00:56 CEST] <ubitux> (without re-opening) [17:01:35 CEST] <ubitux> (between 2 decode calls) [17:03:27 CEST] <wm4> ubitux: mplayer and mpv do it [17:03:44 CEST] <ubitux> ah? in case of lagging? [17:03:59 CEST] <wm4> yeah, for changing framedrop mode at runtime [17:04:09 CEST] <ubitux> great, exactly what i was willing to do [17:04:16 CEST] <ubitux> and so no codec re-open? [17:04:26 CEST] <wm4> whether it works as intended with multihtreading and all is a different question [17:04:30 CEST] <wm4> yes, no reopening [17:04:57 CEST] <ubitux> ok [17:05:05 CEST] <ubitux> thanks :) [17:05:16 CEST] <ubitux> just curious, what do you decide to drop? [17:05:31 CEST] <wm4> a really bad heuristic [17:05:38 CEST] <wm4> which never works well [17:06:15 CEST] <ubitux> seems you decide to drop nonref, ok [17:06:35 CEST] <wm4> that's probably the only thing that doesn't cause corruption [17:20:50 CEST] <durandal_1707> does anybody used framerate filter from ffmbc? [17:22:42 CEST] <Compnn> whats it do ? [17:22:50 CEST] <Compnn> drop / duplicate ? [17:47:45 CEST] <cone-609> ffmpeg 03Michael Niedermayer 07master:daf6bce71be9: avcodec/dvdec: Retry decoding seemingly damaged MBs while skiping likely damaged parts [18:00:13 CEST] <durandal_1707> Compnn: it duplicates drop but also blends [18:00:53 CEST] <durandal_1707> I ported it to New API but it is not working properly [18:06:27 CEST] <TimNich> durandal_1707: is that the oneMArk Himsley wrote? [18:06:50 CEST] <durandal_1707> Yes [18:07:22 CEST] <TimNich> I might use it if it worked properly ;) [18:12:15 CEST] <TimNich> What about it is not working properly? [18:14:21 CEST] <durandal_1707> it picks wrong frame to duplicate, goes into past too much dunno why... [18:17:11 CEST] <TimNich> ISTR there is something different about the framerate handling in ffmbc. I had different behavour using another filter becasue of that. [18:25:02 CEST] <durandal_1707> what filter? [18:26:12 CEST] <kierank> durandal_1707: email mark and ask :) [18:26:48 CEST] <TimNich> I think the w3dif or maybe tinterlace, or a combination [18:28:21 CEST] <TimNich> Mark ahd a filter set that d-ineterlaced. scaled then reinterlaced which worked in bmc but mot mpeg [18:28:30 CEST] <TimNich> *mbc [18:30:03 CEST] <TimNich> ffmbc set the framrate to double then back again somewhere int efilter chain, but ffmpeg didnt [18:30:18 CEST] <TimNich> might be irreleveant oc [19:01:23 CEST] <durandal_1707> actually it was bug in patch [19:12:37 CEST] <philipl> BtbN: gstreamer and vaapi seems like a match made in complexity heaven. [19:12:55 CEST] <philipl> All I know is it's a great way to burn a ton of CPU doing hardware decoding. [19:15:44 CEST] <j-b> Paul, can your framerate filter to frame-doubling? [19:22:58 CEST] <durandal_1707> j-b: it can do this, also adds blending frames, and it's mark himsley one ,not mine [19:23:27 CEST] <j-b> your patch :) [19:23:45 CEST] <j-b> durandal_1707: are you aware of SVP? [19:25:09 CEST] <durandal_1707> What? [19:26:05 CEST] <j-b> http://www.svp-team.com/ [19:29:25 CEST] <durandal_1707> yes I heard of it but didn't looked at their code [19:29:56 CEST] <j-b> but technically, we could do the same, with your filter, right? [19:32:03 CEST] <TimNich> Marks filter is more basic. its just a frame blend % eise based on agjacent frames. [19:33:47 CEST] <TimNich> gooder video had a motion vector frame interpolatot that worked realy well. You could super slo mo and every frame was different even on something like a 40:1 slow down [19:35:53 CEST] <TimNich> There is alos another GPL one whose name escapes me that I built q while back.https://github.com/slowmoVideo/slowmoVideo/wiki [19:36:10 CEST] <JEEB> mpv also has one in their video renderer [19:38:08 CEST] <j-b> the important part is to do frame-doubling [19:54:52 CEST] Action: michaelni has some unfinished frame interpolation code too, i should update it to HEAD and post it i guess [19:55:40 CEST] <TimNich> A really good MV interpolator for frame rate conversion would be a useful tool. Commercial variants licences have too many 0 on the end. [19:57:14 CEST] <michaelni> what i have currently works for some parts of scenes quite nice but for others breaks down really bad, its not really finished [19:59:15 CEST] <TimNich> Certainly its non trivial, did you look at SlomoVideo? it uses ffmpeg itself [19:59:36 CEST] <j-b> Well, for vlc, I see request of SVP-like every other day [19:59:45 CEST] <j-b> and they just want doubling or quadrupling [20:00:02 CEST] <TimNich> Well that would be a start [20:03:24 CEST] <nevcairiel> Those things are all terrible, but for some reason users still want them, no matter the strong image artifacts [20:03:41 CEST] <michaelni> TimNich, didnt look at SlomoVideo, i probably should [20:05:41 CEST] Action: TimNich is going to brave the multiple accidents between here and home [20:11:20 CEST] <cone-609> ffmpeg 03Steve Lhomme 07master:c9edbe4af901: use a wrapper script to call MS link.exe to avoid mixing with /usr/bin/link.exe [20:11:21 CEST] <cone-609> ffmpeg 03Steve Lhomme 07master:58ed7b632842: use a wrapper script to call MS link.exe to avoid mixing with /usr/bin/link.exe [20:11:22 CEST] <cone-609> ffmpeg 03Michael Niedermayer 07master:dae79a185fa5: Merge commit 'c9edbe4af901e9bc9f05a62319637f9760df9a4a' [20:12:14 CEST] <BBB> am I the only one who finds it confusing to often find two commits of the same thing in our tree? [20:12:25 CEST] <BBB> (I suppose one is from ffmpeg and one is from the libav merge?) [20:12:41 CEST] <nevcairiel> Yes [20:13:18 CEST] <nevcairiel> Sometimes I don't understand why it isn't just merged, but instead committed separately before [20:14:32 CEST] <BBB> cant we just skip the merge if we already have the same patch? [20:14:42 CEST] <BBB> like in git rebase you have a skip option [20:14:58 CEST] <BBB> like forget about this one and then its just forgotten in the rebase, as if it never existed [20:15:01 CEST] <BBB> cant merge do that too? [20:15:27 CEST] <jamrial> BBB, nevcairiel: afaik, it's to keep the sha1 values of commits from the libav side working on ours as well [20:16:03 CEST] <jamrial> so if you read an old email mentioning commit abcdef12 from libav, it will work if you check for it in the ffmpeg tree [20:20:38 CEST] <wm4> BBB: IMO it would be better to cherry-pick [20:21:03 CEST] <wm4> but maybe mini doesn't want to be accused of "manipulating" Libav patches or something [20:21:09 CEST] <BBB> j-b: can I come to disney just in the afternoon? [20:21:23 CEST] <wm4> (which is justified, because kindergarten fights between the 2 projects) [20:21:27 CEST] <BBB> j-b: (I dont know how that would work logistically, but I dont think I can do morning) [20:21:44 CEST] <BBB> wm4: :D [20:27:29 CEST] <nevcairiel> jamrial: how would simply merging it without a separate commit on our side break that [20:28:56 CEST] <jamrial> what do you mean with separate patch? what looks like a duplicate, or the "merge" commit? [20:30:26 CEST] <jamrial> if the former, that'd be because the patch was commited to ffmeg first, then on libav and merged (like when someone sends the patch to both ml) [20:31:59 CEST] <nevcairiel> yes, exactly, why would it be commited first if its known to be merged a second later [20:32:04 CEST] <nevcairiel> its just extra noise [20:32:08 CEST] <nevcairiel> but thats just me [20:44:09 CEST] <ubitux> about frame interpolation thing, remember that we can export MVs from codecs now, and use them in filters [20:44:23 CEST] <ubitux> that might be useful as "fast & ugly" option [20:46:35 CEST] <kierank> SVP is batshit insane [20:46:38 CEST] <jamrial> ubitux: not sure if you noticed it but your fate clients haven't run for like three days [20:46:42 CEST] <kierank> why not upscale all your videos to 4k as well [20:47:01 CEST] <ubitux> jamrial: oh damn haven't noticed; will have a look, thx [20:47:28 CEST] <ubitux> the thread instance seems in a dead lock. [20:48:20 CEST] <ubitux> anyway, should be fixed, hopefully [20:48:37 CEST] <jamrial> cool [20:49:55 CEST] <ubitux> kierank: how is it insane? [20:50:11 CEST] <kierank> it's watching computer generated data [20:50:28 CEST] <kierank> and is not what the director (who may well be stupid) intended [20:50:36 CEST] <kierank> could change the colours too [20:50:41 CEST] <ubitux> there are many use cases you know [20:51:00 CEST] <ubitux> like simulating a cheap slowmo on your vacation exploits [20:51:09 CEST] <ubitux> taken on a dumb phone [20:51:22 CEST] <nevcairiel> thats not what people want it for though [20:51:37 CEST] <ubitux> > many use cases [20:51:39 CEST] <ubitux> ;) [20:52:27 CEST] <ubitux> can be used the other way around, by dropping and interpolating to avoid lag effect [20:52:45 CEST] <ubitux> playing with speed is something many users fancy [20:53:10 CEST] <ubitux> might be more relevant to look @ http://slowmovideo.granjow.net/ though [21:28:33 CEST] <cone-609> ffmpeg 03James Almer 07master:ede590c84d73: avcodec/aacsbr: add missing header include [21:34:31 CEST] <durandal_1707> ubitux: I already talked privately about diff filter [21:35:02 CEST] <ubitux> oh ok [21:35:06 CEST] <ubitux> so that was a rage quit? [21:38:50 CEST] <wm4> lol [21:39:03 CEST] <wm4> could be, but doesn't have to [22:16:57 CEST] <cone-609> ffmpeg 03Hendrik Leppkes 07master:0c0cd34f9c7d: configure: fix LARGEADDRESSAWARE flag with MSVC [22:58:28 CEST] <cone-609> ffmpeg 03Shivraj Patil 07master:e21b090bfb7b: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for VP9 intra functions [00:00:00 CEST] --- Sat Jul 25 2015

1 0

ffmpeg.log.20150724
by burek 25 Jul '15

25 Jul '15

[02:30:45 CEST] <dirty_d> lenscorrection is the best filter to use to correct for gopro fisheye right? [03:19:55 CEST] <ig0r_> I get an error message what did I do wrong? http://pastebin.com/KMGCJMGt [03:23:20 CEST] <klaxa> you must use -c copy not only copy [03:24:33 CEST] <dirty_d> c:v is all lonely too [03:25:35 CEST] <ig0r_> I want to join a video stream and an audio stream into an mp4 file. please tell me how :) [03:27:40 CEST] <ig0r_> english is not my first language so if someone could just tell me the correct commands please :) [03:28:48 CEST] <dirty_d> -c:v copy [03:28:52 CEST] <dirty_d> you have them separated [03:29:45 CEST] <ig0r_> I don't know what you mean :( [03:30:43 CEST] <dirty_d> ffmpeg.exe -i v:\video.h264 -i v:\audio.ac3 -c:v copy -c:a copy v:\result.mp4 [03:30:48 CEST] <dirty_d> try that [05:54:05 CEST] <podman> Does anyone have any experience setting up WideVine or PlayReady or other DRM? [08:27:49 CEST] <QQ> Hello. I have a question. I'm using the concat demuxer to concatenate files. Everything works perfect, except I don't know how to keep the metadata of the files. The command I'm using is "ffmpeg -f concat -i list.txt -c copy destination.mp3" (also tried flac files, but without the "-c copy" part). The files are merged, but even though they all have the same metadata not metadata at all is kept. How can I do that? [08:27:51 CEST] <QQ> The content of my list.txt file are "ffconcat version 1.0" (first line), and then for each file I set filename, duration, inpoint and outpoint. [08:54:01 CEST] <k_sze[work]> Wasn't there an ffmpeg option to output some stats and CRC about the frames, instead of actually getting pixel data? I can't remember what it is. [08:54:15 CEST] <k_sze[work]> And I can't quite find it in the man page either. [09:35:24 CEST] <QQ> I found out that I can use "-i filewithmetadata.flac -map_metadata:g:0 1:g", and it works with the flac file; it retains the metadata from the original file. But it doesn't work with mp3. Does anyone know why? [09:43:50 CEST] <durandal_1707> k_sze[work]: -vf showinfo ? [10:00:32 CEST] <QQ> I got everything to work. Sorry for bothering. [10:04:35 CEST] <durandal_1707> QQ: how? [10:05:23 CEST] <Mista_D> can someone point to an instruction for mfx [10:08:38 CEST] <Mista_D> http://pastebin.ca/3072944 - keep getting configure error: "ndefined reference to `MFXInit'" [10:38:35 CEST] <raven737> Hi, quick question, what are the _g files when i make (build) the source? [10:41:03 CEST] <JEEB> in case you are building so that you are stripping debug symbols, the _g binaries still have them [10:41:14 CEST] <JEEB> if you disable stripping then the final binaries will have them [10:46:18 CEST] <raven737> JEEB: thank you! [11:17:14 CEST] <martijnB> I think I'm misunderstanding something, I'm trying to set up ffserver with HLS, after trying to configure the stream in the .conf (ie. setting Format hls, AVOptionVideo hls_time 10) the server won't launch reporting errors like Option not found: 'hls_time' [11:17:54 CEST] <martijnB> I thought that was the right place to configure it, and the right option, I can't figure out why it would come up as not found [12:52:27 CEST] <pragomer1> hello. what is the best ppa for ffmpeg for ubuntu 14.04 ? [18:15:03 CEST] <benbro1> can ffmpeg use pcap or rtpdump files? [18:25:34 CEST] <durandal_1707> benbro1: currently not [18:27:12 CEST] <benbro1> durandal_1707: thanks [18:27:27 CEST] <benbro1> durandal_1707: can I pipe rtp/rtcp packets instead of using udp? [19:02:00 CEST] <durandal_1707> benbro1: dunno [19:21:53 CEST] <ChocolateArmpits> I started reading UHD Guidebook by Harmonic and about Rec709 it mentions "Current television and Blu-ray standards limit maximum brightness to 100 nits and minimum brightness to 0.117 nits" [19:21:59 CEST] <ChocolateArmpits> Does anyone what those 100 nits are about ? [19:22:24 CEST] <ChocolateArmpits> You can find the doc at this link: http://www.harmonicinc.com/sites/default/files/Harmonic_Complete_UHD_Guideb… [19:30:05 CEST] <JEEB> ChocolateArmpits: I will guess it's a roundabout way of saying that you have to use limited range, 8bit and BT.709 YCbCr [19:30:18 CEST] <JEEB> because after that it starts talking about BT.2020 YCbCr [19:31:26 CEST] <ChocolateArmpits> JEEB: But it's rather specific. I thought maybe the 100 nits are related to tests done to determine optimum transform in common viewing environments ? [19:32:04 CEST] <JEEB> no idea, as far as I know the specs don't say jack shit about that. All they care is that you're within limited range and are using BT.709 YCbCr colorimetry [19:35:27 CEST] <ChocolateArmpits> Well yeah, because nits are not a digital measurement [20:51:14 CEST] <Mista_D> what's the quick option to map first four audio tracks even if the source has only two? [20:53:30 CEST] <JEEB> I would recommend using a probing pass first and building your command line on that [20:55:33 CEST] <Mista_D> JEEB: it takes half a page of script ... thougth there was somethign faster, thanks anyways. [20:55:44 CEST] <JEEB> uhh [20:55:51 CEST] <JEEB> you do know that ffprobe can give you JSON? [20:55:52 CEST] <JEEB> right [20:56:17 CEST] <JEEB> and if whatever you're using takes half a page of script to parse json variables, then you're doomed :3 [20:56:25 CEST] <Mista_D> lol [21:53:12 CEST] <bencc> how can I feed RTP packets to ffmpeg other than udp? [21:53:30 CEST] <bencc> can I somehow push pcap or rtpdump file to ffmpeg? [22:05:23 CEST] <Mista_D> MFS error with QSV: http://pastebin.ca/3073652 [22:05:33 CEST] <Mista_D> MFX * [22:06:22 CEST] <Nolski> Anyone here have experience with MLT rendering? I'm thinking if it's possible it might be easier to find some sort of headless renderer and feed it some sort of markup of edits rather than writing my own [23:08:45 CEST] <Nolski> Anyone know of a good way to render a list of ffmpeg commands but only re-encode the video once? I'm worried about losing lots of quality by running ffmpeg a bunch of times on one video [23:09:43 CEST] <Nolski> s/lots/loss/ [23:10:00 CEST] <chungy> make sure to do -c:v copy then [23:11:17 CEST] <Nolski> chungy: quick explanation of what that means? [23:11:32 CEST] <chungy> It just copies the video stream instead of re-encoding it [23:12:22 CEST] <Nolski> what if you're rendering video on top of another one or trimming something or using -filter_complex? [23:17:14 CEST] <Nolski> chungy: ^ [23:18:02 CEST] <chungy> then you can't do that. Try to compact all the stuff into a single command if you can. [23:18:36 CEST] <chungy> Alternatively, use a lossless video codec in the interim. something like "-c:v ffvhuff" or "-c:v h264 -qp 0" [23:19:03 CEST] <Nolski> chungy: h264 is lossless? [23:20:36 CEST] <chungy> Can be [23:20:58 CEST] <chungy> -qp 0 makes it lossless [23:21:18 CEST] <Nolski> neat [23:21:53 CEST] <pzich> it'll also make a pretty huge file though [23:22:27 CEST] <Nolski> That shouldn't be too much of a problem. I'll be deleting the file once the edits are finished. [23:22:36 CEST] <chungy> Yeah, lossless is huge :P [23:22:46 CEST] <Nolski> Shouldn't tooling for this sort of stuff exist already? [23:23:04 CEST] <Nolski> Like a giant ffmpeg wrapper where you just send it an xml or json of edits to some video files and it does it for you? [23:23:34 CEST] <Nolski> I can't find anything that does that that is FOSS. [23:31:47 CEST] <chungy> good idea, I don't know any myself [23:32:08 CEST] <chungy> I've tended to make ad-hoc shell scripts every time I'm doing large batch operations (like converting TV series from DVD to *.mkv files...) [23:32:43 CEST] <Nolski> chungy: yeah same here. Now I'm working on a script that's output by an in browser video editor so this shell script is turning into a very large python application haha. [23:35:12 CEST] <chungy> haha [23:38:51 CEST] <chungy> I was kind of proud of my script for Star Trek TNG... it would fetch the episode title, season number, and episode number from a psql database I have, and tag it and name the file in the proper way to minimize the chance of errors. Of course I deleted it when I was done with it. heh [00:00:00 CEST] --- Sat Jul 25 2015

