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May 2016

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ffmpeg-devel.log.20160525
by burek 26 May '16

26 May '16

[01:46:05 CEST] <rrva> in mpeg-ts streams, for a certain GOP, is there anything else than PTS to correlate which audio packets belong to a certain GOP of video frames packets? [01:50:48 CEST] <cone-557> ffmpeg 03Pavel Nikiforov 07master:413c842a6978: avformat/udp: Add a delay between packets for streaming to clients with short buffer [01:50:49 CEST] <cone-557> ffmpeg 03Michael Niedermayer 07master:9b7a8bddac52: avformat/udp: redesign threaded udp tx code [01:50:50 CEST] <cone-557> ffmpeg 03Michael Niedermayer 07master:9591ca704b0e: avformat/udp: Close the socket after destroying the thread using the socket [02:28:49 CEST] Action: llogan thanks claws-mail for re-downloading 41.5k emails for no reason [03:42:26 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vr6gl [03:42:26 CEST] <KGB> 13FFV1/06master 142a4c5ba 15Jérôme Martinez: Multi-threading support and independence of slices... [04:13:25 CEST] <Compn> llogaaan [04:13:27 CEST] <Compn> huh [04:14:06 CEST] Action: Compn will never understand why croatia has .hr tld [04:15:19 CEST] <durandal_1707> something happened? [04:31:16 CEST] <bofh_> Compn: (Republika) Hrvatska? I guess either they explicitly wanted native name, or the usual ones were already taken. [04:54:23 CEST] <Compn> durandal_170 : i was going to ask him why he keeps 40,000 mails on his server for his client to redownload them :D [04:54:34 CEST] <Compn> [20:28] * llogan thanks claws-mail for re-downloading 41.5k emails for no reason [04:54:55 CEST] <Compn> also what email server allows that kind of storage [04:55:08 CEST] <Compn> i think my isp filled up after 300 last time i was on vacation :D [05:20:55 CEST] <jamrial> Compn: gmail [12:36:45 CEST] <BtbN> andrey_turkin, when merging to libav, don't get too close on how they handle options. [12:37:16 CEST] <BtbN> I'm not a fan of that, I'd very much prefer getting closer to how libx264 works, specially for crf. [12:37:49 CEST] <BtbN> It's entirely non-obvious that one has to use -global_quality to trigger that mode at the moment. [12:40:20 CEST] <andrey_turkin> oh, why? It seemed pretty nice to me [12:40:33 CEST] <andrey_turkin> maybe ratecontrol stuff is a bit weird [12:40:55 CEST] <BtbN> Primarily because it breaks compatiblity. [12:41:05 CEST] <andrey_turkin> ah, that ) [12:41:13 CEST] <andrey_turkin> yeah, totally would break [12:41:35 CEST] <BtbN> People are used to to libx264, and staying close to that was my original idea. [12:42:06 CEST] <andrey_turkin> I'll have to rework those patches then later [12:42:46 CEST] <BtbN> I'm not entirely sure anymore how it ended up using avctx->global_quality for crf/constqp mode, but a crf parameter that matches the libx264 one might be something to add later. [12:42:50 CEST] <andrey_turkin> though I'm not a big fan of how it used to be. Too fragile and interconnected [12:43:15 CEST] <andrey_turkin> actually crf/constqp code seemed similar in both libav and ffmpeg [12:43:24 CEST] <andrey_turkin> vbr is different [12:43:37 CEST] <BtbN> yes, except that you have to specify -rc constqp for it [12:43:42 CEST] <BtbN> instead of just -global_quality [12:44:04 CEST] <andrey_turkin> I think you don't need to specify rc, it should do the right thing [12:44:15 CEST] <andrey_turkin> in both libav and ffmpeg code as is [12:44:53 CEST] <BtbN> ah, they have that only to override it, if the automagic decision which mode to use fails. [12:45:03 CEST] <andrey_turkin> yeah [12:45:23 CEST] <BtbN> That doesn't seem too bad. If it's possible to add it in a way that doesn't break existing commandlines, it would be nice to have. [12:45:57 CEST] <andrey_turkin> actually no. If you specify -rc then it is forced to use that. If you don't specify it then it does autodecision [12:46:45 CEST] <andrey_turkin> I can try to rework those patches to have both -rc and ffmpeg's cbr/twopass options [12:48:38 CEST] <BtbN> Didn't find any problems with your first chunk of patches, as far as I can test on my Kepler hardware. Did some minor formatting fixes to make patcheck happy, otherwise unmodified. [12:49:01 CEST] <andrey_turkin> I just pushed the rest. [12:49:04 CEST] <BtbN> Do you want to change something, or can I push them? [12:49:15 CEST] <andrey_turkin> Let me see one more time [12:49:44 CEST] <BtbN> https://github.com/BtbN/FFmpeg/commits/master [12:50:59 CEST] <andrey_turkin> ah, those 4. I am happy with thouse [12:51:35 CEST] <BtbN> And one small change on top, rebasing on it should be trivial. [12:51:46 CEST] <andrey_turkin> I actually have some doubts about nvenc registration cache but it is from libav code [12:52:38 CEST] <BtbN> I'm not so sure anymore if a 100% merge is desirable. Just focus on the stuff that seems like a good idea to have for now, don't aim to have the exact same code in the end. [12:53:15 CEST] <cone-803> ffmpeg 03Andrey Turkin 07master:82d705e24505: avcodec/nvenc: split large functions into smaller ones [12:53:16 CEST] <cone-803> ffmpeg 03Andrey Turkin 07master:e1691c44f045: avcodec/nvenc: combine input and output surface structures [12:53:17 CEST] <cone-803> ffmpeg 03Andrey Turkin 07master:cfb49fc6f797: avcodec/nvenc: replace custom FIFOs with AVFifos [12:53:18 CEST] <cone-803> ffmpeg 03Andrey Turkin 07master:a8cf25dd9233: avcodec/nvenc: CUDA frames support [12:53:19 CEST] <cone-803> ffmpeg 03Timo Rothenpieler 07master:24fcb2335184: avcodec/nvenc: Require Maxwell for lossless [12:57:28 CEST] <andrey_turkin> I was aiming for 100% merge when doing the rest of the patches (as obvious from first "rename" commit). I can go throuth the patches again and probably can rework most of them so they make code nicer (or bring code closer to libav) while not breaking compatibility [12:58:39 CEST] <andrey_turkin> anyway, you can fetch my branch and compare it to libav; rest of the stuff there are changes I'm not sure about how to handle. Some of that difference should probably go to libav [14:18:27 CEST] <BtbN> I really need to setup some automated tests for nvenc. [14:18:39 CEST] <BtbN> I guess fate tests are just fine, they only need a special machine to run on. [14:26:49 CEST] <nevcairiel> unfortunately testing something like this in fate is .. problematic, because you cant really test the precondition of hardware existing [14:31:08 CEST] <andrey_turkin> funny fact - apparently swscale nv12-nv12 without any actual scaling is broken, and apparently have been for some time [14:31:22 CEST] <BtbN> shouldn't it just.. do