1 0

ffmpeg-devel.log.20150723
by burek 24 Jul '15

24 Jul '15

[00:51:47 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:9e83ac6114de: tests/checkasm/h264qpel: Use LOCAL_ALIGNED_16() [00:51:47 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:f467fc02b475: tests/checkasm/h264pred: Use LOCAL_ALIGNED_16() [02:17:08 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:874b3117ed62: mpegaudiodec: copy AVFloatDSPContext from first context to all contexts [02:17:09 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:fdf166c5d2bb: vc1dec: use get_bits_long and limit the read bits to 32 [02:17:10 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:a3d3e0a6bc39: nutdec: check maxpos in read_sm_data before returning success [02:17:11 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:21769e3884ea: wavpack: use get_bits_long to read up to 32 bits [02:17:12 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:dc85a7533263: huffyuvdec: validate image size [02:17:13 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:a0f50ddcb831: pthread_frame: forward error codes when flushing [02:17:14 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:32df1cd6ac09: ffmpeg: exit_on_error if decoding a packet failed [02:17:15 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:7f84858dcf49: ffmpeg: only count got_output/errors in decode_error_stat [02:17:16 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:e812220a304d: wavpack: limit extra_bits to 32 and use get_bits_long [02:17:17 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:3c96f21d6e9d: webp: fix infinite loop in webp_decode_frame [02:17:18 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:2d89356641f8: snow: remove an obsolete av_assert2 [02:17:19 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:44a9e2dbba3a: hevc: check slice address length [02:17:20 CEST] <cone-469> ffmpeg 03Andreas Cadhalpun 07release/2.6:088733414a65: imc: use correct position for flcoeffs2 calculation [02:17:21 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07release/2.6:b17cec526214: update changelog [03:49:17 CEST] <cone-469> ffmpeg 03Ivan Uskov 07master:1acb19d12bcd: libavcodec/qsvdec_h264.c: SPS parsing is now performed by MFXVideoDECODE_DecodeHeader() in libavcodec/qsvdec.c [04:13:03 CEST] <cone-469> ffmpeg 03James Almer 07master:a176bbc873b1: avutil/softfloat: move av_sincos_sf() back to header [06:13:08 CEST] <thrwaway> I uploaded a bunch of files to upload.ffmpeg.org per ticket #4644 (and http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/195493) but forgot to include the ticket ID in the filenames.. sorry for the confusion. [09:33:56 CEST] <durandal_170> will someone comment my patches? [11:43:54 CEST] <cone-479> ffmpeg 03Alexandra Hájková 07master:ee80f834cbb6: asfdec: set nb_streams to 0 in the asf_read_close [11:43:55 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:cd4c878934a6: Merge commit 'ee80f834cbb6dbacdc1efb4c658a7d775e82ebff' [11:51:14 CEST] <cone-479> ffmpeg 03Alexandra Hájková 07master:aed7715b8fa2: asfdec: increment nb_streams right after the stream allocation [11:51:16 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:40e8ade9ebda: Merge commit 'aed7715b8fa295980c221f1cd095d42cd3bd74a6' [11:54:28 CEST] <durandal_170> Daemon404: so for x32 checking in filter frame for reverse should be split in two? [12:01:51 CEST] <ubitux> michaelni: why the check on resolution in ffv1 for the enabling of multicore? [12:01:54 CEST] <ubitux> if (avctx->slices == 0 && avctx->level < 0 && avctx->width * avctx->height > 720*576) [12:04:43 CEST] <rcombs> baptiste: I'd assume it's not worth it below that? [12:04:45 CEST] <ubitux> it would be nice to have documentation on these levels [12:04:48 CEST] <rcombs> erm, ubitux: ^ [12:04:56 CEST] <rcombs> just my guess, though [12:05:00 CEST] <ubitux> bc -level 3 makes a real difference [12:05:18 CEST] <ubitux> rcombs: maybe but seems a bit arbitrary [12:05:33 CEST] <rcombs> yeah [12:05:38 CEST] <rcombs> should at least be better-documented [12:06:20 CEST] <ubitux> it seems there is a 270 vs 130 fps difference here whether i force -level 3 or not [12:06:24 CEST] <ubitux> for a 512x512 input [12:07:59 CEST] <michaelni> ubitux, higher level also means the decoder must be more recent to support it, but if people prefer we certainly could adjust the threshold [12:09:23 CEST] <ubitux> i think there should be no threshold on that IMO [12:09:38 CEST] <ubitux> i mean whether the resolution change it will be decodable or not with old versions? [12:11:30 CEST] <michaelni> if you have 16x16 pixel video you wont use slices nor benefit speedwise if you do, so why force later syntax ? [12:12:01 CEST] <ubitux> is the threading overhead going to make a difference in speed for a 16x16? [12:12:30 CEST] <ubitux> but anyway, if there is a threshold, why is it related to the level and not the decoding capability? [12:12:59 CEST] <ubitux> the threading* capability sorry [12:14:05 CEST] <michaelni> i think generally the decoder side is more important than the encoder, encoder side is generally one, decoder many and decoders also tend to be lower end [12:16:28 CEST] <michaelni> higher resolution videos should use slices so decoders can decode them in parallel [12:17:19 CEST] <michaelni> its important that realtime decoding isnt prevented due to single slice high resolution video [12:18:06 CEST] <cone-479> ffmpeg 03Anton Khirnov 07master:22ecfcd4c79c: af_channelmap: properly set the supported output channel layouts [12:18:07 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:9c31b396413a: Merge commit '22ecfcd4c79cdf812fdf406525ddf0fd1f7114e4' [12:19:00 CEST] <michaelni> OTOH for really low resolution slices will add significant overhead id expect and when none are used the older syntax should be fine [12:19:37 CEST] <michaelni> but if the threshold is poorly choosen, please change it [12:23:18 CEST] <michaelni> also with overhead i mean here bitrate overhead. For a 16x16 pixel video that should be significant to have more than 1 slice [12:34:25 CEST] <cone-479> ffmpeg 03Tom Butterworth 07master:ebe8b5d947c4: dds: Fix the slice size computation [12:34:26 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:157fa73992ab: Merge commit 'ebe8b5d947c41449c684f17c6826fe6bc46c0360' [12:41:59 CEST] <cone-479> ffmpeg 03Tom Butterworth 07master:ae5a8dca675e: hap: Fix slice size computation [12:42:00 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:b04b02100550: Merge commit 'ae5a8dca675ee544178225256893e679b750cb63' [12:57:57 CEST] <cone-479> ffmpeg 03Alexandra Hájková 07master:7f388c0fabc5: asfdec: remove the wrong condition [12:57:58 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:0a03271ef64d: Merge commit '7f388c0fabc51eca3106e7cc443393269435ab52' [13:10:10 CEST] <kierank> is there a way from the api to list all installed codecs [13:12:25 CEST] <kierank> fdk-aac is clearly installed [13:12:27 CEST] <kierank> but it doesn't work [13:24:44 CEST] <durandal_1707> kierank: how it doesnt work? [13:25:03 CEST] <kierank> I have some configure scripts that end up disabling it [13:25:06 CEST] <kierank> when it's clearly there [13:25:17 CEST] <kierank> but never mind [13:25:22 CEST] <kierank> I am just going to built everything [13:32:46 CEST] <wm4> yes, there's a way [13:33:11 CEST] <wm4> AFAIK you can list both codecs and decoders/encoders [13:33:20 CEST] <wm4> ffmpeg.c uses that too [14:41:28 CEST] <wm4> so, how is closed captions support in ffmpeg? [14:41:54 CEST] <wm4> they're not exposed like normal subtitles, are they? you still need to get the side data from the video decoder output [14:42:09 CEST] <nevcairiel> Yes, its side data [14:42:54 CEST] <wm4> and there's this insane lavfi filter too (what was its name?) [14:43:13 CEST] <nevcairiel> In fact it's part of the lavfi core functions [14:43:45 CEST] <nevcairiel> Which extracts the cc side data into a lavfi graph [14:44:07 CEST] <wm4> but I thought lavfi has no subtitles [14:44:24 CEST] <nevcairiel> It doesnt [14:44:28 CEST] <nevcairiel> It's a big hack [14:44:39 CEST] <JEEB> yeah, the decoded surfaces are "video" [14:44:52 CEST] <JEEB> which you can then overlay on top of actual video with -filter_complex [14:45:00 CEST] <wm4> lol [14:45:10 CEST] <nevcairiel> No I don't think you can even do that [14:45:15 CEST] <wm4> jesus christ on a stick etc. [14:45:21 CEST] <nevcairiel> You can write the raw cc data to a file [14:45:32 CEST] <JEEB> oh, CC [14:45:34 CEST] <JEEB> that's even worse [14:45:44 CEST] <JEEB> rip [14:45:57 CEST] <wm4> I can't find it [14:46:06 CEST] <wm4> maybe it was only a proposed patch? or I'm blind [14:46:14 CEST] <nevcairiel> Blind it is [14:46:32 CEST] <nevcairiel> It's a very obscure feature though [15:10:53 CEST] <ubitux> wm4: nevcairiel: git grep -C2 '\[sub\]' doc [15:12:30 CEST] <wm4> ok [15:12:35 CEST] <wm4> this doesn't tell me much though [15:13:23 CEST] <nevcairiel> in practice, the lavfi thing is not useful at all [15:13:29 CEST] <nevcairiel> unless you want the cc data in a file [15:13:31 CEST] <nevcairiel> for some reason [15:15:47 CEST] <wm4> oh, it was libavdevice/lavfi.c [15:16:14 CEST] <wm4> yeah, that's a terrible hack [15:16:28 CEST] <ubitux> lavfi device is pretty useful [15:16:49 CEST] <wm4> no, it's a terrible hack to work around ffmpeg.c being retarded [15:17:06 CEST] <wm4> and the CC hack in it is even more terrible and hackier [15:17:07 CEST] <ubitux> no, it's typically for users such as ffplay or ffprobe who don't want to support complex filtergraph themselves [15:17:15 CEST] <Daemon404> [14:16] <@ubitux> lavfi device is pretty useful <-- hitler did nothing wrong [15:17:19 CEST] <Daemon404> thats what i read it as [15:17:27 CEST] <wm4> (although in a way it's the right approach) [15:17:36 CEST] <wm4> (the CC hack in the lavfi device I mean) [15:18:11 CEST] <ubitux> i have no opinion on the cc hack [15:18:23 CEST] <ubitux> but lavfi device itself is useful [15:18:49 CEST] <wm4> why? [15:19:06 CEST] <ubitux> 15:17 <@ubitux> no, it's typically for users such as ffplay or ffprobe who don't want to support complex filtergraph themselves [15:19:12 CEST] <ubitux> ffplay -f lavfi testsrc [15:19:27 CEST] <ubitux> ffprobe -f lavfi "amovie=...; movie=... " [15:19:37 CEST] <wm4> sigh [15:19:38 CEST] <ubitux> (multiple sources when your app supports only one) [15:19:40 CEST] <Daemon404> tl;dr the ycant be arsed to use libavfilter's terrible api [15:19:43 CEST] <Daemon404> wm4 is right [15:19:45 CEST] <Daemon404> it's a lazy hack [15:19:46 CEST] <Daemon404> for a shitty api [15:20:09 CEST] <wm4> the fabled ffmpeg high level API would make this hack unnecessary [15:20:11 CEST] <ubitux> well, because multiple inputs are not a simple thing and most of the app are designed for only one input [15:25:25 CEST] <durandal_1707> Daemon404: fine to push reverse fix? [15:26:02 CEST] <Daemon404> no, it's not technically the correct fix [15:26:13 CEST] <Daemon404> sorry, i forgot to reply [15:26:19 CEST] <Daemon404> (i discussed it in here with nevcairiel) [15:26:25 CEST] <Daemon404> i can reply now or send a better fix [15:27:41 CEST] <nevcairiel> checking the other array would probably work, but the real correct fix is to separate them [15:27:56 CEST] <nevcairiel> (or stop using the _fast version) [15:28:11 CEST] <Daemon404> at this point _fast has no benefits [15:28:16 CEST] <Daemon404> may as well use av_realloc [15:34:46 CEST] <durandal_1707> Daemon404: I want to fix this asap, so what about comitting one that check both? [15:38:01 CEST] <Daemon404> that is ok too [15:42:02 CEST] <noncom> hello! [16:06:07 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:744051a57ad5: avcodec/hevc_parse: Print the name of the NAL units in addition to the numerical nal_unit_type in the debug output [16:51:13 CEST] <cone-479> ffmpeg 03Nicolas George 07master:52c75d486ed5: lavc/hevc: rudimentary support for skip_loop_filter. [16:55:57 CEST] <Daemon404> what an utterly asinine way to gain speed [16:57:18 CEST] <wm4> what did you expect [16:57:26 CEST] <wm4> h264 has all these funny speed hacks which you can enable [16:57:32 CEST] <wm4> so of course hevc is going to get them too [16:57:53 CEST] <Daemon404> disabling the loop filter surely wont make it look like shit [16:57:55 CEST] <Daemon404> surely. [16:58:08 CEST] <nevcairiel> people already asked me about skipping non-refs in hevc because their crappy machines cant play it [16:58:17 CEST] <nevcairiel> but i'll never expose those options to the user [16:58:23 CEST] <nevcairiel> never have for h264, never will for anything else [17:00:29 CEST] <cone-479> ffmpeg 03Shivraj Patil 07master:fd7eadd25c77: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for VP9 lpf functions [17:07:45 CEST] Action: BBB still waiting for ffvp9 neon [17:13:00 CEST] <cone-479> ffmpeg 03Ivan Uskov 07master:d50ab820dacf: libavcodec/qsvdec_h264.c: refactoring: functionality of qsv_process_data() has been moved into qsvdec.c [17:23:38 CEST] <durandal_1707> which avs filter is good denoiser? [17:37:04 CEST] <cone-479> ffmpeg 03Shivraj Patil 07master:c03800d5921e: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for VP9 idct functions [17:38:17 CEST] <J_Darnley> durandal_1707: the better ones I remember used scripts centered around MVTools and FFT3D [17:38:30 CEST] <J_Darnley> was it called "qtgmc"? [17:38:46 CEST] <Daemon404> that's for deinterlacing/bobbing [17:38:53 CEST] <Daemon404> not denoising [17:38:56 CEST] <J_Darnley> and to answer the question you asked a few days ago: I haven't been working on the asm. [17:39:14 CEST] <J_Darnley> Daemon404: ah, my bad. [17:43:17 CEST] <J_Darnley> Perhaps it was Temporal Degrain I was thinking of [17:43:21 CEST] <J_Darnley> http://avisynth.nl/index.php/Temporal_Degrain [17:45:26 CEST] <J_Darnley> Or maybe MC Temporal Degrain [17:45:33 CEST] <J_Darnley> http://avisynth.nl/index.php/MCTemporalDenoise [17:45:44 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:8eff61fd45f1: avcodec/mips/vp9_idct_msa: Replace __volatile__ by volatile [18:00:24 CEST] <cone-479> ffmpeg 03Tom Butterworth 07master:083cbc930d07: snappy: Refactor so ff_snappy_uncompress() uses an existing buffer [18:00:25 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:efa1e26122e3: Merge commit '083cbc930d077651ea7e3fbc32ec45352cfed7e7' [18:40:31 CEST] <rcombs> J_Darnley: I'm MC Temporal Degrain and I'm here to say, 'spu adgw? [18:41:23 CEST] <JEEB> lol [18:44:25 CEST] <cone-479> ffmpeg 03Tom Butterworth 07master:11f3d5c69b71: hap: Name enums, remove unused struct member [18:44:26 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:cb59d29fd6b9: Merge commit '11f3d5c69b711a1f1631961921ecd20d31f8336d' [19:21:33 CEST] <Compnn> cmon so close to 400 [19:22:22 CEST] <Compnn> nevcairiel : forcing users to re-encode all files ? :P [19:23:51 CEST] <Compn> wm4 : closed captions are exposed as subtitles from lavf, iirc. you are talking about displaying them ? thats another question. [19:24:39 CEST] <Compn> depending on the format. [19:24:46 CEST] <wm4> no they aren't [19:25:55 CEST] <AstralStorm> I see that ffmpeg does not support edit list properly if at all, right? [19:26:06 CEST] <AstralStorm> for this reason I get invalid duration last packet [19:26:16 CEST] <philipl> wm4: Doesn't the latest Kodi manage to show CCs correctly? Maybe can borrow some code from there [19:26:19 CEST] <AstralStorm> I can wing the initial delay by skipping negative pts packets [19:26:47 CEST] <AstralStorm> but the trailer is seemingly impossible to do in a general (codec independent, container independent) case [19:27:00 CEST] <wm4> philipl: no idea [19:27:08 CEST] <AstralStorm> I found some patches from 2014 [19:28:09 CEST] <AstralStorm> https://ffmpeg.org/pipermail/ffmpeg-devel/2014-December/167157.html - ye olde thread [19:28:18 CEST] <AstralStorm> any idea what should I do? [19:29:06 CEST] <AstralStorm> I can potentially "hack it" by adding initial pts to the final packet [19:29:13 CEST] <AstralStorm> but that seems wrong [19:29:32 CEST] <AstralStorm> *to the duration of the final packet - trimming it [19:30:53 CEST] <AstralStorm> that happens to be true for solo audio but might not be correct; any help? [19:38:44 CEST] <J_Darnley> rcombs: top kek!\ [20:20:29 CEST] <durandal_1707> ubitux: look at allyuv and tell me if there is better way to sort colors [20:21:05 CEST] <ubitux> why u no allrgb? [20:32:01 CEST] <durandal_1707> ubitux: feel free to do it [20:32:27 CEST] <ubitux> sure but i mean, why go for the hard way? [20:32:50 CEST] <ubitux> btw, i see GBRP, is this on purpose? [20:33:10 CEST] <durandal_1707> if you don't need pretty look it is easy [20:33:33 CEST] <durandal_1707> it just for fun, I will remove it [20:34:15 CEST] <ubitux> i would personally just increment from 0 to 255 every plane instead of trying to find a smart kind of procedural formula [20:34:28 CEST] <durandal_1707> -f lavfi -i allyuv,format=gbrp [20:34:43 CEST] <ubitux> btw, CONFIG_CELLAUTO_FILTER [20:35:00 CEST] <durandal_1707> aww will fix [20:35:54 CEST] <durandal_1707> if you need boring mode it could be added easy just add new option [20:41:45 CEST] <ubitux> this allyuv is actually pretty [20:42:09 CEST] <ubitux> so you sure there are all the colors? [20:43:17 CEST] <durandal_1707> I shortly checked with sort [20:43:59 CEST] <durandal_1707> it somehow resembles windows logo [20:45:12 CEST] <ubitux> durandal_1707: btw, http://joco.name/2014/03/02/all-rgb-colors-in-one-image/ [20:47:06 CEST] <durandal_1707> I read all that and was not happy with output [20:47:31 CEST] <cone-479> ffmpeg 03Paul B Mahol 07master:625bf6a55c3d: avfilter/vf_reverse: check also pts_size when reallocating [20:47:32 CEST] <cone-479> ffmpeg 03Paul B Mahol 07master:e59315c4abce: doc/filters.texi: fix two typos in reverse filter description [20:58:58 CEST] <cone-479> ffmpeg 03Tom Butterworth 07master:26e8247c1c0b: avcodec/hap: (trivial) clarify comment [21:12:22 CEST] <cone-479> ffmpeg 03Tom Butterworth 07master:64539e12132c: avcodec/hap: (trivial) rename enum values and document their meaning [21:17:27 CEST] <durandal_1707> ubitux: what you mean by #2558 [21:46:41 CEST] <cone-479> ffmpeg 03Martin Storsjö 07master:44f7df0c9879: dds: Write the palette in the native endian form [21:46:42 CEST] <cone-479> ffmpeg 03Michael Niedermayer 07master:c6965f62a2e3: Merge commit '44f7df0c987965763c609f6dc36974b04182e58d' [21:50:00 CEST] <kierank> durandal_1707: I'm not sure why but smptebars doesn't come out nicely on the scope [21:50:03 CEST] <kierank> neither does hd [21:50:13 CEST] <kierank> I'm going to investigate next week [21:56:47 CEST] <durandal_1707> kierank: what is wrong? [21:57:43 CEST] <kierank> I'll have to take a photo [21:57:51 CEST] <kierank> it's weird because we use the numbers in the spec too [22:01:40 CEST] <durandal_1707> do you read colorspace set in frame? [22:02:41 CEST] <kierank> no, everything stays yuv [22:09:59 CEST] <ubitux> durandal_1707: maybe i was on drug that day [22:10:39 CEST] <durandal_1707> what is better name duck or sidechaincompress ? [23:06:53 CEST] <jamrial> durandal_1707: the latter, probably [23:06:54 CEST] <jamrial> why duck? [00:00:00 CEST] --- Fri Jul 24 2015

1 0

ffmpeg.log.20150723
by burek 24 Jul '15

24 Jul '15

[00:19:03 CEST] <DHE> I'm looking for something that is suitable for a daemonized version of ffmpeg. In particular if something goes wrong, it should restart. (This if for a streaming task). There's an option -max_error_rate which looks like a step in the right direction but doesn't appear to actually do anything beyond exit code [00:19:25 CEST] <DHE> I can script something up, it's a matter of error detection really [01:45:22 CEST] <Seablade_pixel> Quick question, it appears ffmpeg typically creates some sort of average for the console output (IE. fps seems to be an average of the last X seconds) am I correct in this first? If not how is it calculated? If so, is there a way I could get it to show me more instantaneous results? [01:46:27 CEST] <Seablade_pixel> Primarily I am trying to come up with a good way to monitor network issues when using ffmpeg to generate an output stream, and it doesn't seem like it responds quite as fast as I would like and more realtime output could help with this if what I am seeing is correct [02:15:45 CEST] <c_14> If you're streaming at constant bitrate, you could just check the network output, otherwise you could try the -progress option [02:27:44 CEST] <Seablade_pixel> c_14: Thanks, I will look into -progress. Not a constant bitrate in this case. [02:28:50 CEST] <Seablade_pixel> c_14: Pretty much I am capturing from a blackmagic design card and streaming out, I have my suspicions that around 8PM my time when everyone around me gets onto the network, I am seeing drops in network performance, which is causing ffmpeg to back up on encoding frames and fill the buffer of bmdcapture causing it to fail, but that is all conjecture at this point [04:05:08 CEST] <ig0r_> is there a simple way to convert an MKV-file to an MP4-file without re-encoding it? [04:36:58 CEST] <Seablade_pixel> ig0r_: ffmpeg -i INPUT.mkv -c copy OUTPUT.mp4 [04:37:38 CEST] <Seablade_pixel> Might have to do `-c:v copy -c:a copy` instead of just -c copy, can't remember off hand. And if you only want certain streams you will need to use -map to say what streams you want [05:11:41 CEST] <oscargot> Hello, I' [05:11:51 CEST] <oscargot> oops.. [05:13:51 CEST] <oscargot> Hello, is there anyway to directly encode a NV12 pix format image to MPEG4 using libav rather than having to first convert it to YUV420P sws_scale? [06:17:26 CEST] <karkhaz> Hi, I'm trying to trace a segfault in libffmpeg. It happens in a function called ff_pack_2ch_float_to_int16_a_sse2 [06:17:40 CEST] <karkhaz> Where is this function defined? I can't find it anywhere [06:18:12 CEST] <oscargot> Hello, is there anyway to directly encode a NV12 pix format image to MPEG4 using libav rather than having to first convert it to YUV420P sws_scale? Or optimise the conversion step? [06:18:19 CEST] <karkhaz> That function gets called from swri_audio_convert in libswresample/audioconvert.c [06:19:31 CEST] <karkhaz> gdb won't tell me where it is. It get assigned in libswresample/x86/audio_convert_init.c, but otherwise I can't tell where the function is actually implemented. [06:21:28 CEST] <karkhaz> from swri_audio_convert there is a call to ctx->simd_f(...), so simd_f is a pointer to the function ff_pack_2ch_float_to_int16_a_sse2. But I can't see where that function is implemented [06:38:21 CEST] <EO_> is -strftime unsupported? [06:38:41 CEST] <EO_> [mp4 @ 0x26b5be0] Invalid segment filename template 'seg-%Y-%m-%d_%H-%M-%S.mp4' [06:39:08 CEST] <EO_> ^^^ I specified "-strftime 1" in the output params. It works when I just use %d there instead of actual strftime format string entities. [06:40:06 CEST] <EO_> https://www.ffmpeg.org/ffmpeg-all.html#segment_002c-stream_005fsegment_002c… <-- Manual claims support for -strftime [06:47:53 CEST] <EO_> it seems from the error message like use_strftime was not set [06:55:04 CEST] <EO_> heh, works with latest git build. I guess the ffmpeg static build is out of date. [07:27:40 CEST] <Stifler> Hi! Is there a way to make/force ffmpeg attempt to take thumbnails on a known 'corrupt' piece of video? [08:05:54 CEST] <durandal_1707> karkhaz: in x86 subdirectory [08:14:23 CEST] <oscargot> Hello, is there anyway to directly encode a NV12 pix format image to MPEG4 using libav rather than having to first convert it to YUV420P sws_scale? Or optimise the conversion step? [08:20:28 CEST] <durandal_1707> oscargot: conversion is slow for you? [09:53:47 CEST] <Lushi> hi [09:54:56 CEST] <Lushi> I am looking for some help using ffmpeg to create a multi-variant HLS stream. [09:55:20 CEST] <Lushi> I am having trouble getting the segments to line up. [09:55:35 CEST] <Lushi> Is there anyone here that has experience with this? [12:09:39 CEST] <DHE> I'd have answered Lushi but he's gone [13:41:36 CEST] <Lushi> I am looking for some help encodign an HLS file with different bitrates and framerates. [13:41:52 CEST] <Lushi> Does anyone here have experience doing this? [13:42:20 CEST] <Lushi> Also, when is generally a good time to ask questions on this IRC channel? [13:43:28 CEST] <Lushi> I posted a detailed description of the problem to ffmpeg-user list: http://ffmpeg-users.933282.n4.nabble.com/HLS-segment-duration-for-multiple-… [13:43:49 CEST] <Lushi> Does anyone have any ideas or pointers? [14:08:03 CEST] <DHE> Lushi: I'm doing that right now actually [14:09:04 CEST] <DHE> not with multiple framerates, but if you explicitly set your framerate it should be okay. keep in mind that a lot of TV runs at 29.97 fps (or more specifically, 30/1.001 fps) [14:18:57 CEST] <ILEoo> apples mediastreamvalidator errors out on different segment lengths? [14:24:55 CEST] <JEEB> I don't see https://developer.apple.com/library/ios/technotes/tn2235/_index.html showing such warning/error at least [14:30:31 CEST] <Lushi> DHE, thanks for writing back. [14:31:20 CEST] <Lushi> my understanding of HLS with multiple variants is that the segments must have teh same duration [14:31:41 CEST] <DHE> yes, and each should represent an identical segment in time [14:31:49 CEST] <Lushi> correct. [14:32:38 CEST] <Lushi> in addition to that, they should also all have the same EXT-TARGETDURATION [14:33:16 CEST] <Lushi> so, as you can se in my email, when I am running the ffmpeg segmentor on teh same input stream, I often end up with differnt length segments, and even different EXT-TARGETDURATION. [14:33:48 CEST] <Lushi> can the player jump around between streams when they have different length segments? my assumption is no.. [14:35:08 CEST] <JEEB> currently there are things that do not strictly set IDR frames at exact specific points and it seems to work [14:35:17 CEST] <JEEB> I guess http://tools.ietf.org/html/draft-pantos-http-live-streaming-16 is the authority on these things [14:38:12 CEST] <DHE> ffmpeg seems to do the right thing for a single segment, but in the end each .m3u8 is its own private thing and there's no coordination between them. I'm using "-x264opts no-scenecut:keyint=60" to force regular keyframes. if all videos have constant framerates then it all works out. [14:42:08 CEST] <JEEB> all of my TARGETDURATIONs seem to be matching, but naturally the extinf duration can be variable. the TARGETDURATION is just the maximum length of a segment [14:42:19 CEST] <JEEB> which seems to match the spec [14:43:55 CEST] <ILEoo> Lushi: player should check segment for timestamp in the beginning to sync playback positions when switching playlists [15:43:36 CEST] <noncom> hi! [15:48:36 CEST] <Lushi> sorry - was pulled a way for a bit. [15:49:15 CEST] <Lushi> one problem is when you have different TARGETDURATIONs - then the apple tester complains [15:52:14 CEST] <Lushi> another problem is that ffmpeg doesn't even listen to teh recommended lengths - in all my examples I asked for 9 seconds segments - and I got some segments that were up to 12 seconds [15:53:25 CEST] <Lushi> it seems that -force_key_frames fave more accurate splits than -g [15:55:00 CEST] <JEEB> -g just sets the maximum length of a GOP [15:56:33 CEST] <JEEB> I use VLC for the HLS muxing, and it seems to set TARGETDURATION to what I set on the command line. And as long as I have max GOP length in a similar figure (a bit lower just in case), the thing seems to work fine [15:57:39 CEST] <Lushi> when you say muxing, what do you mean? segmenting the file into chunks? or creating a master playlist for multiple streams? [15:58:21 CEST] <JEEB> creating the segments [15:58:59 CEST] <JEEB> but to be honest I'd be surprised if ffmpeg's muxer doesn't follow what you set as long as the GOPs are within that range [15:59:00 CEST] <Lushi> as I showed in my email, I was using ffmpeg to segment the files and create teh single stream m3u8 file - and was getting differnt segment lengths and even different target durations for different bitrates/framerate/resolutions... [15:59:03 CEST] <JEEB> so that could be looked into it [15:59:15 CEST] <JEEB> segment lengths being different is OK [15:59:24 CEST] <JEEB> TARGETDURATION not being what you set is getting herp derp [15:59:53 CEST] <JEEB> unless you have VFR and cannot signal that to libx264 so it keeps using the original GOP lengths picture-wise [15:59:55 CEST] <Lushi> why is it ok for segment lengths to be different? how will the player switch between streams when teh segment lengths differ? [16:00:14 CEST] <JEEB> by parsing the lengths of the segments and then seeking within the segment? [16:00:18 CEST] <JEEB> as far as I can see that works fine [16:00:52 CEST] <JEEB> TARGETDURATION being different might cause complaints, that I don't disagree about. I just can't test the app because I don't have my apple credentials around [16:00:56 CEST] <Lushi> ex: if one stream segment goes from second 0-9 and the other stream goes from second 0-10, then the client can't switch tto teh second stream at the end of the first... [16:01:03 CEST] <Lushi> or do I not understand that correctly? [16:01:21 CEST] <JEEB> why can't it? it should have the playlist loaded for both, no? [16:01:39 CEST] <JEEB> the playlists contain the lengths of the segments [16:02:04 CEST] <Lushi> at the end of the first segment (second 9), there is no start of the second stream segment until second 10. [16:02:35 CEST] <Lushi> so it will load segment 2 of the first stream for 1 second and then switch to the second stream? [16:03:04 CEST] <JEEB> if the player is switching from playlist A to playlists B on the meta playlists, it would check its current timestamp in playback, and load/seek the segment accordingly [16:03:18 CEST] <JEEB> that's exactly why the lengths are there for the segments [16:03:22 CEST] <Mavrik> Not all of them do that reliably though. [16:03:33 CEST] <Mavrik> IIRC Apple can do that well, Android is buggy and STBs mostly crap out [16:04:12 CEST] <JEEB> well as far as I could see the samsung TV I had access to did it well as well. But I do agree that different implementations and their versions can have bugs [16:04:36 CEST] <Mavrik> At least we had huge issues and had to make sure all segments are of same length and have key frames at same interval [16:05:39 CEST] <JEEB> well the experiences around me seem to be of more positive results, thankfully :) [16:05:42 CEST] <Lushi> tI found where in teh apple sdocs it said that segments should be equal length [16:05:43 CEST] <Lushi> https://developer.apple.com/library/ios/documentation/NetworkingInternet/Co… [16:06:06 CEST] <Lushi> "If you already have a media file encoded using supported codecs, you can use a file segmenter to encapsulate it in an MPEG-2 transport stream and break it into segments of equal length." [16:06:52 CEST] <Lushi> this is why I was freaking out when ffmpeg segmentor was making different duration files...and then even more when it was making different TARGETDURATIONs [16:07:07 CEST] <JEEB> and then it proceeds to to say that "For maximum accuracy, you should specify all durations as floating-point values when sending playlists to clients that support version 3 of the protocol or later. (Older clients support only integer values.) You must specify a protocol version when using floating-point lengths; if the version is omitted, the playlist must conform to version 1 of the protocol." [16:07:13 CEST] <JEEB> also the thing is a standard now [16:07:17 CEST] <JEEB> I recommend you just keep to the standard [16:07:37 CEST] <JEEB> https://tools.ietf.org/html/draft-pantos-http-live-streaming-16 [16:07:40 CEST] <Lushi> you can have floating point values and have each one be the same length... [16:08:03 CEST] <Lushi> Mavrik, we are discussing http://ffmpeg-users.933282.n4.nabble.com/HLS-segment-duration-for-multiple-… [16:08:27 CEST] <Lushi> if you have any input, I would be thrilled! - been working on this since last week :) [16:08:32 CEST] <JEEB> take a look at the example in chapter 2 of the spec [16:08:47 CEST] <Mavrik> Hmm, didn't have that use-case, since TV broadcasts are (and must be) CFR [16:09:05 CEST] <Mavrik> Too much hardware out there that chokes if it isn't :) [16:09:15 CEST] <DHE> I haven't tried this yet, but for content known to be 29.97fps I'd use something like "ffmpeg -i input [...] -r 30/1.001 -x264opts keyint=60 output-high.m3u8 [...] -r 15/1.001 -x264opts keyint=30 output-low.m3u8" [16:09:17 CEST] <DHE> have you tried that? [16:10:56 CEST] <Lushi> are you creating multiple segments with the same command line? [16:11:48 CEST] <Lushi> if so, then that might allow ffmpeg to split them all together at the same places...which is what I want... [16:12:15 CEST] <JEEB> it's separate encoders [16:12:16 CEST] <JEEB> so no [16:14:04 CEST] <Lushi> Mavrik: CFR - do you mean CRF (Constant Rate Factor) ? [16:14:40 CEST] <JEEB> no [16:14:43 CEST] <JEEB> constant frame rate [16:14:57 CEST] <Lushi> ah [16:15:26 CEST] <Lushi> JEEB, in the example in chapter 2 of the spec, the first 2 segments have the same duration - only the last one is shorter (which is understandable -since that is the "leftover content" att the end) [16:15:50 CEST] <JEEB> it /could/ be [16:15:57 CEST] <JEEB> anyways, I tried to search for specific limitations [16:15:59 CEST] <JEEB> those were not found [16:16:10 CEST] <JEEB> at least looking for the instances of "duration" or "segment duration" [16:16:28 CEST] <raven737> Hi, i would like to get motion vectors of mpeg streams as quickly as possible (least recourse utilization). Would it be possible to only parse mpeg streams without decoding images to get that data? I don't want any image output , just mv info. Is that possible as it is now or with simple code modification? [16:16:43 CEST] <JEEB> also the fact that iOS devices seem to handle that just fine seems to note that it is not against the standard to do that. [16:18:11 CEST] <Lushi> ok - so you think it is not a problem to have different lengths between different resolutions...but the TARGETDURATION would still be a problem...and without strictly controlling the segemtn durations, you can't strictly control the TARGETDURATION/// [16:18:24 CEST] <Lushi> (as is seen in my first example)... [16:18:54 CEST] <raven737> I saw: https://ffmpeg.org/pipermail/ffmpeg-devel/2014-August/161593.html ... Can I somehow get MV data into CSV files? [16:19:23 CEST] <JEEB> well I just set the parameter that controls TARGETDURATION and my GOPs maximum lengths are just under that value [16:19:31 CEST] <JEEB> unless I suddenly get VFR input it should be OK [16:19:42 CEST] <Lushi> which parameter is that? [16:20:03 CEST] <JEEB> with VLC it's seglen [16:20:12 CEST] <JEEB> I haven't checked the ffmpeg HLS muxer yet [16:20:14 CEST] <Lushi> I got duration 10 when I request "-hls_time 9" [16:20:36 CEST] <Lushi> maybe I'll just try the VLC muxer if you have good luck with that.. is that CLI as well? [16:21:35 CEST] <Lushi> and do you use vlc as the transcoder as well (can it even do that?)