nothing? [14:31:26 CEST] <andrey_turkin> it should [14:31:46 CEST] <andrey_turkin> or rather it should copy data around [14:31:48 CEST] <nevcairiel> now that you mention it, i noticed that a long time ago as well [14:31:54 CEST] <nevcairiel> never really bothered to test again tho [14:32:13 CEST] <BtbN> I really need to get myself new hardware... [14:32:19 CEST] <BtbN> not being able to test most of this is annoying [14:32:48 CEST] <andrey_turkin> instead it copies Y plane and fills UV plane with zeros. Something to do with wrongly assuming it is a GRAY8 or something [14:32:49 CEST] <nevcairiel> buy pascal, so you can get hevc10 encoding going =p [14:32:49 CEST] <BtbN> Maybe I'll get my Christmas present "slightly" early this year. [14:32:59 CEST] <BtbN> Yes, I will get myself a 1080 [14:33:11 CEST] <BtbN> But for Christmas, which is still half a year [14:33:47 CEST] <BtbN> I'm also still hoping for Intel to get Kaby Lake out, so I can add VAAPI hevc10 stuff as well. [14:34:18 CEST] <andrey_turkin> I got me GTX 960 last Christmas and I was happy; latest hardware and stuff. And then they went ahead and presented entire new generation ( [14:34:41 CEST] <BtbN> at least you got the 2nd gen Maxwell [14:34:49 CEST] <andrey_turkin> yeah, that's good [14:34:50 CEST] <BtbN> 960 is better than 970/980 [14:34:54 CEST] <BtbN> for video stuff [14:35:01 CEST] <andrey_turkin> it was like 2 times cheaper too [14:35:16 CEST] <BtbN> kinda silly how the 970/980 are able to hw-encode HEVC, but can't decode it. [16:04:19 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vri9f [16:04:19 CEST] <KGB> 13FFV1/06master 149c92e4b 15Jérôme Martinez: Byte alignment is made explicit... [16:23:00 CEST] <philipl> BtbN: Are you looking at those nvenc cuda patches? [16:23:21 CEST] <philipl> I see they've just been pushed :-) [16:33:13 CEST] <cone-803> ffmpeg 03foo86 07master:72a9a7e9cbc2: avcodec/dca_exss: prepare to be called from parser context [16:33:14 CEST] <cone-803> ffmpeg 03foo86 07master:cb55c5b49e6d: avcodec/dca_parser: set duration for core-less streams [16:33:15 CEST] <cone-803> ffmpeg 03James Almer 07master:493db04c4e49: avcodec/Makefile: add missing dependencies to dca parser [16:36:29 CEST] <kierank> {"txt_duration", "display duration of teletext pages in msecs", OFFSET(sub_duration), AV_OPT_TYPE_INT, {.i64 = 30000}, 0, 86400000, SD}, [16:36:30 CEST] <kierank> ??? [16:36:38 CEST] <kierank> why 30 seconds [16:41:14 CEST] <andrey_turkin> somebody probably decided under normal conditions each page would be retransmitted somehow more frequenct than 30 seconds [16:43:15 CEST] <kierank> ok [16:44:17 CEST] <andrey_turkin> which is a reasonable default for subtitles (you don't want subtitle to be on the screen forever if commands stop coming in) [16:49:11 CEST] <cone-803> ffmpeg 03Matthieu Bouron 07master:fbc9359d859f: lavc/mediacodec: factorize static fields initialization [16:49:12 CEST] <cone-803> ffmpeg 03Matthieu Bouron 07master:d14deeb6bcf4: lavc/mediacodec: add missing MediaCodec.Get{Input,Output}Buffer() checks [16:53:56 CEST] <JEEB> hmm, I wonder if this improves some stuff [16:53:59 CEST] <JEEB> for fallbacks [17:04:50 CEST] <mateo`> JEEB: I don't think it will improve things in this matter [17:05:12 CEST] <JEEB> ok [17:05:25 CEST] <JEEB> so I guess the best way would be to create profile checks in the AVC decoder [17:06:44 CEST] <mateo`> I'll send a patch later that removes the arbitrary resolution limit of the decoders [17:07:40 CEST] <mateo`> As it's totally arbitrary and hardcoded somewhere on the system [17:08:31 CEST] <mateo`> I should also try to export which profiles an encoder support (this might help regarding fallbacks) [17:09:15 CEST] <BtbN> philipl, there's more work on the way, essentialy a merge of the libav and ffmpeg nvenc implementations. [17:10:06 CEST] <mateo`> I will have some time to work it next week (and port the decoder to the new bsf api, update the hwaccel patch, ...) [17:12:22 CEST] <BtbN> philipl, and I'm also planning to add a cuvid hwaccel, but that might take a while [17:13:27 CEST] <andrey_turkin> BtbN: what will it give in terms of gains? [17:13:45 CEST] <BtbN> full hw transcode on nvidia hardware, without the frames ever leaving the vram [17:15:06 CEST] <BtbN> and also accelerated video decoding on nvidia hardware without a running X server. [17:15:15 CEST] <BtbN> As vdpau needs X [17:16:49 CEST] <andrey_turkin> right. I wondered a bit if it may be more lucrative to take a stab at d3d11va/dxva/vdpau->cuda interop; but in the end it probably would mean more work than one more decoder [17:17:02 CEST] <BtbN> The stupid thing is, cuvid only comes with a h264 NAL parser on windows. [17:17:09 CEST] <BtbN> So it can't be a simple decoder. [17:17:11 CEST] <andrey_turkin> and it won't help with headless linux [17:17:20 CEST] <BtbN> It has to be another hwaccel like vdpau and vaapi are. [17:17:27 CEST] <BtbN> as ffmpeg has to do the h264 parsing [17:21:43 CEST] <cone-803> ffmpeg 03Muhammad Faiz 07master:defab0825f41: avfilter/src_movie: add various commands [17:27:37 CEST] <andrey_turkin> BtbN: Here are some reworked patches which should be ready for inclusion in ffmpeg: https://github.com/tea/FFmpeg/tree/nvenc . [17:31:04 CEST] <philipl> BtbN: Jolly good. In that context, it's probably worth revisting the patches from Agatha eventually, as you'll have the infrastructure to do it properly [17:31:35 CEST] <BtbN> i think those will be obsolete then, as these features are already present. [17:35:12 CEST] <philipl> The cuda scaling is? [17:35:32 CEST] <BtbN> there are patches for libav somewhere that add those filters [17:35:39 CEST] <BtbN> and I think they are planned to be merged [17:36:00 CEST] <philipl> cool beans. [17:38:35 CEST] <andrey_turkin> isn't scale_npp already merged? [19:23:09 CEST] <kierank> BBB: http://www.streamingmedia.com/Articles/ReadArticle.aspx?ArticleID=111296 [19:24:37 CEST] <JEEB> yeah, it's not surprising that companies are looking into VP9 given the HEVC landscape being so weird [19:27:19 CEST] <BBB> I should go and be good friends with many of these people ;) [19:28:28 CEST] <jamrial> seeing how they are complaining about encoding speeds, you should :p [19:28:31 CEST] <BBB> is brightcove what used to be known as zencoder? [19:29:22 CEST] <kierank> brightcove bought zencoder [19:30:31 CEST] <BBB> aha I see [19:33:58 CEST] <jamrial> what's the deal with this AV1? and will it replace VP10 or live alongside it as a derived format like sorenson 3 seems to be for h264? [19:34:23 CEST] <RiCON> av1 is a mix of thor + vp10 + daala [19:34:52 CEST] <jamrial> so