? or just to segment the resulting files? [16:21:56 CEST] <DHE> Lushi: I'm using one process to do them all [16:22:03 CEST] <DHE> but same framerates fore ach [16:22:04 CEST] <DHE> each [16:23:16 CEST] <JEEB> Lushi: it can re-encode as well but right now I'm not doing that [16:23:21 CEST] <JEEB> I've done that for other stuff though [16:23:27 CEST] <JEEB> and yes, it's cli [16:23:31 CEST] <JEEB> see #videolan for help [16:30:16 CEST] <Lushi> maybe I'll try that...it just seems strange that this isn't a " [16:30:29 CEST] <Lushi> isn't a "solved problem" for ffmpeg... [16:31:06 CEST] <JEEB> it should be as long as your GOPs are within that maximum length [16:31:07 CEST] <Lushi> I can't be the only person to try to transcode files into multiple bitrate streams for apple devices..that is THE simplest use case of HLS... [16:31:33 CEST] <BtbN> that's the most complicated usecase. [16:31:43 CEST] <Lushi> :) [16:31:59 CEST] <BtbN> And i don't think ffmpeg supports more than plain single-stream hls output. [16:32:16 CEST] <JEEB> yeah, you have to make the meta playlists yourself, and the encodes are going to be separate [16:32:28 CEST] <JEEB> it's just weird if it cannot set the TARGETDURATION to be what the user sets [16:32:33 CEST] <Lushi> so maybe the question is wrong... which of my commands in my email were not correct to set the GOPs to the correct maximum length. [16:33:00 CEST] <JEEB> -g should be all that's needed for maximum GOP length control [16:33:46 CEST] <JEEB> anyways, looking at the HLS muxer's code [16:33:49 CEST] <JEEB> avio_printf(out, "#EXT-X-TARGETDURATION:%d\n", target_duration); [16:34:30 CEST] <Lushi> well - if you look in my example, I set -g 90 and -hls_time 9 and I still got durations of 11.xxx seconds... [16:34:39 CEST] <JEEB> seems like you might not be able to specifically control that target_duration variable [16:34:52 CEST] <JEEB> because just before that printf line I see it going through the segments [16:35:01 CEST] <JEEB> target_duration = ceil(en->duration); [16:35:01 CEST] <Lushi> that value is just the maximum length of each of the segments.. [16:35:58 CEST] <JEEB> and yes it's weird if it doesn't let you control the maximum duration if that hls_time parameter is for that [16:36:06 CEST] <JEEB> and this is with the latest HEAD ffmpeg or so? [16:36:37 CEST] <Lushi> that is correct acording to the spec... the problem is that it is allowing some segments to be longer than I requested...which is bumping up the TARGETDURATION (but only on certain resolutions/bitrates/framerates)...which means that I can't combine them together... [16:37:35 CEST] <Lushi> a version downloaded from http://johnvansickle.com/ffmpeg/ a few days ago: 2.7.1-static [16:38:04 CEST] <JEEB> anyways, I recommend trying out VLC's muxing with say an mp4 sample you encoded with ffmpeg [16:39:06 CEST] <Lushi> the TARGETDURATION wouldn't be a problem if it consistantly set keyframes every 3 seconds and chopped the segemnt after 9 seconds exactly... [16:39:09 CEST] <JEEB> that really sounds quite weird, but it might be that the ffmpeg HLS muxer is specifically limited in some ways that make it just simpler to disable scenecut in libx264 [16:39:46 CEST] <Lushi> but apparently that isn't happening (at least with the arguments I am giving it) ..since each segment is a different duration... [16:40:13 CEST] <Lushi> what does diabling scenecut do? make it not look for the "best" place to put a segment/keyframe? [16:40:27 CEST] <JEEB> it wouldn't set keyframes at every 3 seconds, and the segment length would be variable. that is OK. what is not OK is that even though you are setting a maximum length to a segment it is not following that [16:40:41 CEST] <JEEB> which is why I note that you should definitely try using VLC's muxer [16:40:48 CEST] <JEEB> since it seems to be handling this fine as far as I can see [16:42:05 CEST] <Lushi> why wouldn't setting -force_key_frames "expr:gte(t,n_forced*3)" or -g 90 along with disablign scenecut not force a keyframe every 3 seconds? isn't that what they are made for? [16:42:19 CEST] <Lushi> maybe if I use them both together? [16:42:26 CEST] <JEEB> no, the latter should work just fine [16:44:40 CEST] <Lushi> I'm going to try al 3 together - see if I can brut force this... what is teh command to disable scene cut? [16:44:42 CEST] <JEEB> and probably the first two as well, but that's the worst case equipment [16:45:00 CEST] <JEEB> nah, don't [16:45:06 CEST] <JEEB> it's either the previous two or the latter two [16:45:34 CEST] <JEEB> -x264opts I think? and then you check what is the API option that is the same as --no-scenecut in x264cdli [16:45:37 CEST] <JEEB> *cli [16:48:10 CEST] <Lushi> found something interesting I read earlier and tossed aside. [16:48:14 CEST] <Lushi> https://sonnati.wordpress.com/2011/08/19/ffmpeg-%E2%80%93-the-swiss-army-kn… [16:48:23 CEST] <Lushi> Sometimes you may need to have a consistent, contant gop size across multiple bitrates (i.e. for Http Dynamic Streaming or HLS). To do that set min and max gop size equal and disable completely scene change (i.e. -g 100 -keyint_min 100 -sc-threashold 0). [16:49:59 CEST] <Lushi> I'll report back in a few minutes with results.. [16:50:35 CEST] <JEEB> well I'll just note one more time that the segments' lengths don't have to match, and that the main bugbear seems to be that you're not getting your set maximum segment length saved [16:50:51 CEST] <JEEB> making all GOPs the same length is a workaround around that [17:08:54 CEST] <Lushi> results: are dissapointing [17:08:55 CEST] <Lushi> http://pastebin.com/JqxiMe3U [17:09:53 CEST] <JEEB> yeah... if I cared enough I'd be poking the HLS muxer's code at this point [17:10:03 CEST] <JEEB> thankfully VLC's seems to work for me [17:10:25 CEST] <JEEB> that way I can control TARGETDURATION and that is all I need personally [17:10:45 CEST] <Lushi> should this be something I bring up to the devel list? [17:11:57 CEST] <JEEB> more like the trac issue tracker I guess. if you want to poke the original author he's in the libav project [17:12:07 CEST] <JEEB> lu_zero methinks [17:12:19 CEST] <JEEB> since hlsenc.c has * Copyright (c) 2012, Luca Barbato [17:13:01 CEST] <Lushi> I'll try running that using the segment segmentor (instead of hls) and see what what I get...as well... [17:13:46 CEST] <Lushi> what is the URL for the trac issue tracker? and do I put it in libav or in ffmpeg's tracker? [17:14:22 CEST] <JEEB> libav stuff generally gets merged on an almost daily basis so if you want to have it fixed for everyone then you go for libav and poke the original author [17:25:13 CEST] <pinkj> are there any known issues/expectations regarding multiple Streams from a single Feed in ffserver? I noticed that ffmpeg seems to report higher framerates on the command line when ffserver has only one Stream enabled and lower framerates when I have three Streams enabled. [17:29:14 CEST] <courrier> Hey guys! I'm tweaking the bitrate and fps during a 2pass encoding on the same input, do I need to reexecute the pass 1 each time? [17:30:39 CEST] <DHE> courrier: playing with the bitrate is usually okay. other settings like framerate will ruin the first pass logs [17:31:55 CEST] <courrier> DHE: ok thanks :) [17:38:05 CEST] <Lushi> JEEB, do you think I can post to the avconv tracker using my current ffmpeg arguments? or do I need to redo all my commands and experiments using avconv? [17:38:27 CEST] <Lushi> or should I just post it to ffmpeg and see what they do with it? [17:38:39 CEST] <Lushi> by the way, thanks for all the help! [17:40:36 CEST] <thiagoss> AVFrame->channel_layout seems to be 0 for stereo audio for a wmav2 file I have here. Wouldn't it be better if it contained the default stereo layout? Gstreamer currently uses av_get_channel_layout_nb_channels so it thinks the sample has 0 channels [17:41:16 CEST] <thiagoss> I've been trying to find where that would be set but still couldn't understand where that would be set [17:43:19 CEST] <thiagoss> It seems to be wma specific as 2-channels AAC seem to work nicely [17:56:48 CEST] <thiagoss> The issue seems to have been introduced by commit 9dd0b7ad821ae1b60acd9ac8f6384c03bd28be51 [18:10:50 CEST] <thiagoss> Either that or 464f94b206b041fa383ab4257226cb3f18dfb550 [18:11:34 CEST] <molavy> hi [18:11:58 CEST] <molavy> one convert make me crazy all day [18:12:50 CEST] <molavy> i try make audio output for newrock om4-4s for greeting [18:12:57 CEST] <molavy> on phone calls [18:13:04 CEST] <molavy> but i can't make it work [18:13:32 CEST] <molavy> on manuall there is codec table Table 2-23 Codec Methods Supported by OM [18:13:32 CEST] <molavy> Bit Rate (Kbit/s) Time Intervals of RTP Package [18:13:32 CEST] <molavy> Sending (ms) [18:13:32 CEST] <molavy> G729A 8 10/20/30/40 [18:13:33 CEST] <molavy> PCMU/PCMA 64 10/20/30/40 [18:14:13 CEST] <molavy> and on upload page is on line guide [18:14:14 CEST] <molavy> The size of the IVR file is limited to 300K bytes for PCMU format and 37K bytes for G.729 format [18:15:02 CEST] <molavy> i check codec default files on devices by download and open them using vlc [18:15:08 CEST] <molavy> codec info show [18:15:25 CEST] <molavy> Codec: PCM S16 LE (s16l) [18:15:33 CEST] <molavy> Sample rate: 8000 Hz [18:15:39 CEST] <molavy> mono [18:15:55 CEST] <molavy> Bits per sample: 16 [18:16:22 CEST] <molavy> i don't know how can i convert our recorded video that let me use it on this device [18:16:29 CEST] <molavy> can some one help me with this [18:16:33 CEST] <molavy> ? [18:20:47 CEST] <molavy> any idea? [18:22:50 CEST] <molavy> there is no idea? [18:22:59 CEST] <molavy> i really need help about this [18:25:10 CEST] <AstralStorm> hey, I'm having trouble with decoding ffmpeg-encoded MPEG-4 AAC to the right length [18:25:31 CEST] <AstralStorm> I'm now correctly skipping negative PTS data, but I get an oversized duration last packet [18:25:48 CEST] <AstralStorm> packet has duration e.g. 1536 while nb_frame is 1024 [18:26:42 CEST] <AstralStorm> *nb_samples [18:32:58 CEST] <AstralStorm> I cannot just take the modulo as this could potentially be valid in some formats [18:33:10 CEST] <AstralStorm> (non-mp4, the decoder is generic) [18:34:53 CEST] <Mavrik> I don't understand your issue [18:35:11 CEST] <Mavrik> If you need 1024 long frames, you'll have to split the 1536 to a 1024 + 512 one [18:37:24 CEST] <molavy> Mavrik: what about my problem [18:37:34 CEST] <molavy> can you help me [18:37:35 CEST] <molavy> ? [18:37:37 CEST] <Mavrik> What about it? [18:37:57 CEST] <Mavrik> I don't really understand what do you want. [18:38:00 CEST] <molavy> one convert make me crazy all day [18:38:08 CEST] <molavy> i try make audio output for newrock om4-4s for greeting on phone calls [18:38:31 CEST] <molavy> but i can't upload because it don't accept file [18:38:44 CEST] <molavy> on manual there is codec table [18:38:46 CEST] <molavy> Table 2-23 Codec Methods Supported by OM [18:39:02 CEST] <molavy> Methods Supported by OM Bit Rate (Kbit/s) Time Intervals of RTP Package [18:39:20 CEST] <molavy> G729A 8 10/20/30/40 [18:39:20 CEST] <molavy> PCMU/PCMA 64 10/20/30/40 [18:39:27 CEST] <molavy> and on upload page is one line guide [18:39:33 CEST] <molavy> The size of the IVR file is limited to 300K bytes for PCMU format and 37K bytes for G.729 format [18:39:40 CEST] <molavy> i check codec default files on devices by download and open them using vlc [18:40:06 CEST] <Mavrik> What you pasted is non-sensical and doesn't really help determining what do you need. [18:40:13 CEST] <molavy> codec info show : Codec: PCM S16 LE (s16l) Sample rate: 8000 Hz mono Bits per sample: 16 [18:40:49 CEST] <molavy> Mavrik: what you need to know [18:42:10 CEST] <molavy> ? [18:42:20 CEST] <Mavrik> container, codec, parameters [18:42:53 CEST] <AstralStorm> Mavrik: that's how I encoded it [18:43:06 CEST] <AstralStorm> the problem is in decoding - last packet *decoded* is too long duration [18:45:35 CEST] <AstralStorm> ffprobe says duration is correct and start is also correctly offset (by one frame delay) [18:46:19 CEST] <molavy> Mavrik: if i know who to export and and parameters i don't ask it here [18:46:32 CEST] <molavy> how [18:46:48 CEST] <AstralStorm> no, wait, last packet is overlong, wtf [18:46:54 CEST] <AstralStorm> why. [18:55:25 CEST] <molavy> i try this [18:55:30 CEST] <molavy> with no success [18:55:31 CEST] <molavy> ffmpeg -i NewMorning.wav -f s16le -ar 8000 -ac 1 -acodec pcm_s16le output.pcm [18:58:31 CEST] <AstralStorm> hmm, it seems I mucked with pts/dts on flushed packets [18:58:44 CEST] <AstralStorm> so this is potentially why I got wrong packet duration [19:00:54 CEST] <molavy> there is no one here that help me [19:01:08 CEST] <molavy> i just want make an output [19:03:47 CEST] <Lushi> JEEB, DHE, Maverik: I filed a bug. https://trac.ffmpeg.org/ticket/4733 [19:06:14 CEST] <AstralStorm> hmm, weird, the extra is actually similar to what output of fdkenc is [19:06:41 CEST] <AstralStorm> last packet made by fdkenc in -G 1 mode is overlong too and even more so [19:12:52 CEST] <molavy> can some one help me with this [19:13:07 CEST] <AstralStorm> oh, I see, ffmpeg does not support edit list? [19:13:18 CEST] <molavy> i just want convert a file for use in phone call greeting system [19:13:24 CEST] <AstralStorm> as in ISO/IEC 14496-12:2003 [19:14:44 CEST] <AstralStorm> hmmh [19:20:05 CEST] <molavy> 302 user with no answer [19:20:40 CEST] <molavy> i forced go to windows and use windows tools with gui [19:39:47 CEST] <Lushi> JEEB, you still there? [19:39:55 CEST] <JEEB> never [19:40:08 CEST] <Lushi> :) [19:43:25 CEST] <Lushi_> ...sorry..my network went down for a minute... [19:49:58 CEST] <oscargot> @durandal_1707: yes it is fairly slow and takes about 150ms per conversion when resolution is 1080p [19:51:03 CEST] <Mavrik> hmm, that sounds overly excessive [19:51:04 CEST] <durandal_1707> oscargot: I forgot what is slow? [19:51:13 CEST] <Mavrik> considering it's a simple swap [19:51:21 CEST] <Mavrik> are you copying stuff around too much? [19:52:22 CEST] <oscargot> here is my original question: Hello, is there anyway to directly encode a NV12 pix format image to MPEG4 using libav rather than having to first convert it to YUV420P sws_scale? Or optimise the conversion step? [19:52:38 CEST] <oscargot> Mavrik: I am trying to convert NV12 to YUV420p [19:52:42 CEST] <Mavrik> I mean, there's no way your pix format conversion should take more than the encode. [19:52:47 CEST] <Mavrik> Yes. [19:54:56 CEST] <oscargot> hmmmm the time measurement was done by measure the time right before/after sws_scale, I'm not doing any copying myself between this time. [19:55:46 CEST] <Mavrik> If you're on an embedded device, you could do conversion youself in NEON or something [19:56:00 CEST] <Mavrik> As I said, NV12 -> YUV420p means just deinterleaving a plane [19:56:28 CEST] <oscargot> Mavrik: yes that's true, I've not heard of NEON before, I will look into it thanks [19:56:39 CEST] <Mavrik> NEON is an ARM instruction set [19:56:44 CEST] <Mavrik> what kind of device do you have? [19:59:19 CEST] <oscargot> I'm working with an arm processor so it should apply? [20:01:23 CEST] <Mavrik> If it has that instruction set. [20:01:30 CEST] <Mavrik> (It's not mandatory) [20:37:17 CEST] <benbro1> what's the standard FPS for browsers? 29 fps? [20:38:22 CEST] <JEEB> I don't think there is such a thing :P [20:38:39 CEST] <JEEB> you encode with the frame rate your content is, or do VFR or whatever [20:38:47 CEST] <JEEB> all containers contain timestamps [20:39:04 CEST] <JEEB> (I mean, all of the ones supported by browsers) [20:40:50 CEST] <benbro1> ok. thanks [20:43:30 CEST] <AStorm> so folks what about decoding m4a with ffmpeg gaplessly? [20:43:36 CEST] <AStorm> as it is now the final packet is overlong [20:43:59 CEST] <AStorm> and I need to handle that in a way that does not break other decoders [20:47:42 CEST] <benbro1> JEEB: should I use 25 fps or 29 fps for live streaming? [20:51:03 CEST] <JEEB> benbro1: whatever your source rate is? [20:51:14 CEST] <JEEB> I mean, I see no fucking reason to change your source rate for anything [20:52:30 CEST] <JEEB> AStorm: if it's supposedly marked with edit list then it might be better for you to try and get that virtual timeline stuff into mainline, although it will be quite funky and hard most probably [20:53:18 CEST] <AStorm> JEEB: I think both are [20:53:26 CEST] <AStorm> even ffmpeg puts an edit listin [20:53:29 CEST] <JEEB> yeah [20:53:32 CEST] <JEEB> there's always an edit list [20:53:40 CEST] <JEEB> but it's usually for the whole length, or implicit [20:53:50 CEST] <benbro1> JEEB: I had a file with 1000 fps. not sure why [20:53:51 CEST] <Mavrik> Also, you certanly shouldn't use 29 fps :P [20:54:01 CEST] <AStorm> fdkenc seems to put in elst too [20:54:14 CEST] <benbro1> Mavrik: what's wrong with 29 fps? what should I use instead? [20:54:21 CEST] <AStorm> yeah, 29.97 is the right FPS ;p [20:54:24 CEST] <JEEB> benbro1: sounds like something misparsed FLV's timebase as frame rate :P [20:54:28 CEST] <JEEB> AStorm: 30000/1001 [20:54:32 CEST] <AStorm> yup [20:54:44 CEST] <Mavrik> Isn't lately 60 or 30 more like default [20:54:50 CEST] <AStorm> yep [20:54:52 CEST] <benbro1> I converted wmv to mp4 and the result was 1000.1 fps [20:54:55 CEST] <AStorm> 60 FPS is all the rage [20:54:56 CEST] <Mavrik> or is US still recording stuff in 59.97? [20:55:11 CEST] <AStorm> also 50 FPS [20:55:29 CEST] <Mavrik> 50 is for interlaced PAL 25 [20:55:33 CEST] <Mavrik> rarer these dasy [20:55:35 CEST] <Mavrik> *days [20:55:46 CEST] <AStorm> you mean noninterlaced [20:56:23 CEST] <Mavrik> Erm, no. [20:56:41 CEST] <Mavrik> But I did mean fieldrate :) [20:57:25 CEST] <benbro1> so I should use "-r 20.07" ? [20:57:33 CEST] <benbro1> sorry "-r 29.97" [20:57:51 CEST] <Mavrik> benbro1, you should not use anything [20:57:57 CEST] <Mavrik> and leave the framerate you have [20:58:11 CEST] <Mavrik> framerate conversion just ruins quality [20:58:12 CEST] <benbro1> but what if the source frame rate isn't ideal? [20:58:12 CEST] <JEEB> I'm pretty sure your content isn't 1000fps but rather WMV's VFR antics caused you to have a lulzy timebase [20:58:39 CEST] <JEEB> VMW encoders have been known to create VFR if stuff doesn't change on screen [20:58:43 CEST] <Mavrik> benbro1, how will deleting frames and readding duplicates improve the situation? It won't, if it's ruined, it's ruined. But I doubt yours is, it just has wrong metadata. [20:59:30 CEST] <JEEB> of course it's possible that whatever re-encoded your VMW drove off the retard cliff [20:59:51 CEST] <JEEB> but then you take that WMV into your pretty hands again and start jerking the WMV into something less proprietary [21:00:51 CEST] <Mavrik> *shudders* [21:02:17 CEST] <benbro1> ok [21:15:21 CEST] <benbro1> can ffmpeg convert pcap capture of RTP to a video? [00:00:00 CEST] --- Fri Jul 24 2015

1 0

ffmpeg-devel.log.20150722
by burek 23 Jul '15

23 Jul '15

[01:31:06 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:385eb066ce8c: avformat/asfdec_f: Increase the amount of information provided in cases of errors [01:31:07 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:5d79a0731675: avformat/asfdec_f: Do not print errors if packets do not start with ECC [01:31:08 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:0671dc5c5327: avformat/asfdec_f: Improve packet resync heuristic [05:18:39 CEST] <cone-051> ffmpeg 03Niklesh 07master:93e80a343bfb: movtextenc.c: Reorganize the code for easier maintenance [05:18:39 CEST] <cone-051> ffmpeg 03Niklesh 07master:d373b508b540: movtextenc.c: Add support for text highlighting [08:58:42 CEST] <dericed> IETF session on FFV1/MKV about to start. jabber xmpp:dispatch@jabber.ietf.org?join audio http://congresshall3.conf.meetecho.com:8000/congresshall3.opus meetecho https://congresshall3.conf.meetecho.com/meetecho/login.jsp?ietf=dispatch [11:47:20 CEST] <cone-500> ffmpeg 03Carl Eugen Hoyos 07master:35b33f1a195c: lavf/mxfdec: Set codec_tag AVup for Avid 1:1 input. [11:47:20 CEST] <cone-500> ffmpeg 03Carl Eugen Hoyos 07master:6b2bb3d231f8: Cosmetics: Reindent after last commit. [12:20:21 CEST] <Compn> dericed : cool. that was 3am for me ;) [14:20:26 CEST] <iive> michaelni: are you around? [14:21:18 CEST] <durandal_1707> michaelni: does this crash for you: ffmpeg -f lavfi=color=s=2x2:d=4000 -vf reverse -f null - [14:21:43 CEST] <D404|Ghetto> i wouldnt be surprised if it did [14:23:06 CEST] <michaelni> "./ffmpeg -f lavfi -i color=s=2x2:d=4000 -vf reverse -f null -" does not crash here [14:23:44 CEST] <durandal_1707> ahh then my system realloc is borked [14:24:14 CEST] <iive> Raz-: michaelni is around, you can talk with him in query or here. however you like. [14:26:44 CEST] <D404|Ghetto> durandal_1707: that would seem pretty unlikely [14:26:53 CEST] <D404|Ghetto> what sort of crash is it [14:28:01 CEST] <durandal_1707> Segmentation fault, and just when are few frames left to flush [14:30:44 CEST] <D404|Ghetto> durandal_1707: hmm [14:30:48 CEST] <D404|Ghetto> it could be a bug maybe? [14:30:55 CEST] <D404|Ghetto> iirc i tested and valgrind was clean [14:31:07 CEST] <D404|Ghetto> if youre on bsd it may be pickier [15:00:08 CEST] <cone-469> ffmpeg 03hSÇ 07master:3e35f8efa18d: avcodec: loongson optimize pixblockdsp with mmi [15:00:09 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:9837d3b06844: avformat/asfdec_f: Parse ECC byte according to spec [15:21:20 CEST] <AstralStorm> hello, I'm having trouble with ffmpeg finding muxer for m4a... after digging I've ended up at av_match_name [15:21:24 CEST] <AstralStorm> http://ffmpeg.org/doxygen/trunk/avstring_8c_source.html#l00342 [15:21:48 CEST] <AstralStorm> ipod muxer says extensions = "m4v,m4a" [15:22:49 CEST] <dericed> Compn: was 3am for me too, zzzzzz [15:23:15 CEST] <AstralStorm> line 352 there is fishy, shouldn't it be FFMIN? [15:24:25 CEST] <AstralStorm> not even that, this len seems quite wrong [15:30:16 CEST] <AstralStorm> nah, it is correct [15:30:59 CEST] <AstralStorm> other than tossing namelen for the non-first entry [15:31:52 CEST] <AstralStorm> nah, the loop will exit early then, it's ok [15:32:51 CEST] <AstralStorm> wait a second, what is av_match_name supposed to do? it's not av_match_extension [15:33:00 CEST] <AstralStorm> it compares the whole name to the extension [15:33:55 CEST] <AstralStorm> the bug is in av_guess_format [15:34:18 CEST] <AstralStorm> no, again, av_match_ext is used [15:34:57 CEST] <nevcairiel> av_match_ext is a tiny wrapper around av_match_name that strips the extension of the filename and then compares it to the extension list with av_match_name [15:35:00 CEST] <nevcairiel> codel ooks fine [15:35:26 CEST] <AstralStorm> indeed, the path contains more dots [15:35:42 CEST] <AstralStorm> I wouldn't suspect strrchr to be wrong [15:36:39 CEST] <AstralStorm> but, it's avio_open failing with protocol not found [15:36:46 CEST] <AstralStorm> av_register_all is certainly called [15:37:09 CEST] <nevcairiel> maybe your build forgot to include the file protocol? [15:37:15 CEST] <nevcairiel> that happened to me once [15:37:17 CEST] <AstralStorm> no, it's definitely there [15:37:25 CEST] <AstralStorm> and the encoder worked for writing raw AAC bitstreams [15:37:30 CEST] <AstralStorm> now I'm trying to get m4a [15:37:37 CEST] <AstralStorm> that should pick ipod muxer [15:38:47 CEST] <AstralStorm> hmm, is the list of protocols in configure comma separated or space separated or? [15:39:04 CEST] <nevcairiel> if you use --enable-protocols= then its comma separated [15:39:11 CEST] <nevcairiel> configure will give you an output of things enabled [15:39:13 CEST] <nevcairiel> so you can check [15:39:31 CEST] <nevcairiel> check that your muxer, codec and protocol are there [15:39:52 CEST] <AstralStorm> --enable-protocol=concat,data,file,subfile \ [15:40:08 CEST] <nevcairiel> that should probably work [15:40:40 CEST] <nevcairiel> but better to double check [15:41:25 CEST] <AstralStorm> how can I list the protocols? [15:44:06 CEST] <AstralStorm> ah, I see, I need to iterate av_oformat_next [15:56:25 CEST] <AstralStorm> interesting, now it worked? I potentially have a race condition somewhere vs av_register_all [15:56:44 CEST] <AstralStorm> that said: av: Encoder did not produce proper pts, making some up. (aac native) [15:57:04 CEST] <AstralStorm> I'll do it manually of course [16:09:20 CEST] <wm4> AstralStorm: the register functions are not thread-safe [16:09:24 CEST] <wm4> they also touch global state [16:09:26 CEST] <wm4> have fun [16:10:26 CEST] <AstralStorm> yup, noticed [16:27:35 CEST] <Swinehorn> hey guys. I'm compiling ffmpeg with VS2013 (--toolchain=msvc), and I want add libfdk-aac. Unfortunately fdk-aac doesn't have VS solution, I compiled .lib file by MinGW but this lib file is not valid for linking with ffmpeg. [16:27:36 CEST] <Swinehorn> can you please advice, how to make compatible fdk-aac .lib file ? [16:28:12 CEST] <nevcairiel> you will need to build it with msvc somehow, building your own solution if they dont provide one [16:28:19 CEST] <nevcairiel> mingw static libraries do not work with msvc builds [16:28:32 CEST] <j-b> take the .dll generated by MinGW. Create the .lib from it. [16:28:45 CEST] <cone-469> ffmpeg 03Tom Butterworth 07master:c7e6443441ed: Support the Hap chunked frame format [16:29:01 CEST] <nevcairiel> i suppose that can work if you can accept a separate dll [16:29:01 CEST] <Swinehorn> j-b, you're genius)) [16:29:15 CEST] <AstralStorm> weird, I get the "making some up" message when I set pts to 0 [16:29:28 CEST] <AstralStorm> essentially setting them to -1024 for first frame, 0 for second etc. [16:30:32 CEST] <j-b> dumpbin /exports "BLA.dll" > "BLA.lib" in the Visual Studio Command Prompt [16:31:25 CEST] <Swinehorn> I'll try thx very much) [16:32:24 CEST] <AstralStorm> mpv now says the pts are fine, first frame is -0.021s (48000 ksps) [16:38:15 CEST] <cone-469> ffmpeg 03Luca Barbato 07master:3ae0e721c7b6: checkasm: Always link statically [16:38:16 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:c1692439e081: Merge commit '3ae0e721c7b6e0483801b9039b3d140e3b68b7f5' [16:41:49 CEST] <AstralStorm> ah, I see, I should also set dts, seems ffmpeg actually checks this [16:46:07 CEST] <cone-469> ffmpeg 03Janne Grunau 07master:e605bf3b590d: checkasm: remove empty array initializer list in h264pred test [16:46:08 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:e1b5a2e46ef0: Merge commit 'e605bf3b590d295f215fcc9fd58eb11be55b68cb' [16:46:35 CEST] <durandal_1707> D404|Ghetto: checking pts instead of frames in filter frame fixes it [16:53:21 CEST] <cone-469> ffmpeg 03Anton Khirnov 07master:ecee1148af49: qsvenc_hevc: use the correct HW plugin UID [16:53:22 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:fbdbe7a5d262: Merge commit 'ecee1148af4989e1f9e16f0cdc9f98ad2045538c' [17:06:21 CEST] <cone-469> ffmpeg 03Alexandra Hájková 07master:5655236a6720: asfdec: factor out seeking to the Data Object outside while [17:06:22 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:fce350be0e75: Merge commit '5655236a67203d923755f285584c6e68abe7e33f' [17:12:37 CEST] <Swinehorn> j-b, ffmpeg with fdk-aac successfully compiled and works, great!!) [17:13:19 CEST] <cone-469> ffmpeg 03Alexandra Hájková 07master:93f16f338f9e: asfdec: close the demuxer properly when read_header is failing [17:13:20 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:fa7defc89a7b: Merge commit '93f16f338f9e8aba0c006752eb3afc3fe6e137fd' [17:21:59 CEST] <durandal_1707> D404|Ghetto: sent patches [17:22:43 CEST] <cone-469> ffmpeg 03Alexandra Hájková 07master:2a187a074a7f: asfdec: avoid crash in the case when chunk_len is 0 or pkt_len is 0 [17:22:44 CEST] <cone-469> ffmpeg 03Vittorio Giovara 07master:ea4d46e72945: dds: Fix enum declaration [17:22:45 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:6b9be608ce92: Merge commit '2a187a074a7f5ad9f01f72ac9715ddfcb2dbb8ec' [17:22:46 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:eaf15bba03b4: Merge commit 'ea4d46e72945cba37feb7aa154eb970732f513e4' [17:23:59 CEST] <D404|Ghetto> durandal_1707: i dont get why that fixes it [17:26:59 CEST] <nevcairiel> different sizes in those variables? [17:28:44 CEST] Action: durandal_1707 laughs at libav diff filter [17:30:08 CEST] <nevcairiel> i agree with D404|Ghetto, those two arrays are always allocated at the same time to the same size, it makes no sense for the patch to help [17:31:14 CEST] <nevcairiel> unless fast realloc does something weird [17:31:18 CEST] <nevcairiel> which it may [17:31:22 CEST] <nevcairiel> its one of those magic functions [17:31:29 CEST] <nevcairiel> .. so their size does desync [17:31:47 CEST] <D404|Ghetto> i dont think thats possible [17:32:14 CEST] <D404|Ghetto> hmm... oh right... [17:32:40 CEST] <D404|Ghetto> [B[Bnope [17:32:55 CEST] <D404|Ghetto> it's impl makes it clear they ought to be the same... [17:33:13 CEST] <nevcairiel> why [17:33:18 CEST] <D404|Ghetto> http://ffmpeg.org/doxygen/trunk/mem_8c_source.html#l00480 [17:33:51 CEST] <D404|Ghetto> size and min=size are a function of nothing but themselves [17:34:07 CEST] <nevcairiel> plus integer division [17:34:17 CEST] <D404|Ghetto> but theyre the same [17:34:21 CEST] <nevcairiel> and on 32-bit the pointer is only 4 bytes [17:34:33 CEST] <nevcairiel> so it could behave differently [17:34:39 CEST] <D404|Ghetto> ah... [17:34:48 CEST] <D404|Ghetto> how evil. [17:35:24 CEST] <D404|Ghetto> i guess the real fix is to check both or track manually... [17:35:30 CEST] <D404|Ghetto> both of which make av_fast_realloc utterly useless [17:35:33 CEST] <D404|Ghetto> compared to av_realloc [17:35:33 CEST] <D404|Ghetto> :D [17:39:46 CEST] <cone-469> ffmpeg 03Vittorio Giovara 07master:57214b2f7f9b: dds: Fix palette decoding [17:39:47 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:d08d8b61aa9f: dds: Fix 32bpp bitmaps decoding [17:39:48 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:05cf687d130a: Merge commit '57214b2f7f9b1ccfd61e232e8989b5ee850f169c' [17:39:49 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:b68e445f9b34: Merge commit 'd08d8b61aa9f07d3ea993fe5392f7408c958d221' [17:47:11 CEST] <Compn> iive : bulgarian hosting offer on the list now [17:48:05 CEST] <iive> :) [17:56:40 CEST] <cone-469> ffmpeg 03Paul B Mahol 07master:787d370e1424: avfilter: add deband filter [18:00:05 CEST] <iive> i'm not involved with any of these firms, so you can be sure I won't take over ffmpeg :D [18:00:35 CEST] <Raz-> :)) [18:07:39 CEST] <cone-469> ffmpeg 03Vittorio Giovara 07master:a16854892c3a: dds: Add a rgba fate test [18:07:40 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:d231d2be7101: Merge commit 'a16854892c3af945d3ab0015699a0c9884f0a89a' [18:17:24 CEST] <cone-469> ffmpeg 03Vittorio Giovara 07master:21c90d86d27c: mpegvideo: Add missing include [18:17:25 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:1d7fa1ac8932: Merge commit '21c90d86d27c2143354c7d782050a779b0986eb1' [18:52:07 CEST] <cone-469> ffmpeg 03Donny Yang 07master:a906e86a8dbd: apng: Fix decoding images with the PREVIOUS dispose op [19:25:24 CEST] <AstralStorm> hmm, about this AAC, there's a nice icassp 2006 paper about multidimensional linear programming for quantization mode, long block and short block picking [19:25:41 CEST] <AstralStorm> Multidimensional Optimization of MPEG-4 AAC Encoding; Claus Bauer, Matt Fellers, Grant Davidson; Dolby [19:25:46 CEST] <AstralStorm> not sure about patents on this [19:29:28 CEST] <JEEB> ffmpeg doesn't really care too much about patents, otherwise a lot of the stuff wouldn't be implemented [19:29:50 CEST] <JEEB> like DTS or Dolby formats and friends [19:31:40 CEST] <AstralStorm> I think they've posed the problem for CBR there, for some rate threshold [19:32:20 CEST] <AstralStorm> for VBR the optimization would aim at some PSNR or perceptual entropy value... for ABR this algorithm does not work, as you need additional constraint [19:32:44 CEST] <AstralStorm> that constraint is unfortunately nonlinear [19:35:09 CEST] <AstralStorm> I mean, for VBR there's a nice equality constraint on PE or PSNR [20:00:33 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:c40ecffd31d0: Replace AV_PKT_DATA_QUALITY_FACTOR by AV_PKT_DATA_QUALITY_STATS [20:00:34 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:557e011bf167: avcodec/libxavs: Export pict_type in side data [20:16:40 CEST] <atomnuker> If I give init_put_bits() a NULL buffer can I use the context to just measure how many bits it has pretended to write? [20:17:59 CEST] <durandal_1707> nope that will crash IIRC [20:33:20 CEST] <atomnuker> PutBits is used by a ton of encoders [20:33:45 CEST] <atomnuker> Would it be a crime to put another field in the context to make the put_bits() only increment it and not actually write anything? [20:41:45 CEST] <iive> can you do that without putting condition (if) in all the related functions? [20:42:43 CEST] <atomnuker> yep [20:42:55 CEST] <atomnuker> only in put_bits() and in put_bits_count() [20:43:11 CEST] <kierank> lglinskih: hi [20:43:13 CEST] <iive> but these are the expensive function.... [20:43:37 CEST] <atomnuker> it's just a single comparison, can it really be that bad? [20:43:41 CEST] <iive> if()s are expensive, one wrong prediction and you have stall of 300 cycles. [20:44:25 CEST] <nevcairiel> yeah those functions are over-optimized, overhead in them should be avoided [20:44:42 CEST] <kierank> atomnuker: you could do it how x264 does it [20:44:56 CEST] <kierank> it uses a macro to redefine the function [20:45:06 CEST] <nevcairiel> i suppose that might work [20:45:09 CEST] <atomnuker> yeah, that was the original plan [20:55:04 CEST] <BBB> I think thats better/easier, yes [20:56:08 CEST] <iive> i don't remember how they are written... do they have a check for remaining space? [20:57:21 CEST] <iive> aka, one possible hack is to give them buffer with length zero, the check would always fail and prevent actual write to the buffer. The hack would be to increment the written amount. [20:58:11 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:caba19a5becc: avcodec/dnxhdenc: Set pict type for AV_PKT_DATA_QUALITY_STATS correctly [20:58:12 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:975257ffe650: avcodec/libx264: Export choosen pict_type [21:11:28 CEST] <lglinskih> kierank: hi! I came back to work: I'm trying to test draw_horiz_band for other codecs. Now I'm playing with h263 test file. [21:11:36 CEST] <kierank> ok [21:11:44 CEST] <kierank> wm4 had some comments about seek test [21:14:26 CEST] <wm4> lglinskih: you had this idea to decode the file completely in a first pass, and then trying again with seeking, and comparing the results to the first pass [21:14:43 CEST] <wm4> lglinskih: that sounds like a very good way to get a useful test [21:19:47 CEST] <lglinskih> wm4, kierank: ok! [21:23:31 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:3d083f6ffd8c: avformat/dump: Also print pict_type in dump_sidedata() for AV_PKT_DATA_QUALITY_STATS [21:23:32 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:5362df2ee383: avcodec: remove unused sd variables [21:23:44 CEST] <wm4> lglinskih: another, more fuzzy test: check where a seek actually lands, and see whether the result is acceptable [21:24:00 CEST] <wm4> if you seek to 5 seconds, and end up at 5 minutes, the result is obviously not so great [21:25:51 CEST] <lglinskih> =) [21:26:46 CEST] <wm4> on the other hand, you can't expect seeks to be exact, of course, only within what's technically possible [22:23:26 CEST] <cone-469> ffmpeg 03Jovan Zelincevic 07master:631496e057a2: avcodec: Table creation for AAC_fixed_decoder (PS-module) [22:23:27 CEST] <cone-469> ffmpeg 03Djordje Pesut 07master:5fd81cf6f082: avcodec: Implementation of AAC_fixed_decoder (PS-module) [22:23:28 CEST] <cone-469> ffmpeg 03Nedeljko Babic 07master:978a8540b6ce: tests: Add aac_fixed decoder test [22:35:44 CEST] <cone-469> ffmpeg 03Michael Niedermayer 07master:4845f6687d4f: fate: Make ffprobe tests depend on avdevice [23:06:15 CEST] <BBB> wm4: frame-exact seeking is not so hard, in fact [23:07:09 CEST] <wm4> but you can't do it in the demuxer [23:08:36 CEST] <BBB> oh demuxer only test [23:08:40 CEST] <BBB> ok fine nevermind then [23:36:28 CEST] <cone-469> ffmpeg 03Nedeljko Babic 07master:a9d986c2ced8: avcodec: Minor macro polishing [23:36:29 CEST] <cone-469> ffmpeg 03Jovan Zelincevic 07master:9e3135f49e45: Edit documentation and versioning for the fixed point AAC decoder [00:00:00 CEST] --- Thu Jul 23 2015