it will live alongside all those instead? [19:34:59 CEST] <nevcairiel> perhaps [19:35:01 CEST] Action: jamrial is reminded of a certain xkcd strip [19:35:10 CEST] <nevcairiel> the dalaa people said they will use dalaa more as a testbed for now [19:35:27 CEST] <nevcairiel> not sure how vp10 people see that [19:48:10 CEST] <rcombs> call me back when something has an encoder that actually competes with x264 [19:48:27 CEST] <rcombs> or when hardware encoders aren't bad [19:49:26 CEST] <BBB> I think we had this discussion on #x264dev a while ago [19:49:47 CEST] <BBB> on the high end, which is where netflix arguably lives (computing time is essentially free), hevc/vp9 clearly beat h264 [19:49:56 CEST] <BBB> especially at high resolutions [19:50:11 CEST] <JEEB> the formats beating AVC isn't a surprise to anyone, really [19:50:19 CEST] <rcombs> well sure [19:50:24 CEST] <BBB> x264 competes well on the end where computing time is expensive [19:50:24 CEST] <JEEB> the psychovisual optimizations for libvpx on the other hand... [19:50:30 CEST] <rcombs> ^ [19:50:51 CEST] <rcombs> and on psy [19:50:55 CEST] <JEEB> basically, I'm OK with 0.1fps if the result is good and the rate control isn't bad [19:50:59 CEST] <JEEB> which I think BBB's encoder does [19:51:01 CEST] <BBB> libvpx != vp9, as awesome developers of competing, commercial vp9 encoders would no-doubt say [19:51:21 CEST] <JEEB> yes, which is why I specified that the formats beating AVC isn't a surprise [19:51:33 CEST] <JEEB> the thing that people have currently for VP9 is libvpx though, which is what it is [19:53:24 CEST] <jamrial> BBB: so i take your encoder, much like your decoder, is also replicating libvpx bugs that sneaked into the bitstream, right? [19:53:39 CEST] <BBB> hm & well ... [19:53:40 CEST] <BBB> so ... [19:54:14 CEST] <BBB> most of these bugs are in obscure corners of the bitstream that were either sneaked in to make third parties happy, or as palceholders for unfinished features [19:54:35 CEST] <BBB> if they were sneaked in to make third parties happy, like the whole scalable mess, I just dont support it, because nobody cares [19:54:51 CEST] <BBB> if theres bugs in unfinished placeholders for features, then & well & its hard to say what I would do [19:55:05 CEST] <jamrial> i see [19:55:06 CEST] <BBB> I mean, I could reproduce the bug, but likely that makes the placeholder useless [19:55:19 CEST] <BBB> theres a few bugs that are in the base bitstream that I reproduce, yes [19:55:33 CEST] <BBB> I mean, I dont have a choice, otherwise the files would not decode [19:55:44 CEST] <BBB> but mostly these are innocent and have no coding impact, theyre just silly [19:58:58 CEST] <Compn> werent we skipping vp* and going for hevc? :P [19:59:00 CEST] Action: Compn ducks [20:00:50 CEST] <andrey_turkin> apparently hevc is old news now ) [20:04:17 CEST] <jamrial> Compn: patents happened, it seems. but hevc isn't dead, knowing it's used for broadcasting and uhd blu ray [20:06:44 CEST] <andrey_turkin> people are conservative. I've seen something that wasn't MPEG4 or AVC (with 4:2:0 JPEG pixelformat) 2 or 3 times in the last two years [20:07:57 CEST] <andrey_turkin> I mean MPEG colorspace. And one of those 2 or 3 times was same old AVC just with JPEG colorspace [20:10:30 CEST] <Compn> andrey_turkin : you havent seen webm ? [20:10:46 CEST] <BBB> some people just dont watch youtube :-p [20:10:50 CEST] <Compn> i guess :P [20:10:51 CEST] <BBB> or they watch it on internet explorer [20:10:54 CEST] <Compn> hue hue hue [20:11:10 CEST] <andrey_turkin> well I mostly work with broadcasting but OTT translations I've seen were in AVC as well [20:11:28 CEST] <andrey_turkin> I guess in VoD domain things might be different [20:12:46 CEST] <andrey_turkin> but then there is youtube and...? any other major providere does vp8/vp9? [20:13:09 CEST] <BBB> that article mentions a few [20:13:21 CEST] <BBB> kierank: wasnt there a streamingmedia.com conference or so? [20:13:24 CEST] <BBB> what was that again [20:13:33 CEST] <kierank> there are loads of those [20:13:38 CEST] <kierank> not very technical imo [20:14:00 CEST] <BBB> any useful ones? [20:15:22 CEST] <kierank> nothing on par with vdd [20:15:43 CEST] <kierank> but you should probably go to streamingmedia blah blah [20:16:34 CEST] <BBB> well, look, I should go to both [20:16:56 CEST] <BBB> the problem (between quotes for a good reason) with vdd is that its not very suitable for corporate software like my encoder (and it shouldnt be) [20:17:09 CEST] <BBB> I think its good to keep vdd intentionally focussed on opensource [20:17:48 CEST] <kierank> I know, i'm just saying there aren't conferences at the same technical level as vdd [20:18:01 CEST] <kierank> there are in other related fields (siggraph etc) [20:18:12 CEST] <kierank> but multimedia seems to have a lot of blah blah and not much technical stuff [20:18:59 CEST] <BBB> its easy to blah blah with a madonna video in the background [20:19:09 CEST] <BBB> nobody pays attention to the blah blah because oooo soooo awesooooome [20:22:27 CEST] <BBB> I did get a comment once, that someone liked my marketing material because it didnt contain the characters 50% [20:33:41 CEST] <JEEB> BBB: yeah... those numbers come up so many goddamn times I'm surprised more people haven't noticed that it's a theoretical figure [20:46:34 CEST] <rcombs> so apparently some versions of mingw-w64 (no clue which) open files in mkstemp() with `_O_TEMPORARY` [20:47:03 CEST] <rcombs> ar proceeds to close that fd and open a new one at the same path [20:47:11 CEST] <rcombs> lovely race condition there [20:47:14 CEST] <JEEB> oh yeah [20:47:17 CEST] <JEEB> I think I know that one :P [20:48:06 CEST] <rcombs> got a sane solution? [20:48:37 CEST] <rcombs> I was planning to (make somebody else) rebuild binutils with ar patched to reopen the file without `_O_TEMPORARY` and then close the original fd [20:48:40 CEST] <JEEB> not really [20:48:58 CEST] <rcombs> q why is mingw so bad [20:51:40 CEST] <BBB> oh god I just saw a youtube video and it looked so grossly poorly compressed I wanted to kill myself [20:51:45 CEST] <BBB> I right-clicked, and it was avc1 :-p [20:52:00 CEST] <JEEB> well youtube eff's everything [20:52:01 CEST] <rcombs> at least it's not random webms on sankaku or 4chan [20:52:03 CEST] <JEEB> be it AVC or VP9 [20:52:57 CEST] <iive> rcombs: i don't understand, why both temp files race condition. isn't mkstemp supposed to generate different names for the open files? [20:53:43 CEST] <rcombs> iive: right, but ar closes the fd it gets from mkstemp and reopens at the same path [20:53:59 