1 0

ffmpeg.log.20150722
by burek 23 Jul '15

23 Jul '15

[00:58:40 CEST] <edoceo> Is one pixel format better/faster than another? Maybe one format is more lossless? [00:59:25 CEST] <Mavrik> Uhm [00:59:40 CEST] <Mavrik> Different pixel formats lose different amount of data (and compress differently). [00:59:50 CEST] <Mavrik> Of course, you're dependant on what your player supports. [01:00:05 CEST] <Mavrik> YUV420P is pretty much the only widely supported pixel format. [01:10:59 CEST] <edoceo> My player is ffmpeg, I just need a high quality (and hopefully fast) format for ffmpeg to process, then at the end I make widely compatible format [01:11:23 CEST] <Mavrik> What are you grabbing? [01:11:29 CEST] <edoceo> Like, I have N number of source videos, I process and stitch together with ffmpeg, then finalise [01:11:40 CEST] <edoceo> Most of the source video is from webcams [01:11:50 CEST] <edoceo> some is recorded browser sessions [01:12:09 CEST] <edoceo> Output eventually is scaled/cropped to 640x480 [01:12:14 CEST] <Mavrik> Avoid conversions of pixel format then, you probably have yuv420 anyway [01:12:40 CEST] <Mavrik> Depending on speed size, use losless x264 (slower) or huffyuv (faster) for intermediate storage [01:12:53 CEST] <Mavrik> Encode output to H.264 in YUV420p for maximum compatiblity [01:13:36 CEST] <edoceo> Yea, most of the source webm files are yuv420p [01:16:34 CEST] <Mavrik> Yeah, you're going to need better equipment than webcams to get YUV444 or something :) [01:16:51 CEST] <edoceo> Cool, so now I have an unknown pixel format issue [01:16:58 CEST] <edoceo> I use this command to make a grid: http://edoceo.io/paste?p=rqyZO0 [01:17:37 CEST] <Mavrik> unknown pixel format issue? [01:17:38 CEST] <edoceo> But when I ffprobe I get this: http://edoceo.io/paste?p=bjCvDu [matroska,webm @ 0x68a2a0] Could not find codec parameters for stream 0 (Video: mpeg2video, 640x480): unspecified pixel format [01:18:39 CEST] <Mavrik> That might be a bug. [01:18:57 CEST] <Mavrik> since I see nothing wrong with your command line [01:19:03 CEST] <Mavrik> Can you try with 2.7.1 ffmpeg? [01:19:12 CEST] <Mavrik> grab one of the static builds [01:20:59 CEST] <edoceo> The command I sent has my local version but I also get the error from a recent static: ffmpeg version N-63893-gc69defd Copyright (c) 2000-2014 the FFmpeg developers [01:21:14 CEST] <Mavrik> hrmf. [01:21:54 CEST] <Mavrik> does the output play? [01:22:00 CEST] <Mavrik> might just be an analyze issue [01:22:26 CEST] <edoceo> Does not play, but there was a newer static than what I had, so gimmie a second [01:23:42 CEST] <edoceo> That latest static still has the issue [01:24:21 CEST] <edoceo> v: ffmpeg version N-48359-g6b96c70-static [01:24:43 CEST] <Mavrik> Might wanna report it then. [01:24:59 CEST] <edoceo> Can I force a pix_format for input so it woudl know? [01:25:11 CEST] <Mavrik> yes, add -pix_fmt before -i [01:27:18 CEST] <edoceo> Ok, I'll know in about 30 minutes if that work, thanks! [01:43:59 CEST] <edoceo> If I put -pix_fmt before -i I get: Option pixel_format not found. [03:00:10 CEST] <blurider1> If I compile ffmpeg with NEON support for ARM CPUs, will it work on cpus without NEON support? [03:03:00 CEST] Last message repeated 1 time(s). [08:56:18 CEST] <kelnoky> just wanted to make an alias for an ffmpeg command I use frequently, however it doesnt seem to work with the output "-vcodec: No such file or directory". what did I do wrong here: alias louder='ffmpeg -i $1 -vcodec copy -af volume="4.0" $2' [08:57:42 CEST] <pzich> kelnoky: is this bash? you can't use parameters in bash aliases [08:57:56 CEST] <kelnoky> pzich: yeh, it is and oh.... [08:58:03 CEST] <kelnoky> I just want something so simple [08:58:09 CEST] <pzich> a simple solution is to write a function, ala http://stackoverflow.com/a/7131683 [08:58:23 CEST] <kelnoky> someone suggested I use a function, but even with a tutorial article on it I dont get it :D [08:59:01 CEST] <kelnoky> hm [08:59:04 CEST] <pzich> it's just: louder() { ffmpeg -i $1 -vcodec copy -af volume="4.0" $2 } [08:59:10 CEST] <kelnoky> that answer is better than the tutorial I had though [08:59:19 CEST] <kelnoky> yeh, I think I got it now [09:00:05 CEST] <kelnoky> thanks! [09:00:09 CEST] <pzich> np [09:00:33 CEST] <pzich> also, quote your arguments if you want to handle whitespace and other special chars correctly [09:01:54 CEST] <kelnoky> ah, so if my filenames had whitespaces or other stuff in it [09:01:58 CEST] <kelnoky> ok, thanks [09:02:08 CEST] <pzich> the second answer has an example, if you need it [09:02:11 CEST] <kelnoky> single quotes or double quotes? [09:02:17 CEST] <kelnoky> ' or " [09:02:19 CEST] <pzich> double, single won't expand the arguments [09:02:22 CEST] <kelnoky> ok [09:02:27 CEST] <pzich> you'll get a literal '$1' [09:02:40 CEST] <kelnoky> it works :) [09:43:53 CEST] <bse__> i'm trying to convert a security camera video (mxg format) to mp4, for easy access in a webbrowser. the video is at 5 to 10 fps (varying). ideally i'd like to ignore the timestamps in the input video and encode it frame-by-frame to mov (no duplication, no dropping). though whatever i try, i keep dropping frames. i'm stuck with ffmpeg 0.8.17 (debian wheezy). any ideas for a -vf i could try? [09:45:35 CEST] <chungy> wheezy doesn't even have ffmpeg, it's libav pretending to be ffmpeg. Either way, the release is ancient and you should probably upgrade :P [09:47:53 CEST] <bse__> yea :/ should try harder to convince my boss [09:48:29 CEST] <chungy> can build ffmpeg manually if you have a compiler and all [09:51:25 CEST] <bse__> loooots of dependencies iirc... but worth looking into, good idea thx [10:01:58 CEST] <DHE> not THAT many dependencies. third party codecs, yes if you want them. x264 is a big one. otherwise ffmpeg will tolerate most dependencies missing and build with stock codecs only [10:04:18 CEST] <termos> is there an easy way for either ffprobe or ffmpeg to dump the onMetaData so I can view it? [10:07:19 CEST] <termos> aha, flvmeta seems to do it but it only accepts files not rtmp streams [10:22:16 CEST] <Marko> Hello does anybody know value range of libschroedinger -q:v parameter? [10:24:36 CEST] <xerox> I wondered if there was an ffmpeg channel, so happy there is one :) [10:25:33 CEST] <xerox> is it possible to have ffmpeg not print out the version+build options+library versions every time it's launched, but still have the various Input, Output, Stream mapping, and progress output? Can't quite find a loglevel that does that. [10:27:16 CEST] <chungy> -hide_banner [10:27:31 CEST] <xerox> chungy: oh sweet thank you! [11:10:16 CEST] <woew> hi, I need to convert a 24 fps source to 25 fps with identical runtime (by repeating select frames). As a bonus I want to burn in correct timecode (@25 fps). How can I do it? [11:12:11 CEST] <woew> Target is PAL DVD so the conversion to 25 fps would be implicit in my call (-target pal-dvd)? [11:18:17 CEST] <QQ> Hello. I have a simple question: I have a few lossy files and I want to add a silence in them without losing quality. I found out I can create a new file with the "adelay" filter, and it works perfectly. My question is: does ffmpeg reencode the lossy file (mp3 or ac3) to add the silence when I use the adelay filter, or it copies the audio stream without reencoding it? Do I lose quality or not? Thanks a lot for any help! [11:39:51 CEST] <chungy> QQ: ffmpeg will reencode and you will lose quality. You probably want more specialized tools (like mp3splt/mp3wrap) for this... [11:40:23 CEST] <QQ> Thank you for your answer. [14:09:39 CEST] <CustosL1men> hi [14:10:25 CEST] <CustosL1men> I'm trying to build something that needs avcodec_get_context_defaults, avcodec_alloc_context and avcodec_open [14:10:40 CEST] <CustosL1men> but these are not present in 2.6.3 [14:10:44 CEST] <CustosL1men> when last were they present [14:11:07 CEST] <JEEB> see the apichanges file and hope that was documented in there [14:13:18 CEST] <AstralStorm> hello, I'm having some trouble with aac (native) encoder initial padding [14:13:37 CEST] <AstralStorm> it seems that it's not handled correctly - my output is longer by exactly one frame [14:13:49 CEST] <AstralStorm> ffmpeg 2.7.1 [14:14:10 CEST] <techtopia> your better of demuxing source audio [14:14:25 CEST] <AstralStorm> I'm not demuxing anything - import is done in audacity :) [14:14:31 CEST] <techtopia> encoding with nero'd aac encoder or qaac [14:14:38 CEST] <techtopia> then remux into encoded video [14:14:43 CEST] <JEEB> AstralStorm: are you outputting wav with ffmpeG? [14:14:54 CEST] <AstralStorm> no, I'm encoding mpeg 4 aac with libavcodec/libavformat [14:15:07 CEST] <JEEB> yes, but how are you reading that encoded thing into audacity? [14:15:13 CEST] <AstralStorm> ... import? :) [14:15:29 CEST] <JEEB> and what does that do? [14:15:39 CEST] <AstralStorm> audacity can handle aac, no idea how [14:15:47 CEST] <JEEB> probably through lavf/lavc [14:15:58 CEST] <JEEB> but I wouldn't be surprised if it ignores the encoding delay [14:16:00 CEST] <AstralStorm> I'm assuming audacity is handling it correctly... maybe badly [14:16:05 CEST] <JEEB> please file an issue with audacity [14:16:25 CEST] <AstralStorm> how can I check if the pts are correct? [14:16:44 CEST] <AstralStorm> according to mpeg 4, the initial padding frame should have negative pts [14:16:53 CEST] <JEEB> yes [14:17:05 CEST] <JEEB> and I almost forgot to ask your container, but I guess this is MPEG-4 Part 12? [14:17:10 CEST] <JEEB> aka mp4 [14:17:36 CEST] <AstralStorm> yes [14:17:37 CEST] <JEEB> anyways, if the format can note the container delay in some way, the decoded timestamp should be negative in lavc [14:18:12 CEST] <AstralStorm> I'm assuming ffmpeg command line util will decode it fine now [14:18:14 CEST] <AstralStorm> let's see [14:18:27 CEST] <JEEB> if you decode with ffmpeg into wav that will not be correct [14:18:30 CEST] <JEEB> funny enough [14:18:47 CEST] <JEEB> because wav is considered a thing that can't take negative timestamps, so the timestamps are moved to begin from zero :D [14:18:56 CEST] <JEEB> instead of cutting samples from the beginning [14:19:03 CEST] <AstralStorm> uhm, shouldn't ffmpeg just eat the samples while decoding then? [14:19:10 CEST] <AstralStorm> produce no output [14:19:38 CEST] <JEEB> I only remember that a few months ago if you decoded into wav it would move the timestamps and conserve all of it [14:19:48 CEST] <AstralStorm> oh my [14:19:48 CEST] <JEEB> which is incorrect and leads to the wav files becoming longer with each generation [14:19:55 CEST] <iive> Raz-: join #ffmpeg-devel [14:20:13 CEST] <Raz-> iive, okay will do [14:25:41 CEST] <AstralStorm> let's see if faad does the right thing [14:26:10 CEST] <AstralStorm> nope, faad says it's longer too [14:26:10 CEST] <JEEB> faad doesn't as far as I know really read much encoding delay information at all [14:26:18 CEST] <JEEB> it might just cut off a predetermined amount [14:26:28 CEST] <JEEB> or nothing [14:26:33 CEST] <AstralStorm> nah, faad -i said 8.021s instead 8s [14:27:16 CEST] <AstralStorm> maybe the decoder does the right thing... or maybe libavcodec/libavformat writes wrong ADTS [14:27:43 CEST] <AstralStorm> file says this: testpts.aac: MPEG ADTS, AAC, v4 LC, 48 kHz, stereo [14:27:48 CEST] <JEEB> oh raw aac [14:27:55 CEST] <JEEB> that has no way of actually signaling encoder delay [14:27:57 CEST] <AstralStorm> not raw aac, *mpeg adts* [14:28:14 CEST] <JEEB> well yes, that is usually noted as "raw aac" [14:28:17 CEST] <AstralStorm> ahha [14:28:38 CEST] <AstralStorm> ok, so I'll change the extension to m4a, potentially libavformat will guess mpeg-4 part 12 format [14:28:40 CEST] <JEEB> there's two ways of having the AAC bit stream and that's one of them [14:28:55 CEST] <JEEB> the other was whatever some broadcasters use in MPEG-tS [14:29:46 CEST] <JEEB> the whole encoder delay thing was really messy until rather recently, and even then you have to assume that the decoder doesn't drop anything by itself and you can just use the information from the container [14:30:04 CEST] <AstralStorm> hmm, so av_guess_format guesses raw aac for .aac [14:30:10 CEST] <AstralStorm> pretty interesting choice [14:30:24 CEST] <JEEB> because it's not specified in the bit stream itself, so pretty much all of the commercial things just cut off an arbitrary amount of the output [14:30:44 CEST] <JEEB> in containers then you can handle it better, but you have to hope that the decoder doesn't cut anything off in that case [14:30:52 CEST] <JEEB> lavc doesn't but I have no idea about other things [14:31:35 CEST] <JEEB> also when you demux a stream into a raw bit stream, and then remux into mp4 or so you naturally lose the encoder delay since it can't be noted in the raw bit stream (AFAIK) [14:32:14 CEST] <JEEB> L-SMASH's muxer should be able to mux aac while you set the encoder delay, if you need a simple app to create samples with [14:32:21 CEST] <JEEB> I mean, simpler than ffmpeg :D [14:33:11 CEST] <AstralStorm> doesn't ADTS actually have position information? [14:34:02 CEST] <JEEB> I don't remember it having any kind of encoder delay / PTS thing [14:34:09 CEST] <Paranoialmaniac> none [14:34:18 CEST] <JEEB> well there you have it from the horse's mouth [14:34:46 CEST] <JEEB> you can pretty much think of it as one type of raw bit stream for AAC [14:35:39 CEST] <AstralStorm> indeed, ADTS doesn't have anything [14:36:04 CEST] <AstralStorm> maybe one could trick it by setting the length to 0 [14:36:11 CEST] <AstralStorm> but I bet that's wrong and broken [14:36:28 CEST] <AstralStorm> however, why libavformat guesses raw aac for .aac? [14:37:17 CEST] <JEEB> uhh, that just makes sense? [14:37:33 CEST] <JEEB> dot-aac should be raw AAC bit stream [14:37:57 CEST] <JEEB> just like dot-hevc or dot-264 are for raw HEVC or AVC [14:39:24 CEST] <AstralStorm> hmm ok [14:39:56 CEST] <AstralStorm> I'll set it to .m4a and see [14:45:00 CEST] <AstralStorm> which ffmpeg protocol do I need for MPEG-4 Part 12? [14:45:10 CEST] <AstralStorm> as it is, ffmpeg says no URL protocols [14:45:18 CEST] <AstralStorm> I've enabled only a few obvious ones [14:54:00 CEST] <AstralStorm> hmm, it's not a protocol issue, seems there's no m4a muxer [14:54:10 CEST] <AstralStorm> there's m4v and mp4 ones [14:54:17 CEST] <JEEB> m4a should go through movenc methinks [14:54:58 CEST] <AstralStorm> yes, but seems av_guess_format(nullptr, filePath, nullptr) w/ extension m4a doesn't find it [14:55:14 CEST] <AstralStorm> and then libavformat grumbles about missing protocol [14:55:20 CEST] <AstralStorm> there's also "ipod" muxer [14:55:25 CEST] <AstralStorm> I think m4a goes through that [14:55:40 CEST] <JEEB> not sure... lemme check what ffmpeg does when you try to output "m4a" [14:56:09 CEST] <JEEB> ah yes [14:56:14 CEST] <JEEB> seems to go through that one [14:56:24 CEST] <JEEB> Output #0, ipod, to 'test.m4a': [14:56:42 CEST] <AstralStorm> I didn't disable any muxers... is it not enabled by default? [14:57:25 CEST] <JEEB> it should be [15:04:02 CEST] <AstralStorm> oh, maybe it's trying to add rtp hint track [15:05:53 CEST] <AstralStorm> I didn't specifically enable that though [15:19:22 CEST] <AstralStorm> I'm not sure, but it seems there's a bug in av_match_name [15:23:16 CEST] <Niska_> Hi i'm traing to play an m3u8 resource but aftware retriving the playlist the connection seems to be closed for ffplay [15:23:27 CEST] <Niska_> Probably i need to keep the http connection alive to play the m3u8 segments [15:23:38 CEST] <Niska_> If i don't use -headers $'Connection: keep-alive\r\n' the connection it is closed before the playlist download is completed [15:53:35 CEST] <Swinehorn> hey guys. I'm compiling ffmpeg with VS2013 (--toolchain=msvc), and I want add libfdk-aac. Unfortunately fdk-aac doesn't have VS solution, I compiled .lib files by MinGW but this lib file is not valid for linking with ffmpeg [15:54:33 CEST] <Swinehorn> can you please advice, cow to make compatible fdk-aac .lib file ? [16:11:47 CEST] <xerox> I have an H.264 stream that is weird at about a minute from the end, the player dies when playing, but if I skip after it still plays but does not update the time, what tools can I use to see what the problem is and maybe fix it? [16:13:44 CEST] <xerox> actually this happens in different files, is there a way to use ffmpeg to "straighten them up"? make the seeking work, e.g. [16:15:07 CEST] <kepstin-laptop> xerox, if it's just a muxing problem, you could try using ffmpeg to remux the file, e.g. 