CEST] <rcombs> so since mkstemp opens with _O_TEMPORARY, the file is deleted on close [20:54:04 CEST] <rcombs> (WTF?) [20:54:28 CEST] <rcombs> so if another mkstemp happens between that close and the subsequent open, you're fucked [20:54:52 CEST] <rcombs> the worst part being that both processes think everything's fine until they try to rename later [20:55:15 CEST] <rcombs> also apparently ar doesn't seed rand() with anything useful [20:57:24 CEST] <JEEB> :D [20:57:39 CEST] <JEEB> "waiter, where is my whiskey!?" [21:01:04 CEST] <iive> so, there are actually 2 separate problems. using _O_TEMPORARY that deletes file at close and having bad rand that generates same pattern. [21:02:43 CEST] <durandal_1707> libavar [21:09:19 CEST] <rcombs> iive: well rand() is supposed to generate the same pattern for the same seed [21:09:35 CEST] <rcombs> and ar apparently doesn't seed it at all, so I'd imagine there's some fixed default (0?) [21:09:56 CEST] <iive> that's patently bad idea. [21:10:24 CEST] <rcombs> not seeding? [21:11:15 CEST] <rcombs> the docs don't specify what RNG function mkstemp() uses, so the application developer can't know what they're supposed to seed [21:11:26 CEST] <iive> "If no seed value is provided, the rand() function is automatically seeded with a value of 1." [21:11:49 CEST] <rcombs> but it shouldn't matter since mkstemp() retries until it hits a file that doesn't already exist [21:12:14 CEST] <rcombs> (or until the int storing the number of tries overflows, but that's not especially likely) [21:12:46 CEST] <iive> well, if int overflows.. you might not have space in the directory for more files :D [21:13:41 CEST] <rcombs> yeah [21:13:53 CEST] <rcombs> and then whether or not rand() is seeded isn't really relevant [21:19:25 CEST] <iive> it's quite relevant, if you run two ar in parallel... [21:19:54 CEST] <Plorkyeran> that should only make it slower [21:20:05 CEST] <Plorkyeran> you have to handle collisions regardless of whether or not you properly seed it [21:24:28 CEST] <iive> if we assume that ar are running in perfect sync, then both mkstemp would make the check for existence at the same time and then attempt to create it at the same time. [21:25:02 CEST] <iive> unless windows have flag that makes file creating fail if the file already exists [21:32:20 CEST] <rcombs> iive: O_CREAT + O_EXCL does that [21:33:39 CEST] <iive> mkstemp does mention that it does use the exclusive flag. [22:30:31 CEST] <andrey_turkin> BtbN: where should I send a CUDA-related patch if it is meant for both libav and ffmpeg? just libav-devel, just ffmpeg-devel, both? [22:31:00 CEST] <BtbN> I'd say both. [22:31:10 CEST] <andrey_turkin> ok [22:31:29 CEST] <BtbN> The CUDA stuff in ffmpeg doesn't realy have a maintainer though, as it was just merged from libav. [22:31:43 CEST] <BtbN> I'll try to have a look at it [22:31:59 CEST] <Compn> make sure you mention that it was sent to both [22:32:09 CEST] <andrey_turkin> ah, ok [22:32:11 CEST] <Compn> sometimes devs get weird ideas ... and fail to cooperate [22:32:27 CEST] <Compn> then code differs between projects for idiot reasons [22:32:42 CEST] <Compn> then we have two decoders for the same codec hah! [00:00:00 CEST] --- Thu May 26 2016

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ffmpeg.log.20160525
by burek 26 May '16

26 May '16

[01:39:55 CEST] <Aksen> Hi all, I'm trying to merge 8 mono wav files into a single 5.1+LtRt interleaved wav (SMPTE 302M format). I'm getting errors but when I google I can't find any examples of others using this encoder, so I'm not sure what my next step is. http://pastebin.com/exXQbhge [01:43:46 CEST] <rrva> in mpeg-ts streams, for a certain GOP, is there anything else than PTS to correlate which audio packets belong to a certain GOP of video frames (packets)? [02:05:46 CEST] <JerryT> Any suggestions on how to go about getting 5 frames from a video starting after the first minute of footage and then every 1/5th of the movie. So say the movie is 5 minutes it would get 5 frames from the first minute, 2nd, 3rd, 4th, 5th minute. Then put those images into 1 long image sprite horizontally. And ideally have the images be high quality frames? [02:26:08 CEST] <dsl42> JerryT, https://trac.ffmpeg.org/wiki/Create%20a%20thumbnail%20image%20every%20X%20s… should be the answer to the first part of your questions. no clue how to put them into one image with ffmpeg afterwards, i probably would just use gimp. would not be surprised if it would be possible with ffmpeg though [02:27:30 CEST] <dsl42> ok.. [02:32:18 CEST] <llogan> Aksen: "s302m codec not supported in WAVE format" [02:35:13 CEST] <llogan> try another output format. such as mpegts (.ts,.m2t,.m2ts,.mts, etc) [02:35:33 CEST] <Aksen> thanks, i'll give it a shot [02:36:38 CEST] <Aksen> the ts extension worked. I'll check to see if it's workable for my client [03:31:20 CEST] <okanaganrusty> i'm playing with ffmpeg trying to overlay two videos and an audio stream but keep getting an error - Error initializing complex filters, can someone please let me know what i'm doing wrong, or point me in the direction to solve the filter issue i have (http://pastebin.com/Lqk7GqKs) [03:34:51 CEST] <llogan> okanaganrusty: you're missing the complete output [03:35:50 CEST] <okanaganrusty> the output is at http://pastebin.com/5Ht36HZH [03:47:18 CEST] <llogan> okanaganrusty: didn't get to look closely, but use vstack instead of nullsrc+overlay. it's easier and faster [03:47:26 CEST] <llogan> i mean hstack [03:47:35 CEST] <okanaganrusty> okay, i'll look at that. thanks [03:47:48 CEST] <llogan> and you can use anullsrc or aevalsrc instead of /dev/zero if you prefer [03:48:12 CEST] <llogan> got to go [04:46:29 CEST] <asdfffdsa> Using ffmpeg to reduce the size of a video file. I got about 90% done but then I ran out of space on the HD, causing ffmpeg to stop (process still running and taking up CPU, but the output file size remains the same so i'm not sure what it's doing). Is there any way I can make ffmpeg "resume" without killing and restarting the command? [04:46:45 CEST] <asdfffdsa> (I've sinced cleared up some more HD space) [04:51:41 CEST] <asdfffdsa> well i have to restart but i will be back later [04:58:31 CEST] <CoJaBo> hate it when that happens lol [05:27:41 CEST] <sheenobu> I'm able to use ffmpeg to take webcam input and x11grab input, overlay them (yay!), and output to a dummy v4l2 device. The thing I want to do, after that, is be able to turn off the inputs and keep the dummy v4l2 device up. [05:27:59 CEST] <sheenobu> gstreamer might an option [05:29:49 CEST] <sheenobu> as is some sort of RTMP aware thing. (webcam, x11grab) -> smart rmtp server that can detect and switch inputs -> rtmp client -> v4l2 dummy device [07:04:23 CEST] <kristofferR> I'm trying to record a web stream, but sometimes it's unstable so ffmpeg thinks the streams end prematurely. How can I make ffmpeg wait longer for incoming packets? [07:58:06 CEST] <arbi> Hi! Can anyone recommend a good video composer for linux that can handle mp4 files? (Cinelerra has problems with mp4) [08:04:40 CEST] <Betablocker> https://alternativeto.net/software/cinelerra/ [08:05:00 CEST] <Betablocker> try windows movie maker [08:05:06 CEST] <Betablocker> kidding&. [08:06:15 CEST] <arbi> :) [08:26:42 CEST] <Me4502> Hi, I'm trying to use ffmpeg to convert a single PNG into an MP4 of a length of 1 second [08:27:00 CEST] <Me4502> This is what I'm using now - ffmpeg -f image2 -framerate 1 -i input.png output.mp4 [08:27:19 CEST] <Me4502> However, the video doesn't play. In details it has the correct dimensions and length however [08:27:38 CEST] <Me4502> If I press play in VLC - the screen flickers to the dimensions and goes back - no image appears [08:34:11 CEST] <Me4502> Here's the video file if anyone wants it http://s000.tinyupload.com/index.php?file_id=75587366096982834330 [08:34:19 CEST] <Me4502> I re-did it with 10 seconds, just incase 1 was too short [08:42:12 CEST] <thebombzen> Me4502: VLC automatically goes black when the video ends [08:42:25 CEST] <Me4502> I also tried with the windows 10 photos/videos app [08:42:28 CEST] <thebombzen> try playing it with ffplay. which freezes when the video ends. [08:42:47 CEST] <thebombzen> also try setting the video codec. to something like mpeg4 [08:43:10 CEST] <Me4502> Is codec between input and output? [08:43:40 CEST] <Me4502> whelp.. Setting codec fixed it [08:43:41 CEST] <Me4502> Thanks [08:44:41 CEST] <thebombzen> if you don't set the codec it will use libx264 profile High 4:4:4 Predictive [08:44:49 CEST] <thebombzen> which windows probably can't read cause it's dumb [08:44:58 CEST] <Me4502> Oh, I presumed it'd be able to [08:45:01 CEST] <Me4502> Makes sense [08:45:22 CEST] <thebombzen> vlc should display the video for one second, but it's probably programmed to be "done" when the last frame plays [08:45:34 CEST] <Me4502> Possibly [08:45:38 CEST] <thebombzen> and when it's done it goes to the black screen. [08:45:39 CEST] <Me4502> And seeing as there is only one frame [08:46:11 CEST] <Me4502> Actually yeah - VLC works if I have 2 frames, doesn't if I have 1 [08:46:31 CEST] <thebombzen> there you go [08:46:51 CEST] <Me4502> Another quick question - https://trac.ffmpeg.org/wiki/Concatenate#demuxer [08:47:02 CEST] <Me4502> Is that mylist.txt absolute paths only? [08:47:05 CEST] <Me4502> Or can I use relative? [08:52:05 CEST] <klaxa> as you can tell from the examples it's relative [08:52:20 CEST] <Me4502> Okay - thanks [09:23:26 CEST] <arbi> quit [09:54:49 CEST] <prelude2004c> hey everyone, good day [09:55:10 CEST] <prelude2004c> question.. if i have ffmpeg -f output .. how do i pipe this to send the hls files to a webdav server instead of a local output [09:55:28 CEST] <prelude2004c> can't seem to find reliable info online [10:14:20 CEST] <sheenobu> prelude2004c, i wouldn't be surprised if ffmpeg couldn't output webdav. are you on a *nix? [10:14:32 CEST] <sheenobu> rather..output /to/ webdav [10:14:35 CEST] <prelude2004c> yup nix [10:14:52 CEST] <sheenobu> then your best bet may be to mount the webdav mount and just output to a file on the mount [10:15:44 CEST] <sheenobu> prelude2004c, actually https://www.ffmpeg.org/ffmpeg-protocols.html#http [10:18:32 CEST] <prelude2004c> i have a remote mount setup but i am having issues where sometimes if the remote mount lags it messes up the output [10:18:53 CEST] <prelude2004c> so i am trying to just push it to an http:// where it can retransmit and keep a queue [10:21:03 CEST] <BtbN> Run ffmpeg on the target machine instead. [10:21:03 CEST] <sheenobu> ahh [10:22:20 CEST] <sheenobu> i can actually use webdav as an output but i'm not seeing the output [10:23:07 CEST] <sheenobu> you can even use pipe as an output to wget -X POST prelude2004c ? [10:23:40 CEST] <prelude2004c> yup.. just have to learn how to do it [10:23:52 CEST] <sheenobu> figuring it myself [10:23:53 CEST] <BtbN> Why not just run nginx-rtmp on the target machine? [10:23:56 CEST] <prelude2004c> will it execute the segment_delete as well? [10:28:23 CEST] <prelude2004c> maybe i run inotify wait.. and then use curl to grab data and push to webdav ? [10:28:26 CEST] <prelude2004c> would that be easier ? [10:30:24 CEST] <prelude2004c> maybe i am over thinking it [10:31:26 CEST] <BtbN> nginx-rtmp can output hls and dash from an rtmp input, or run ffmpeg to do the conversion. [10:31:39 CEST] <BtbN> No need for strange "curl webdav streaming" hacks. [10:45:31 CEST] <prelude2004c> this is odd. [10:45:41 CEST] <prelude2004c> 016/05/25 04:44:41 [error] 18637#0: *28484444510 open() "/var/www/citshd/citshd2M69.ts" failed (2: No such file or directory), client: 10.0.200.219, server: , request: "POST /webdav/citshd/citshd2M69.ts HTTP/1.1", host: "10.0.200.2" [10:45:53 CEST] <sheenobu> you'll want a PUT [10:46:10 CEST] <prelude2004c> so i pushed http:// and webdav is running on nginx.. ( actually pushing data from other location ) .. odd how its erroring out [10:46:22 CEST] <prelude2004c> you mean request PUT instead of POST ? [10:46:26 CEST] <sheenobu> yes [10:46:30 CEST] <sheenobu> PUT creates files in webdav [10:46:32 CEST] <prelude2004c> how do i change that ? [10:46:37 CEST] <prelude2004c> ffmpeg have an option ? [10:46:56 CEST] <prelude2004c> i am simply outputing to http://server/webdav/$files [10:47:22 CEST] <sheenobu> ahh [10:48:27 CEST] <sheenobu> http://ffmpeg.org/ffmpeg-all.html#http [10:48:29 CEST] <sheenobu> there is a 'method' option [10:48:40 CEST] <prelude2004c> yup reading ont hat.. but how to use it :) [10:48:44 CEST] <prelude2004c> on the output i guess [10:49:27 CEST] <sheenobu> -a copy -f type -method PUT http://... [10:49:37 CEST] <prelude2004c> checking [10:50:02 CEST] <Elirips> Hello. I'm using ffmpeg to have fun with some rtsp-streams. Question: When will ffmpeg disconnect from an rtsp stream if no data is received? never? 60 seconds? 