'ffmpeg -i file.mp4 -c copy file-copy.mp4' [16:15:26 CEST] <xerox> kepstin-laptop: funnily enough, I have already did that, except from flv to mp4 [16:15:39 CEST] <xerox> (indeed copying the audio and video streams) [16:15:45 CEST] <kepstin-laptop> that won't help if any of the data in the streams themselves is messed up tho :/ [16:15:51 CEST] <xerox> yeah :( [16:16:32 CEST] <xerox> ffmpeg seems to be able to do about a trillion different things, so I thought it was worth asking if maybe a sanity check for h264 was in there, somehow [16:19:26 CEST] <kepstin-laptop> If you just want to check to see if ffmpeg sees any issue decoding a file, you can do something like 'ffmpeg -i file.mp4 -an -f rawvideo /dev/null' [16:19:49 CEST] <xerox> is it possible to start from a specific timestamp? [16:20:04 CEST] <kepstin-laptop> that'll decode the file and print any messages the decoder has [16:20:19 CEST] <kepstin-laptop> sure, use -ss (probably as an input option, which means before the -i) [16:20:46 CEST] <xerox> it's an hour long so even if it's fast it is probably easier to skip :D [16:21:34 CEST] <kepstin-laptop> hmm. actually, would be faster to do "-f null -" instead of "-f rawvideo /dev/null" :) [16:21:58 CEST] <xerox> they seem about as fast [16:23:39 CEST] <xerox> it's very intense on the cpu, decoding this fast [16:23:58 CEST] <AstralStorm> actually data itself should be fine [16:24:03 CEST] <AstralStorm> what matters is pts/dts [16:24:18 CEST] <xerox> I got some rrors! [16:24:34 CEST] <AstralStorm> -c copy is not good enough, you need to remux it [16:25:03 CEST] <xerox> http://pastebin.com/raw.php?i=z2AP4CQF [16:26:15 CEST] <xerox> (this is on a 16 minute one) [16:26:46 CEST] <kepstin-laptop> AstralStorm, not sure what you mean, -c copy does remux it. (it won't change any ptd/dts values tho) [16:27:00 CEST] <AstralStorm> exactly, and you need to recalculate them [16:27:09 CEST] <kepstin-laptop> xerox, looks like the video stream's actually corrupt in that file :/ [16:27:19 CEST] <xerox> is there anything that can be done? [16:27:47 CEST] <kepstin-laptop> xerox, if you think the ffmpeg error recovery result looks ok, you could try re-encoding the video, I suppose. [16:27:57 CEST] <xerox> sometimes I do get similar warnings when I go through the flv->mp4 remux [16:28:08 CEST] <xerox> as in it says something about non-monotonous something [16:28:16 CEST] <xerox> but it never made them unplayable before [16:28:58 CEST] <kepstin-laptop> xerox, in this case, it looks like the non monotonic pts is a side-effect of the decoder hitting an error, not the main problem itself. [16:29:59 CEST] <xerox> ok [16:32:41 CEST] <xerox> thank you for the information, much appreciated [16:43:17 CEST] <AstralStorm> hmm, about encoding some more with libavformat and broken aac encoder [16:43:31 CEST] <AstralStorm> seems I get to set negative dts too [16:43:55 CEST] <AstralStorm> but should I set pts to negative or to AV_NOPTS_VALUE? [17:40:25 CEST] <AstralStorm> hmm, ffplay and avplay handle negative pts and dts correctly in MPEG-4 [17:40:41 CEST] <AstralStorm> almost no other applications do [17:40:46 CEST] <blurider> How do I compile ffmpeg with neon support for HEVC? [17:41:03 CEST] <AstralStorm> if there is any, --enable-neon will do [18:00:50 CEST] <elon> I want to blend/overlay three or more equal-resolution and equal-length videos; here's an example of what I'm trying to accomplish: http://www.gfycat.com/ClumsyDesertedDassierat . In the docs, I found two video filters that might be able to accomplish said task, "blend" and "overlay", but I'm not really sure which of the two I should look in to. Can anyone point me in the right direction? (links to relevant docs/forum posts are fine too!) [18:05:47 CEST] <durandal_1707> elon: its not completely obvious but looks both are or something similar is used can't tell from sample [18:07:47 CEST] <blurider> AstralStorm: thanks. compiling now. [18:25:12 CEST] <bencc> what container can I use for capturing a vp8 and opus RTP streams? [18:30:51 CEST] <kepstin-laptop> bencc, probably matroska would make sense. [18:38:34 CEST] <bencc> kepstin-laptop: mkv? [18:52:38 CEST] <AstralStorm> well, it seems the aac native encoder in 2.7.1 is pretty meh quality :) [18:52:48 CEST] <AstralStorm> well, compared to excellent fdkaac [18:55:06 CEST] <JEEB> there have been some improvements lately + there's even bigger patches on the trac, waiting for the author to post a final version on the mailing list [19:00:19 CEST] <AstralStorm> nice [19:00:42 CEST] <AstralStorm> funny thing, my ISO-compatible files are misdecoded everywhere... except by faad :) [19:01:09 CEST] <AstralStorm> fdkaac encoded file with -G 1 and 2 is also misdecoded by faad [19:02:58 CEST] <Niska_> Can avplay keep-alive http connection? [19:06:43 CEST] <AstralStorm> you mean ffplay? [19:06:49 CEST] <AstralStorm> I don't think so [19:07:07 CEST] <AstralStorm> the native aac encoder seems to be using long blocks almost always [19:09:20 CEST] <Niska_> AstralStorm: Yes ffplay [19:21:59 CEST] <nyuszika7h> why does vf=subtitles take like 2-3 minutes every time before the encode starts? [19:45:07 CEST] <Hi-Angel> While building ffmpeg from source, I am getting a bunch of undefined references in the linking step :/ [19:45:27 CEST] <AstralStorm> Hi-Angel: we take no responsibility for broken environments :) [19:46:05 CEST] <Hi-Angel> AstralStorm, well, since the «configure» step worked just fine, it seems the problem with ffmpeg [19:46:22 CEST] <AstralStorm> nope, configure step in ffmpeg is some pretty dumb script [19:46:30 CEST] <AstralStorm> it does what it's told and looks only at headers [19:46:58 CEST] <AstralStorm> it's not autoconf [19:50:03 CEST] <Hi-Angel> AstralStorm, either way, the undefined references are refer to function of ffmpeg, like «av_log» so it's a problem not with my environment. [19:51:18 CEST] <AstralStorm> that is interesting; which ffmpeg version? [19:51:30 CEST] <AstralStorm> also which linker? [19:51:31 CEST] <Hi-Angel> 2.7 [19:52:11 CEST] <Hi-Angel> Hm, I guess the linker is ld. I am not sure, should I mention it, but I compiling with «-flto» [19:52:18 CEST] <AstralStorm> yes, you should've [19:52:23 CEST] <AstralStorm> that's potentially not supported [19:52:58 CEST] <Hi-Angel> Ah, the linker is «gcc» [19:54:12 CEST] <Hi-Angel> Not supported? Why? o.o Aren't it is the performance critical app, which ought to be one of the first to be compiled with -flto? [19:54:27 CEST] <Hi-Angel> s/app/libs [19:55:01 CEST] <kepstin-laptop> my experience with ffmpeg is that it does compile with -flto on 64bit linux, but it probably depends on which options you have enabled. [19:56:03 CEST] <Hi-Angel> kepstin-laptop, well, nothing especial, «-O3 -march=native -flto -fno-stack-protector». But okay, let me try recompile without «flto» [19:57:30 CEST] <kepstin-laptop> do note that when using -flto you probably have to make sure it's using the linker plugin when running ar,ranlib, etc. [19:57:49 CEST] <kepstin-laptop> oh, no, i was mistaken, doesn't work with lto for me [19:58:40 CEST] <Hi-Angel> 9 [19:58:53 CEST] <kepstin-laptop> the errors I get are apparently due to lto not handling inline assembly well, i think [19:59:29 CEST] <Hi-Angel> Is it undefined references? [20:13:40 CEST] <Hi-Angel> Yay, without «-flto» it built _____ [20:22:16 CEST] <Hi-Angel> Hm Interestingly, I found in the «configure» options an option «--enable-lto». I'll try to use, let's see what happen [20:26:07 CEST] <OneSync1983> hello [20:26:20 CEST] <OneSync1983> Good day/evening everyone [20:27:27 CEST] <OneSync1983> I'm kind of new here so, i don't know the exact rules to follow around here [20:28:36 CEST] <c_14> Ask your question, if you have console output upload it to a pastebin service. [20:29:12 CEST] <OneSync1983> Oh great! Well i have one question [20:29:49 CEST] <OneSync1983> Is it possible to trim a video on a certain audiolevel threshold? [20:30:44 CEST] <c_14> What do you mean? Cut the video so it stops when it reaches a certain audio volume? [20:31:51 CEST] <OneSync1983> exactly, so i have a audio fade-out but the video can already be black at that point. So -blackdetect doesn't solve it for me i think [20:34:48 CEST] <Niska_> Hi I've a very low probe score on an m3u8 segment Probing mpegts score:-1094995529 can avoid probing or directly forcing it [20:34:59 CEST] <c_14> OneSync1983: maybe silenceremove with -shortest [20:37:21 CEST] <OneSync1983> c_14: okay i will look into that [20:40:11 CEST] <OneSync1983> But are both of my streams not the same length at origin? [20:41:30 CEST] <OneSync1983> I thought something more in the way of detecting a certain audio level, which also could be used to trim audio files. But then also use this calculation to trim the entire file [20:42:06 CEST] <Niska_> Why ffplay doesn't probe mp4 pointed in m3u8 [20:42:16 CEST] <c_14> Yes, silenceremove will find the "silence" at the end of the audio file, and -shortest should then cut the video stream at that point. [20:43:15 CEST] <OneSync1983> Oh that sound great indeed! [20:44:16 CEST] <OneSync1983> Thanks for your thinking c_14 [20:47:30 CEST] <Hi-Angel> Yay, compilation with «--enable-lto» works fine! But after I tried to use the build lib, I am getting in a player «h264 aren't supported»; what? :/ [20:47:56 CEST] <c_14> Decoding h264? [20:48:27 CEST] <c_14> That should work unless you explicitly disabled things in your configure. [20:48:32 CEST] <c_14> Are you sure it's linking correctly? [20:48:45 CEST] <Hi-Angel> Yep. [20:48:59 CEST] <c_14> To which question? [20:50:36 CEST] <Hi-Angel> I haven't disabled it; to be honest I don't even see anything h264 related except of options that seems to refer to a side tools, or libs Not sure And yes, it likied correctly, at least with no errors [20:51:15 CEST] <c_14> There's stuff like --disable-protocol --disable-codec etc. also --disable-all and --disable-everything [20:51:57 CEST] <Hi-Angel> I only used «--disable-programs» and «--disable-docs» [20:52:30 CEST] <c_14> Is your program linking statically or dynamically? [20:52:46 CEST] <kepstin-laptop> enabling lto build disables some of the inline assembly [20:52:50 CEST] <Hi-Angel> It is a shared lib «libavcodec» [20:52:56 CEST] <kepstin-laptop> dunno whether that affects the h264 decoder. [20:53:17 CEST] <c_14> kepstin-laptop: It might slow it, it shouldn't disable it. [20:53:23 CEST] <Hi-Angel> Okay, anyway, is there a way to enable it explicitly? [20:53:34 CEST] <c_14> --enable-decoder=h264 [20:53:41 CEST] <Hi-Angel> Okay, thank you! [20:53:50 CEST] <c_14> configure should tell you if it's enabled though [20:56:02 CEST] <Hi-Angel> Hm Yep, I see with the old options configure shows in the list of enabled decoders «h264» [20:56:53 CEST] <Hi-Angel> Okay, I'll try then recompile without lto. [21:13:45 CEST] <Bertl> greetings! is there a simple way to check the actual framerate (i.e. the FPS) with ffplay? [21:14:17 CEST] <Bertl> i.e. not the framerate given in the movie, but the framerate achieved when playing it back? [21:22:29 CEST] <kepstin-laptop> actual fps is a tricky subject, since many formats support variable-framerate streams [21:23:59 CEST] <kepstin-laptop> could probably do something with ffprobe and the -show_frames option, then calculate the average time between pts values with a script of some sort. [21:25:21 CEST] <Hi-Angel> Interesting thing: while with flto the «libavcodec.so» is H11Mb, without flto it is& 50 Mb! D: [21:26:28 CEST] <Bertl> kepstin-laptop: the reason I want to check is that I'm currently testing playback with a 60Hz movie and it looks a little odd and jerky [21:26:54 CEST] <Bertl> but ffplay does not complain that the decoding would be too slow or something like that [21:27:08 CEST] <Bertl> actually it doesn't say anything unusual [21:27:09 CEST] <kepstin-laptop> Bertl, that's more likely to be an issue with synchronizing the frame updates to the video card display refresh. Are you on linux? [21:27:22 CEST] <Bertl> yes, Linux it is [21:27:46 CEST] <kepstin-laptop> what graphics card? Have you tried mplayer/mpv/whatever with an opengl video output? [21:27:59 CEST] <Bertl> ati with glamor [21:28:20 CEST] <Bertl> mplayer is too slow to decode it, I failed to mention, it is also 4k :) [21:28:55 CEST] <kepstin-laptop> ... hmm, you're probably not gonna have a great experience with that. I never got the open source ati drivers with glamor to get smooth video going even on my 7850. [21:30:12 CEST] <kepstin-laptop> hmm. or maybe i'm getting different computers confused. it either wasn't smooth or i didn't get working vsync at all - one or the other. [21:31:14 CEST] <Bertl> so how would that ffprobe/-show_frames work? [21:31:37 CEST] <kepstin-laptop> (i'm not sure how mplayer would be 'too slow to decode it', since mplayer and ffplay should both be using the same ffmpeg libraries to do the decoding) [21:32:02 CEST] <Bertl> probably I'm using the wrong mplayer options :p [21:32:22 CEST] <Bertl> at least mplayer tells me that my system is too slow to play the movie [21:32:35 CEST] <kepstin-laptop> Bertl, when you use ffprobe -show_frames it should include the pts value for each frame. You could feed that output into a script that calculates the average delta and maybe the std dev. [21:33:08 CEST] <Bertl> so I run that in parallel to the ffplay, yes? [21:33:14 CEST] <kepstin-laptop> the std dev value would tell you if there's a lot of timing jitter with the frames. if it's low, the video's ok so it's a display problem [21:33:25 CEST] <kepstin-laptop> you'd pipe the output of ffprobe into a script, yeah [21:33:32 CEST] <kepstin-laptop> dunno if there's an easier way to do that. [21:34:41 CEST] <kepstin-laptop> but if mplayer, which should in theory be using the same decoder as ffplay, is printing the 'your system is too slow' message, that means that the video decoder isn't keeping up with the audio speed [21:34:47 CEST] <kepstin-laptop> so it would be displaying at <60fps [21:35:01 CEST] <kepstin-laptop> (could be anything in the video path, really) [21:35:17 CEST] <kepstin-laptop> i dunno how good the a/v sync in ffplay is, it might just not have noticed the issue. [21:36:09 CEST] <Bertl> I think the problem with mplayer is, that in my distro, without special options, it uses 100% of one CPU, where ffplay OTOH, uses all CPUs evenly [21:37:18 CEST] <kepstin-laptop> huh, weird. that means that it's either explicitly disabling the multithreaded decoder, is using a different version of ffmpeg (or maybe even libav, depending on distro), or is bottlenecking somewhere else (bad video output selected maybe?) [21:37:42 CEST] <Bertl> maybe, how to debug? [21:37:49 CEST] <Bertl> (probably wrong channel :) [21:51:39 CEST] <dami0> is there a way to output to standard output when recording/ [21:51:41 CEST] <dami0> ?* [21:57:11 CEST] <BtbN> - [21:58:54 CEST] <dami0> i tried that, it just died [22:01:05 CEST] <dami0> oh, i googled and it turns out i need to specify format. derp. [22:01:10 CEST] <dami0> thanks [22:16:26 CEST] <EO_> Why is the -timeout option not being respected? http://pastebin.com/Yf1Wt0qL [22:17:40 CEST] <EO_> It's documented as being available here: https://www.ffmpeg.org/ffmpeg-all.html#http [22:20:28 CEST] <bencc> can ffmpeg handle webrtc packets (after DTLS decryption)? [22:21:26 CEST] <BtbN> EO_, the documentation seems to be wrong on that parameter. [22:21:37 CEST] <BtbN> There is no timeout option in the http format. [22:21:55 CEST] <EO_> :( [22:23:25 CEST] <EO_> I'm looping over a cheap webcam which can sometimes hang on an HTTP request...and that kills the whole video stream. Timeout would have been perfect. [22:24:33 CEST] <BtbN> use curl or something for the http, and just pipe it into ffmpeg [22:25:11 CEST] <EO_> ffmpeg can take a continuous stream of concatenated JPEG data on stdin? [22:25:23 CEST] <BtbN> it can take everything it would download via http from stdin [22:25:29 CEST] <EO_> neato! [22:25:40 CEST] <BtbN> you might have to tell it what it is though, if it's not obvious [22:26:36 CEST] <EO_> On another subject, have you guys had to deal with cheap chinacam x264 encoded streams? If so, any advice on options to make the streams not present errors all the time? [22:27:26 CEST] <BtbN> chinacam x264? [22:27:27 CEST] Action: EO_ is doing jpeg snapshot looping because the x264 stream decodes atrociously in ffmpeg and vlc... [22:28:38 CEST] <EO_> yeah...lemme paste you some sample errors [22:32:05 CEST] <BtbN> you mean h264 i guess? [22:32:24 CEST] <EO_> yes. sorry about lag, had to kill a spider. :) [22:38:04 CEST] <EO_> http://pastebin.com/2DXJy231 <-- there. this is a sample chinacam experience. both of mine behave like this (different models). [22:38:32 CEST] <BtbN> that looks like one royaly broken h264 stream [22:38:43 CEST] <BtbN> if it comes from a hardware camera, i don't think there's much you can do about it [22:40:46 CEST] <EO_> yeah, I'm doing a "-c copy" sample right now. Even in that mode it's complaining about missing RTP packets! [22:41:04 CEST] <EO_> What's weird is the software they ship with these decodes the stream perfectly. [22:41:09 CEST] <EO_> (ActiveX plugin) [22:42:12 CEST] <BtbN> Well, missing RTP packets don't have anything to do with h264 [22:42:24 CEST] <BtbN> if it's missing half of the streams package, of course the image is broken [22:47:32 CEST] <EO_> yeah, I guess I need to diddle a bit more to see how much of this is transport errors and how much is encoding errors. [23:22:12 CEST] <manfromafar> haaaa time to learn the cli encoding methods and how to get everything nice and neat [00:00:00 CEST] --- Thu Jul 23 2015