10 seconds? is there any param to fine-tune when ffmepg will disconnect? I know about -stimeout - is there anyhing else? [10:51:21 CEST] <prelude2004c> wow, look at that :) [11:42:24 CEST] <sheenobu> prelude2004c, working? [11:42:38 CEST] <prelude2004c> yup.. [11:42:39 CEST] <prelude2004c> thank you [11:42:44 CEST] <sheenobu> great, i'm glad i could help [11:43:01 CEST] <prelude2004c> thank you very much.. just tweaking some scripts for automation.. but looks good [11:43:57 CEST] <sheenobu> great [12:04:22 CEST] <sheenobu> hrm. this is a problem if your WebDAV server is under a non-valid TLS cert [12:36:40 CEST] <wallbroken> ffmpeg -i input -vf "hflip,vflip,scale=1024:-2" [12:36:44 CEST] <wallbroken> ffmpeg -i input -vf "transpose=1,scale=576:-2" [12:36:51 CEST] <wallbroken> is there some difference between those commands? [12:41:36 CEST] <c_14> wallbroken: yes [12:41:55 CEST] <wallbroken> the first rotates of 90 degrees [12:41:58 CEST] <wallbroken> the seconds of 180 [12:42:07 CEST] <c_14> yes [12:42:11 CEST] <c_14> Also, the scales are different [12:42:12 CEST] <wallbroken> but whi the first uses hflip, vplip and the second uses transpose? [12:42:38 CEST] <c_14> maybe hflip,vflip is faster than transpose,transpose [12:44:22 CEST] <sheenobu> transpose=1 is 90 degrees, but scale=576:-2 rotates and scales? [12:44:48 CEST] <sheenobu> to get 180? [12:45:05 CEST] <c_14> no? [12:45:09 CEST] <c_14> scale doesn't rotate [12:45:12 CEST] <c_14> it scales [12:45:38 CEST] <sheenobu> right. how does transpose=1,scale=576:-2 make 180? [12:45:52 CEST] <c_14> it doesn't [12:46:00 CEST] <c_14> hflip,vflip does 180 [12:46:15 CEST] <sheenobu> ah. wallbroken said it backwards then, and I was /very confused/ [12:46:27 CEST] <sheenobu> or my IRC client got the messages backwards. [12:51:33 CEST] <wallbroken> c_14 the question is why is not used the same thing for both 90 and 180 [12:52:02 CEST] <c_14> 10:42 <c_14> maybe hflip,vflip is faster than transpose,transpose [12:52:56 CEST] <wallbroken> and is not possible to make transpose 180 ? [12:54:48 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#transpose [13:09:36 CEST] <misuzu__> I'm using hls stream as input for ffmpeg and server returns 403 for some reason. I found that ffmpeg sends Range header and this may be the reason it does not work. I found this on mail list: https://ffmpeg.org/pipermail/ffmpeg-cvslog/2012-October/055212.html and not sure how to use it. Is this patch already on upstream? If yes what command is used [13:09:37 CEST] <misuzu__> to disable range header? [13:14:17 CEST] <c_14> If it's in cvslog it's upstream, set -seekable 0 [13:15:11 CEST] <misuzu__> c_14: thanks! [13:15:23 CEST] <c_14> It should do that automatically though [13:15:27 CEST] <c_14> What version of ffmpeg are you running? [13:17:59 CEST] <misuzu__> 3.0.2 [13:18:08 CEST] <misuzu__> It works! [13:18:16 CEST] <misuzu__> c_14: Thank you! [14:02:11 CEST] <Hababam> hi, does any of yo guys know how to set the SO_RCVBUF within ffmpeg ? [14:02:25 CEST] <Hababam> i have tried to edit the network core files, but it didn;t work [14:04:03 CEST] <DHE> for udp, add "?buffer_size=1048576" or whatever value you want [14:04:12 CEST] <DHE> tcp makes a distinction between send_buffer_size and recv_buffer_size [14:07:46 CEST] <Hababam> hmm i will try that [14:15:55 CEST] <Hababam> hmm, still getting the rtm missed X packets error [14:16:02 CEST] <Hababam> rtp * [14:17:04 CEST] <BtbN> if you're processing slower than you're getting data that will happen at some point. [14:17:59 CEST] <DHE> current versions of ffmpeg do have a speed multiplier displayed. you should be getting something >= 1.00x [14:18:44 CEST] <Hababam> hmm getting around 0.9 to 1 [14:19:20 CEST] <DHE> would you say it's going up or down? [14:19:33 CEST] <DHE> at the end of the day 1.00 is where you want to be for live content [14:19:53 CEST] <Hababam> it bouncing between them [14:20:12 CEST] <DHE> that does not leave me with a warm fuzzy feeling... [14:20:30 CEST] <Hababam> yeahh i know that feel -.- :/ [14:20:58 CEST] <DHE> Can you paste[bin] your commandline? go ahead and censor out any details that are private (rtmp URLs or passwords, etc) but leave all parameters present [14:21:22 CEST] <Hababam> sure 1 sec [14:22:28 CEST] <BtbN> usualy for a live stream you'd expect to start somewhere above x1.00, slowly approaching 1.00 from above. [14:22:58 CEST] <DHE> typically, since ffmpeg will be buffering the input before starting and hence be able to process the buffer first. [14:23:34 CEST] <Hababam> http://pastebin.com/raw/RJxEprrU [14:24:10 CEST] <BtbN> speed=0.562x [14:24:22 CEST] <BtbN> you're processing _way_ too slow. [14:25:04 CEST] <DHE> computer over [14:25:43 CEST] <Hababam> hmm, [14:25:52 CEST] <Hababam> let me check on antoher server i have [14:25:53 CEST] <DHE> it's just H264->raw which is a surprise. it's not CPU-bound, which means you're probably maxing out something else. [14:25:59 CEST] <BtbN> it's not related to the server [14:26:05 CEST] <BtbN> it's getting the stream just fine. [14:26:06 CEST] <DHE> 1080p at 25fps in raw pixels is probably too much [14:26:14 CEST] <BtbN> What you are doing with the stream locally is what is too slow. [14:26:42 CEST] <Hababam> hmm, i need the stream out of a decklink card at 1080p 25 fps [14:26:55 CEST] <BtbN> What is the input video? [14:27:04 CEST] <BtbN> so, resolution and framerate [14:27:22 CEST] <DHE> 1080p, 25fps apparently [14:27:23 CEST] <Hababam> 720p 25 [14:27:27 CEST] <Hababam> input [14:27:28 CEST] <BtbN> ah [14:27:31 CEST] <BtbN> Stream #0:0: Video: h264 (Main), yuv420p(tv, bt709), 1280x720, 25 fps, 25 tbr, 90k tbn [14:27:45 CEST] <BtbN> so you are upscaling and doing a pix_fmt conversion [14:27:50 CEST] <DHE> ah, input [14:27:55 CEST] <Hababam> yup [14:27:59 CEST] <BtbN> and one that doesn't look too optimized to me [14:28:04 CEST] <BtbN> so that could easily be your bottleneck [14:28:39 CEST] <Hababam> i just trew it together, so no optmimizing yet [14:29:10 CEST] <BtbN> I'm talking about yuv420p->uyvy422 [14:29:22 CEST] <DHE> software scaling alone is actually pretty good... [14:29:24 CEST] <DHE> this on linux? [14:29:29 CEST] <Hababam> yup ubuntu [14:29:36 CEST] <Hababam> 16.04 [14:29:43 CEST] <DHE> custom build by the looks of it? [14:29:49 CEST] <Hababam> yup [14:30:04 CEST] <Hababam> cause in the standard build there's no decklnk support [14:30:05 CEST] <DHE> if you run the "ffmpeg_g" version (with debug symbols) you can profile it running "perf top" (as root) in another terminal [14:30:18 CEST] <DHE> press 'q' to quit perf [14:31:08 CEST] <furq> you could maybe try using zscale instead of swscale [14:31:33 CEST] <Hababam> ffmpeg_g how would i run that ? [14:31:37 CEST] <Hababam> jsut ./ffmpeg_g [14:31:39 CEST] <Hababam> ? [14:31:49 CEST] <DHE> no, same ffmpeg commandline, just use the ffmpeg_g binary instead [14:33:45 CEST] <Hababam> i'm not quite sure how to do that [14:46:08 CEST] <DHE> ./ffmpeg_g -i rtsp://****@87.195.159.0:554/mode1 .... [14:48:51 CEST] <Hababam> bash: ./ffmpeg_g: No such file or directory [14:48:54 CEST] <Hababam> is the response [14:57:38 CEST] <DHE> it's in the compilation directory [15:03:31 CEST] Action: gmh is away: Idle [15:04:29 CEST] Action: gmh is back (gone 00:00:05) [15:06:10 CEST] <Elirips> Hello. I'm connection to rtsp stream over vpn using tcp. This works fine 2 out of 3 times, but sometimes I get '[rtsp @ 003fc100] Could not find codec parameters for stream 0 (Video: h264, 1 reference frame, none): unspecified size' and the hint to increase -analyzeduration. If set -analyzeduration and -probesize to 6000M, but still get the error sometimes. Any hints? see http://pastebin.com/pb4U30vM for the full command-line and ou [15:07:54 CEST] <Elirips> Even more confusing is, that it seems if I omit '-stimeout 30000' it will never fail - but also stall forever if the vpn connection is not enabled [15:10:31 CEST] <Elirips> also interesting, if watching traffic in wireshark, everytime when ffmpeg fails wireshark reports a retransmission of at least one package.. [16:21:04 CEST] <kauchman> Hello, I'm trying to live stream video from IP camera via ffmpeg and ffserver. Input from camera is rtp(h264) and i want to transcode it into webm so i can include it in web page. Now everything seems to work except that when i play the video, it plays from beggining of the feed file (that is when the ffmpeg command started). Is there a way to start playing "live" video? Thank you very much for any help. [16:25:00 CEST] <DHE> well webm isn't intended for real-time streaming like that. more like for youtube [16:27:25 CEST] <kauchman> Thank you. Is there maybe a more suited format? I tried Ogg(theora/vorbis) but it wouldnt load video at all, only the loader kept spinning [16:31:35 CEST] <kepstin> for live video in html5, you probably want to look into using DASH or HLS with a javascript player that handles the format. [16:33:06 CEST] <kauchman> Thank you, I will look into that. [16:56:19 CEST] <P4Titan> Hello all. I'd like to ask: what would the best way be to chose a specific C compiler to build ffmpeg? [16:56:31 CEST] <c_14> The first one you run across that isn't broken. [16:56:48 CEST] <P4Titan> Is there a way to specify directly [16:56:54 CEST] <c_14> --cc= afaik [16:56:55 CEST] <P4Titan> rather than rely on the $PATH variable [16:57:00 CEST] <c_14> in configure [16:57:08 CEST] <P4Titan> hmm, interesting [16:57:11 CEST] <c_14> configure should probably also read the environment CC variable [16:58:23 CEST] <P4Titan> Yes, --cc= does exist [16:59:15 CEST] <furq> there's also --cross-prefix if you're cross-compiling [17:00:07 CEST] <c_14> looks like it doesn't read CC [17:02:05 CEST] <P4Titan> what does --cross-prefix take [17:02:20 CEST] <P4Titan> a directory where gcc and other executables can be found? [17:04:41 CEST] <DHE> more like a prefix to the gcc name. like arm-gnu-linux-[gcc] [17:04:46 CEST] <DHE> (I made that up, not a real prefix) [17:06:18 CEST] <furq> P4Titan: all the cross compiler toolchain binaries will have a prefix [17:06:29 CEST] <furq> like i686-w64-mingw32 [17:06:34 CEST] <furq> if you're not cross compiling then don't use that [17:06:54 CEST] <P4Titan> my case is a little specific [17:07:37 CEST] <P4Titan> basically, I am using gcc and other tools that are specifically optimized to take C code and create a flash executable instead of a regular executable [17:07:51 CEST] <P4Titan> I just wish to route the ffmpeg compilation through that gcc compiler [17:08:16 CEST] <P4Titan> currently I set $PATH to include that gcc before the system one [17:08:22 CEST] <P4Titan> but that is rather bad practice [17:11:55 CEST] <vade> hi all. Im trying to get an AVFilterChain to do audio format conversion for me, and am having duration issues in my output stream. Im using the libavfilter api, using a buffer source filter configured with my input streams time base, sample rate, sample format, an aformat filter set up to my output streams sample rate and format, and finally a buffer sink filter configured for output format, sample rate, and linked in that order. Looking t [17:11:56 CEST] <vade> ffmpeg_filter.c, it *appears* unless I am reading it wrong, it actually goes buffer -> sink -> format ? and sink isnt configured at all except for channels? [17:23:58 CEST] <andross> hey ho [17:24:11 CEST] <andross> been a while but i decided to return to my project [17:24:26 CEST] <andross> i just compiled ffmpeg on ubuntu# [17:24:34 CEST] <andross> which files will i need for development on windows? [17:25:12 CEST] <andross> i cross compiled of course [17:29:00 CEST] <JEEB> andross: easiest is to do --prefix=/home/my_username/my_win_prefix [17:29:03 CEST] <JEEB> and then do make install [17:29:19 CEST] <JEEB> it will install all the "development" files into that prefix in the directory structure required [17:30:22 CEST] <andross> i think i might have done something like that already [17:30:27 CEST] <andross> im not sure, it was a while ago [17:30:53 CEST] <andross> do you remember when i was here before JEEB ? [17:33:43 CEST] <andross> JEEB: how do i do this prefix command, i tried it with configure but it said no such command [17:34:01 CEST] <andross> i currently instaleld everything into the same folder by mistake(?) i guess [17:34:13 CEST] <andross> although does that matter? [17:34:48 CEST] <andross> i installed into the same folder as the source code [17:35:27 CEST] <andross> (reminder im not a linux user so dont really know what im doing) [17:35:33 CEST] <JEEB> andross: ./configure --prefix=/one/two/something [17:35:40 CEST] <JEEB> when you configure your FFmpeg build [17:35:52 CEST] <andross> ah forgot the ./ i guess [17:35:58 CEST] <JEEB> uhh [17:36:02 CEST] <JEEB> how did you compile FFmpeg then? :D [17:36:06 CEST] <JEEB> or even more [17:36:12 CEST] <JEEB> how did you succeed in cross-compilation? [17:36:46 CEST] <andross> i was slightly more familiar with linux shell then [17:36:49 CEST] <andross> but promptly forgot [17:36:56 CEST] <andross> as i didnt use it [17:37:08 CEST] <andross> and the full set of options was provided by someone here [17:37:11 CEST] <JEEB> and of course you still require all the other configuration parameters [17:37:19 CEST] <JEEB> you just add the prefix [17:37:22 CEST] <andross> god damn id rather not go through all that again [17:37:30 CEST] <andross> i already have it all compiled within the source code folder [17:37:43 CEST] <JEEB> but that isn't in the form other things use it [17:38:09 CEST] <JEEB> also