1 0

ffmpeg-devel.log.20150721
by burek 22 Jul '15

22 Jul '15

[00:00:46 CEST] <rcombs> also, I use a simple linear search (and in one case a reverse one) instead of a binary search when looking through the fragment index, but I suppose that could change [00:06:16 CEST] <baptiste> rcombs, the code should reuse as much as possible as the mfra code [00:06:33 CEST] <baptiste> I see mfra has an option to specify either dts/pts [00:07:21 CEST] <baptiste> I never understood why they added sidx instead of reusing mfra [00:08:48 CEST] <baptiste> I believe you can execute mov_switch_root just before returning from seek successfully [00:08:55 CEST] <baptiste> that should simplify the function a bit [00:14:49 CEST] <rcombs> uhhhh [00:15:59 CEST] <rcombs> I didn't understand either of the things you just said [00:17:43 CEST] <baptiste> sorry :) [01:00:10 CEST] <wm4> rcombs: is that search over all samples or just all fragments? [01:00:46 CEST] <rcombs> wm4: just fragments [01:00:57 CEST] <wm4> then why care [01:01:31 CEST] <wm4> there shouldn't be that many fragments right? [01:01:43 CEST] <wm4> although I have no idea about the magnitude [01:02:09 CEST] <rcombs> I saw a few hundred in these youtube files [01:02:23 CEST] <rcombs> 1 per second wouldn't be absurd [01:02:41 CEST] <rcombs> so I care enough to point out that it might be worth optimizing, but not enough to actually do it [01:02:43 CEST] <cone-809> ffmpeg 03Vittorio Giovara 07master:5d3addb93794: Add a quality factor packet side data [01:02:44 CEST] <cone-809> ffmpeg 03Michael Niedermayer 07master:e5bae39f46e5: Merge commit '5d3addb937946eca5391e40b5e6308e74ac6f77b' [01:02:45 CEST] <cone-809> ffmpeg 03Michael Niedermayer 07master:809780ca425a: avcodec/utils: Do not merge side data for encoders [01:14:31 CEST] <ZeuZ> Guys, regarding the C API, where can I find an example of something akin to invoking ffmpeg like: -f image2 -framerate 2 -i /CAMARAS/Camara1/camara1.%d.jpg -y /CAMARAS/Camara1/registro.`echo $clips`.avi [01:14:40 CEST] <ZeuZ> I mean, can I input something akin to camara1.%d.jpg as a filter for each image to be taken? Or should I parse them in order and add them? [01:27:55 CEST] <cone-809> ffmpeg 03Vittorio Giovara 07master:40cf1bbacc62: Deprecate avctx.coded_frame [01:27:56 CEST] <cone-809> ffmpeg 03Michael Niedermayer 07master:495eee01238b: Merge commit '40cf1bbacc6220a0aa6bed5c331871d43f9ce370' [01:47:51 CEST] <cone-809> ffmpeg 03Janne Grunau 07master:f56d8d8dd72b: h264: aarch64: intra prediction optimisations [01:47:52 CEST] <cone-809> ffmpeg 03Michael Niedermayer 07master:5d5f8b29b4d4: Merge commit 'f56d8d8dd72b1ab52aa814c5a0fccabf8040ef68' [01:59:44 CEST] <cone-809> ffmpeg 03Janne Grunau 07master:9ed6f9a17cc1: arm: use a local label instead of the function symbol in ff_prefetch_arm [01:59:45 CEST] <cone-809> ffmpeg 03Michael Niedermayer 07master:eca84946eaa1: Merge commit '9ed6f9a17cc1f7d3699a1223783dadc1ee222069' [02:01:43 CEST] <cone-809> ffmpeg 03James Almer 07master:651448a8a7bd: avcodec/aacsbr: fix compilation with hardcoded tables [02:30:46 CEST] <KGB> [13FFV1] 15michaelni pushed 27 new commits to 06master: 02http://git.io/vmxkR [02:30:46 CEST] <KGB> 13FFV1/06master 14bba8cca 15Ashley Blewer: convert ffv1.lyx to markdown [02:30:46 CEST] <KGB> 13FFV1/06master 1420f8948 15ablwr: adjusts tables to comply with github-flavored markdown rendering [02:30:46 CEST] <KGB> 13FFV1/06master 141986fd6 15ablwr: corrects broken tables and moves code block tables to headerless [02:43:45 CEST] <jamrial> either github's markdown parser sucks, or the resulting ffv1.md file isn't good [02:45:31 CEST] <rcombs> or both! [02:45:42 CEST] <KGB> [13FFV1] 15michaelni pushed 2 new commits to 06master: 02http://git.io/vmxtR [02:45:42 CEST] <KGB> 13FFV1/06master 140a8e9ae 15Michael Niedermayer: Add missing '*'... [02:45:42 CEST] <KGB> 13FFV1/06master 1427f6cb5 15Michael Niedermayer: update slice count restriction to what is used in the implementation... [02:45:44 CEST] <rcombs> sturgeon's law says it's both [03:37:03 CEST] <KGB> [13FFV1] 15dericed opened pull request #11: fix markdown table rendering on github (06master...06markdown-fixes) 02http://git.io/vmxch [03:50:12 CEST] <cone-809> ffmpeg 03Tom Butterworth 07master:d8488090e507: libavcodec/hap: Name enums [03:50:14 CEST] <cone-809> ffmpeg 03Tom Butterworth 07master:440c26e9c46b: libavcodec/hap: remove unused struct member [03:50:14 CEST] <cone-809> ffmpeg 03Tom Butterworth 07master:6b96c70f2d89: avcodec/hapdec: don't log texture format every frame, do it once per decode session [03:52:14 CEST] <KGB> [13FFV1] 15michaelni closed pull request #11: fix markdown table rendering on github (06master...06markdown-fixes) 02http://git.io/vmxch [04:45:29 CEST] <KGB> [13FFV1] 15dericed opened pull request #12: Cleaning (06master...06cleaning) 02http://git.io/vmxzN [05:05:34 CEST] <KGB> [13FFV1] 15dericed opened pull request #13: move code from tables to codeblocks (06master...06tables-to-codeblocks) 02http://git.io/vmxwg [06:41:27 CEST] <jamrial> commit bc3f19641ce37d14f9cfd7e2693c4833b63fba7d started the filter-channelmap memleaks [06:41:49 CEST] <jamrial> someone that knows their way around lavfi's filtergraph stuff should check it [06:47:34 CEST] <rcombs> wait, people know their way around lavfi's filtergraph stuff? [06:56:45 CEST] <jamrial> haha [06:58:26 CEST] <durandal_1707> what that commit changed? [06:59:36 CEST] <rcombs> that can be simplified to "people know their way around software?" [07:00:00 CEST] <rcombs> because sturgeon's law applies to both software and developers [07:03:18 CEST] <jamrial> durandal_1707: commit message is "split creating and (re-)configuring complex filtergraphs" [10:15:46 CEST] <__gb__> hi philipl, that's something that should be possible, yes [10:18:12 CEST] <KGB> [13FFV1] 15michaelni closed pull request #12: Cleaning (06master...06cleaning) 02http://git.io/vmxzN [10:41:17 CEST] <cone-626> ffmpeg 03Zhang Rui 07master:cee7acfcfc1b: MAINTAINERS: add myself as a maintainer for async protocol [14:06:47 CEST] <cone-626> ffmpeg 03Rostislav Pehlivanov 07master:331c1e74941d: aacenc: move the generation of ff_aac_pow34sf_tab[] [14:53:15 CEST] <D404|Ghetto> interesting [14:53:18 CEST] <D404|Ghetto> IRT at VDD [15:13:34 CEST] <cone-626> ffmpeg 03hSÇ 07master:21e2f030bfa1: avcodec: loongson move simple idct functions to a separate file [15:13:35 CEST] <cone-626> ffmpeg 03hSÇ 07master:b50910e9a9cf: avcodec: loongson optimize xvid idct with mmi [15:13:36 CEST] <cone-626> ffmpeg 03Tom Butterworth 07master:5d0616ce0e64: avcodec/hapdec: log reason for failure when texture type doesn't match stream [15:17:14 CEST] <ZeuZ> Hello, I'm looking into programatically convert a series of images into an AVI compressed stream. Can somebody help me find the documentation that pertains to that? [15:23:46 CEST] <ZeuZ> Took this as basis: https://ffmpeg.org/doxygen/trunk/api-example_8c-source.html -> yet there are some things I've got doubts with like line 250-255 [15:53:31 CEST] <thardin> carl isn't in here? [15:53:43 CEST] <thardin> anyway, I replied to his patch [16:02:54 CEST] <TimNich> ULs everywhere, ugh. [16:03:15 CEST] <D404|Ghetto> UL? [16:05:12 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02http://git.io/vmjuK [16:05:12 CEST] <KGB> 13FFV1/06master 142217aaf 15Michael Niedermayer: Suggest a pagebreak before Genral description... [16:06:12 CEST] <D404|Ghetto> ..wut? [16:06:24 CEST] <D404|Ghetto> oh. ffv1. [16:17:53 CEST] <cone-626> ffmpeg 03Zhang Rui 07master:aee909acfb34: fate: add test for async protocol [16:17:54 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:593731efa8ee: tests/checkasm/Makefile: Fix checkasm with SDL [17:03:46 CEST] <cone-626> ffmpeg 03Luca Barbato 07master:219b39a71a56: parseutil: Use non ambiguous aliases for uhd [17:03:47 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:be0fd074579c: Merge commit '219b39a71a5694b1c14a07b86477f665a5b6849b' [17:44:40 CEST] <durandal_1707> J_Darnley: how is asm going? [17:49:56 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:f2d7409c7256: avformat/async: Add missing else [17:49:57 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:39867f3e098a: avfilter/af_channelmap: Move ff_add_channel_layout() call to querry_format() [17:57:14 CEST] <cone-626> ffmpeg 03hSÇ 07master:c303ad85a9e4: avcodec: loongson optimize blockdsp with mmi [18:04:25 CEST] <cone-626> ffmpeg 03James Almer 07master:3e46c7dbbeb0: avcodec/options-test: don't alloc avctx->coded_frame [18:04:26 CEST] <cone-626> ffmpeg 03James Almer 07master:d32547a24a3f: avcodec/utils: silence some deprecation warnings [18:34:53 CEST] <durandal_1707> sidechaincompess is too long, awaiting shorter candidates [18:40:02 CEST] <cone-626> ffmpeg 03Bernd Bleßmann 07master:99ad832c7ea9: libavfilter/vf_scale: implement process_command [19:41:18 CEST] <durandal_1707> noone? [19:44:21 CEST] <Compn> filter ? [19:44:26 CEST] <Compn> or api ? [19:44:32 CEST] <Compn> you just want a new name schema? [19:45:43 CEST] <Compn> ZeuZ : https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images is for the user way. you want to make your own program that does it ? confused on what exaclty you want. [19:51:22 CEST] <durandal_1707> Compn: shorter name for filter... [19:53:57 CEST] <Compn> whats it do? [19:54:01 CEST] <Compn> then i can name it :P [19:55:12 CEST] <Compn> +This filter acts like normal compressor but has the ability to filter detected [19:55:12 CEST] <Compn> +signal before sending it to the compression stage. [19:55:14 CEST] <Compn> humm [19:56:28 CEST] <Compn> whats the second signal ? i mean is it audio background hiss noise ? [20:02:10 CEST] <cone-626> ffmpeg 03Bernd Bleßmann 07master:d2b78fe6b71f: libavfilter/vf_crop: implement process_command [20:03:40 CEST] <durandal_1707> Compn: anything what user pick, its meant to provide ducking together with amege and pan filter [20:05:14 CEST] <durandal_1707> For example if you say something on 2nd input output of the filter will lower volume so what you saying can be heard [20:35:54 CEST] <cone-626> ffmpeg 03Luca Barbato 07master:977105407cae: hap: Decode using optimal slices sizes [20:35:55 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:2de5737ee21d: Merge commit '977105407cae55876041dddbf4ce0934cdd4cd6c' [20:37:17 CEST] <Compn> durandal_1707 : i have no idea what to call that then :P [21:09:07 CEST] <cone-626> ffmpeg 03Luca Barbato 07master:6b2b26e7af3e: dds: Decode using optimal slices sizes [21:09:08 CEST] <cone-626> ffmpeg 03Michael Niedermayer 07master:8fe00dd18a4a: Merge commit '6b2b26e7af3ede0abfb46eb5725c26d1083f50bc' [21:28:33 CEST] <cone-626> ffmpeg 03Zhang Rui 07master:5e2098d9064e: avformat/async: fix interrupt_callback usage and return code [21:28:33 CEST] <cone-626> ffmpeg 03Zhang Rui 07master:830d3a0ebe24: avformat/async: rename async_interrupt_callback to async_check_interrupt [23:11:32 CEST] <cone-626> ffmpeg 03Vesselin Bontchev 07master:323ec6b5e214: Add support for TEA (Tiny Encryption Algorithm) [00:00:00 CEST] --- Wed Jul 22 2015

1 0

ffmpeg.log.20150721
by burek 22 Jul '15

22 Jul '15

[00:47:39 CEST] <antiatom> klaxa: Yes figured it out, thank you. [01:11:10 CEST] <ZeuZ> Guys, regarding the C API, where can I find an example of something akin to invoking ffmpeg like: -f image2 -framerate 2 -i /CAMARAS/Camara1/camara1.%d.jpg -y /CAMARAS/Camara1/registro.`echo $clips`.avi ? [01:11:46 CEST] <ZeuZ> I mean, can I input something akin to camara1.%d.jpg as a filter for each image to be taken? Or should I parse them in order and add them? [01:13:51 CEST] <ZeuZ> ups [01:14:02 CEST] <ZeuZ> is this the wrong channel to ask this? [03:05:03 CEST] <Zeranoe> Can FFmpeg work with 'IRD DVB/IP'? [10:37:50 CEST] <AlicanC> I am trying to put an overlay on a video and I am getting a wierd result: [10:37:52 CEST] <AlicanC> -i vid.webm -i overlay.png -filter_complex "[1:v]scale=320:240 [ovrl],[0:v]fps=15,scale=480:-1:flags=lanczos,[ovrl]overlay=W-w-10:H-h-10" -y out.gif [10:39:28 CEST] <AlicanC> It was working fine until i decided to scale the overlay. [10:55:57 CEST] <AlicanC> Not copy-pasting from stackoverflow and actually reading the documentation helps: [10:55:58 CEST] <AlicanC> ffmpeg.exe -i vid.mp4 -i over.png -filter_complex "[1:v] scale=100:-1 [over]; [0:v] fps=15, scale=480:-1:flags=lanczos [main]; [main][over] overlay=W-w-10:H-h-10" -y out.gif [11:18:31 CEST] <Marckov85> hello guys i need help with merging audio and video together [11:18:58 CEST] <Marckov85> am using this command : [11:19:11 CEST] <Marckov85> ffmpeg -i input_vid.mp4 -itsoffset 0.1 -i input_audio.mp3 [11:19:11 CEST] <Marckov85> -vcodec copy -acodec copy output.mp4 [11:19:42 CEST] <Marckov85> anyone can give a hand? [11:23:14 CEST] <Marckov85> olaaaaa [11:27:20 CEST] <AlicanC> ffmpeg -i video.mp4 -i audio.mp3 -c:v copy -c:a libmp3lame out.mp4 [11:27:51 CEST] <AlicanC> Marckov85, I don't know what you are trying to achieve with "-itsoffest 0.1" but this works for me. [11:28:14 CEST] <chungy> you can generally achieve the same thing with just "-c copy" but yea [11:28:26 CEST] <chungy> -itsoffset is supposed to delay the audio by 0.1s I guess. [11:29:40 CEST] <AlicanC> chungy, yeah, this is also fine: ffmpeg -i video.mp4 -i audio.mp3 -c copy out.mp4 [11:33:08 CEST] <Marckov85> mmm [11:33:13 CEST] <Marckov85> i am getting a result [11:33:27 CEST] <Marckov85> but the first time i play the video [11:33:34 CEST] <Marckov85> it says not supported media on android [11:33:40 CEST] <Marckov85> than i play it again it plays [11:33:55 CEST] <Marckov85> i think somehting is going wrong with the first few frames [11:38:36 CEST] <chungy> android's media support is probably extremely picky. I bet if you try it without -itsoffset, it'd work [11:38:53 CEST] <chungy> or just install VLC on your Android device *shrugs* [11:40:33 CEST] <Marckov85> ummm [11:40:40 CEST] <Marckov85> i tried it wihout the offset [11:40:43 CEST] <Marckov85> same problem [11:40:50 CEST] <Marckov85> ffmpeg -i video.mp4 -i audio.mp3 -c copy out.mp4 [11:40:56 CEST] <Marckov85> i tried shis one [11:41:18 CEST] <Marckov85> and without offset the song is faster than the video :( [11:42:29 CEST] <Marckov85> on android am using the videoview player [11:42:31 CEST] <Marckov85> by code [11:42:40 CEST] <Marckov85> the video with no audio works fine [11:42:44 CEST] <Marckov85> when i merge the audio [11:42:55 CEST] <Marckov85> am having this shitty problem [13:02:11 CEST] <Marckov85> weird only not playable on htc [13:02:22 CEST] <Marckov85> on sumsang it is playable well [13:02:30 CEST] <Marckov85> something with the header is wrong [13:11:08 CEST] <goncalomsdias067> I'm trying to extract the audio from one file that has ac3 audio and convert it to pcm_s16le with this command: ffmpeg -i canalplus.ts -map 0:1 -c:a pcm_s16le cplus_converted.ts [13:11:25 CEST] <goncalomsdias067> I got: Error while decoding stream #0:1: Invalid data found when processing input [13:11:48 CEST] <goncalomsdias067> am I doing something wrong? :S [13:45:21 CEST] <ridders24> is it possible to point ffmpeg at a physical disk to carve out frames from h264 streams? [14:04:00 CEST] <AlicanC> Marckov85, you should check those devices' compatibility lists. They might have better support for webm, ogg, ogv, aac, etc. [14:14:25 CEST] <Marckov85> okay thank youu [15:21:48 CEST] <cleanrock> anyone here got hw accelerated HEVC video with GTX 960 on linux ? i do not get it on archlinux with mpv 0.9.2 (ffmpeg 2.7.1) with vo and hwdec set to vdpau [15:40:49 CEST] <ffmpegDEV> hey [16:22:12 CEST] <pingback> Hi i've got an error with ffmpeg ver. 2.7.2 with an m3u8 [16:22:21 CEST] <pingback> Invalid data found when processing input [16:23:04 CEST] <pingback> If I try to play one of the segment in the m3u8 play fine but i need to follow the playlist [16:59:30 CEST] <Jeffries> guys trying to resize a YUV video [16:59:43 CEST] <Jeffries> helpppp [17:26:05 CEST] <Jeffries> (IS ANYONE HERE) [17:26:11 CEST] <Jeffries> urrrrrrrrkh [17:28:35 CEST] <DHE> ... [17:31:30 CEST] <Jeffries> DHE [17:31:36 CEST] <Jeffries> help me resize a YUV video [17:31:38 CEST] <Jeffries> pl0x [17:33:40 CEST] <DHE> I don't know how [17:34:42 CEST] <durandal_1707> Jeffries: use scale filter [17:35:11 CEST] <Jeffries> trying, isn't working [17:35:23 CEST] <Jeffries> ffmpeg isn't very good at handing raw yuv video operations [17:35:46 CEST] <Jeffries> (probably because it just makes calls to codecs for these operations, and yuv has no codec and is just rawvideo) [17:36:00 CEST] <Jeffries> I'll probably encode it to h.264 and resize it there [17:36:04 CEST] <Jeffries> sec [17:39:06 CEST] <Mavrik> what. [17:39:16 CEST] <Mavrik> no :P [17:42:10 CEST] <Jeffries> http://pastie.org/10304469 [17:44:23 CEST] <kepstin-laptop> Jeffries, scale is a filter, not a codec... Use it with the '-f' option. [17:46:06 CEST] <kepstin-laptop> Jeffries, also, for raw video input, you have to specify the frame size (and probably also framerate) as an input option. See https://www.ffmpeg.org/ffmpeg-formats.html#rawvideo [18:17:31 CEST] <Jeffries> ffmpeg -i akiyo.yuv -c:v libx264 -preset veryslow -qp 0 akiyo.mp4 is my command [18:18:55 CEST] <Jeffries> http://pastie.org/10304527 [18:19:13 CEST] <Jeffries> I'm trying to encode a YUV video to h.264, what is wrong here [18:19:29 CEST] <kepstin-laptop> Jeffries, scroll up, i've already told you what's wrong. [18:19:36 CEST] <Mavrik> Indeed. [18:19:48 CEST] <Mavrik> No container means that there's no way of determining basic raw parameters. [18:19:56 CEST] <Mavrik> Probably pix format has to be specified as well [18:20:18 CEST] <kepstin-laptop> pix fmt defaults to yuv420p, which is the most common type. [18:20:28 CEST] <kepstin-laptop> but yeah, might be needed as well :) [18:20:36 CEST] <Jeffries> I'm explicitly stating that format, and nothing [18:20:47 CEST] <Jeffries> ffmpeg -i akiyo.yuv -f 1280:720 -c:v libx264 -preset veryslow -qp 0 -pix_fmt yuv420p akiyo.mp4 [18:21:05 CEST] <Jeffries> really, I just want lossless encoding :O [18:21:07 CEST] <kepstin-laptop> Jeffries, you need to explicitly state the video frame size and framerate as *input* options, following the syntax in the page I linked [18:21:29 CEST] <Jeffries> k [18:21:57 CEST] <kepstin-laptop> (and possibly also pixel format) [18:23:23 CEST] <kepstin-laptop> (in particular, when using the 'ffmpeg' tool, input options must be before the -i filename) [18:23:50 CEST] <Jeffries> yeah, adjusted it [18:23:52 CEST] <Jeffries> ffmpeg -i akiyo.yuv -pixel_format yuv420p -video_size 176x144 -framerate 25 -f 1280:720 -c:v libx264 -preset veryslow -qp 0 -pix_fmt yuv420p akiyo.mp4 [18:23:59 CEST] <Jeffries> this still doesn't seem to work [18:24:16 CEST] <Jeffries> I wonder why this isn't working for me [18:24:34 CEST] <kepstin-laptop> Jeffries, that '-f' option makes no sense, "1280:720" isn't a filter. [18:25:08 CEST] <Jeffries> but you've just told m- [18:25:31 CEST] <Jeffries> alright, level the playing field here, explain what I need to do to scale [18:25:41 CEST] <kepstin-laptop> I said "scale is a filter, so you use it with -f" [18:25:52 CEST] <kepstin-laptop> so the end result would be "-f scale=1280x720" [18:26:24 CEST] <kepstin-laptop> er, wait, I'm wrong about that completely [18:26:27 CEST] <kepstin-laptop> -0f is format :/ [18:26:41 CEST] <Jeffries> kay [18:26:41 CEST] <kepstin-laptop> you have to use -filter instead sorry [18:28:11 CEST] <Jeffries> it still tells me that it can't find the video size [18:28:14 CEST] <kepstin-laptop> so "-filter scale=1280:720" should do it. Although since you're not using any other filters, you can also just do "-s 1280x720" [18:28:22 CEST] <Jeffries> for some reason, though I'm explicitly stating it [18:28:23 CEST] <kepstin-laptop> Jeffries, paste the current full output? [18:28:24 CEST] <Jeffries> ffmpeg -i akiyo.yuv -video_size 176x144 -pixel_format yuv420p -framerate 25 -filter scale=1280:720 -c:v libx264 -preset veryslow -qp 0 -pix_fmt yuv420p akiyo.mp4 [18:28:53 CEST] <kepstin-laptop> Jeffries, ok, you missed another thing I said. "(in particular, when using the 'ffmpeg' tool, input options must be before the -i filename)" [18:29:28 CEST] <Jeffries> http://pastie.org/10304538 [18:29:39 CEST] <Jeffries> uh sorry :P [18:31:42 CEST] <Jeffries> I'd buy you dinner if I knew you in real life [18:31:45 CEST] <Jeffries> Thank you! [00:00:00 CEST] --- Wed Jul 22 2015

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