you don't need to necessarily re-compile, you just need to re-configure to set the prefix :P [17:38:30 CEST] <andross> i had a text document somewhere with the full configuration [17:38:44 CEST] <JEEB> and then you do make/make install so that the required stuff gets done and you should have a prefix that you can just tar/zipball and move to a windows host for usage [17:41:26 CEST] <andross> i found a text file i saved with a configuration: http://hastebin.com/tepigusaqa.lua [17:41:30 CEST] <andross> but i dont believe it's the right one [17:41:39 CEST] <andross> i mean it doesnt look like a windows build [17:42:27 CEST] <andross> im sure i saved the right one in a text file somewhere [17:42:47 CEST] <andross> is there a way i can see recent text files with gedit? [17:44:13 CEST] <andross> okay i think i found it from config.log [17:45:13 CEST] <andross> how doe sthis look JEEB: [17:45:17 CEST] <andross> ./configure --cross-prefix=i686-w64-mingw32- --arch=i686 --target-os=mingw32 --disable-static --enable-shared --pkg-config='pkg-config --static' --extra-ldflags=-static-libgcc --enable-libmp3lame --prefix=/home/andross/ffmpeg-win [17:47:35 CEST] <andross> what the hell, the command doesnt work any longer [17:47:50 CEST] <andross> ERROR: libmp3lame >= 3.98.3 not found [17:50:07 CEST] <andross> i feel like i remember getting that error before but i have no idea how i resolved it [17:50:12 CEST] <andross> i def have lame installed [17:53:59 CEST] <andross> anyone?? [17:59:01 CEST] <andross> JEEB :( [17:59:31 CEST] <andross> do i need to do that path garbage [18:07:58 CEST] <vizier> Hey i have ffmpeg on mac and i just want to setup a script so that whenever i download a file to a folder it will convert it and put it in a new folder [18:08:32 CEST] <andross> this is driving me absolutely insane [18:08:53 CEST] <andross> i cant work like this, i cant just stare at a screen completely dumbfounded [18:08:54 CEST] <vizier> Can anyone help with this i dont know much about what im doing but i read that this was a fast and easy background process to setup so it automatically converts [18:09:21 CEST] <thebombzen> andross [18:09:37 CEST] <thebombzen> did you try --pkg-config='i686-w64-mingw32-pkg-config --static' [18:10:08 CEST] <thebombzen> I think --pkg-config overrides --cross-prefix [18:10:33 CEST] <thebombzen> also try --enable-cross-compile [18:10:41 CEST] <thebombzen> so it doesn't try to execute the binaries. [18:13:28 CEST] <thebombzen> well okay [18:13:48 CEST] <thebombzen> vizier: does wget exist on a mac? [18:13:54 CEST] <thebombzen> otherwise try using cURL [18:14:05 CEST] <vizier> i used homebrew to install ffmpeg [18:14:17 CEST] <furq> vizier: https://github.com/emcrisostomo/fswatch [18:17:11 CEST] <vizier> i think i can use automator app like this guy suggested http://apple.stackexchange.com/questions/129929/automating-ffmpeg-using-aut… [18:17:34 CEST] <thebombzen> I automate stuff with cron. but that's just me. fswatch sounds like exactly what you're looking for. [18:18:14 CEST] <vizier> ok i will look that that i didnt know this was going to be so hard lol [18:19:06 CEST] <vizier> ffmpeg just converts right you cant move or rename stuff to a suggested format like sickrage or something right [18:19:58 CEST] <vizier> i hardly know anything about what im doing im a biochemist [18:23:18 CEST] <andross> hiii [18:23:24 CEST] <andross> sorry someone highlighted me right before my pc crashed [18:23:38 CEST] <andross> think his name began with a v? [18:23:45 CEST] <andross> voltagex ? [18:24:57 CEST] <andross> fuck my stupid fucking pc for crashing at the worst possible moment [18:25:49 CEST] <andross> and fuck irssi for not keeping logs [18:28:42 CEST] <andross> you there JEEB ? [18:28:56 CEST] <andross> vizier ? [18:29:58 CEST] <iive> <andross> i cant work like this, i cant just stare at a screen completely dumbfounded [18:29:58 CEST] <iive> <vizier> Can anyone help with this i dont know much about what im doing but i read that this was a fast and easy background process to setup so it automatically converts [18:30:17 CEST] <andross> hey iive [18:30:27 CEST] <andross> someone highlighted me just after that, then my pc crashed [18:30:54 CEST] <iive> <thebombzen> andross [18:30:54 CEST] <iive> <thebombzen> did you try --pkg-config='i686-w64-mingw32-pkg-config --static' [18:30:54 CEST] <iive> <thebombzen> I think --pkg-config overrides --cross-prefix [18:30:54 CEST] <iive> <thebombzen> also try --enable-cross-compile [18:30:54 CEST] <iive> <thebombzen> so it doesn't try to execute the binaries. [18:31:08 CEST] <andross> hi thebombzen [18:31:57 CEST] <thebombzen> HELLO [18:31:58 CEST] <thebombzen> oops [18:32:02 CEST] <thebombzen> hello. did you try that [18:32:10 CEST] <andross> one second [18:32:33 CEST] <furq> there's no need for mingw to have its own pkg-config [18:32:42 CEST] <furq> just set PKG_CONFIG_PATH to point to your mingw libs [18:32:51 CEST] <andross> im almost certain that command i pasted worked previously [18:33:03 CEST] <andross> at least when building inside of the source code folder [18:33:08 CEST] <thebombzen> vizier: ffmpeg takes an input file name and an output filename. you can let those be whatever you want. and in different directories [18:33:10 CEST] <andross> the only difference is the prefix line [18:33:19 CEST] <furq> like i said, you probably need to set PKG_CONFIG_PATH if it can't find libraries [18:33:42 CEST] <andross> hey furq it may have been you that originally helped me with this [18:33:46 CEST] <furq> probably [18:34:06 CEST] <andross> can you remember roughly the PKG_CONFIG_PATH command i needed? [18:34:25 CEST] <andross> because i dont know where the mingw libs even are [18:34:29 CEST] <vizier> if someone would be rich if they made a gui where you can select an input folder, convert file inside it, apply metadata, rename it given a selected format, and move it to an output folder [18:34:32 CEST] <furq> there should be a lib/pkgconfig directory somewhere which is full of .pc files [18:34:37 CEST] <furq> presumably in the directory where you built ffmpeg [18:34:51 CEST] <furq> `find . -name *.pc` if you can't find it [18:35:04 CEST] <furq> then just set PKG_CONFIG_PATH to that directory before running configure [18:35:27 CEST] <andross> and that will fix the lame not found issue? [18:36:58 CEST] <furq> it should do [18:38:26 CEST] <andross> here's the output when i run the find command furq: http://hastebin.com/tohohijiwe.js [18:39:02 CEST] <furq> replace . with / [18:40:19 CEST] <andross> furq: http://hastebin.com/jidizaxuku.avrasm [18:40:23 CEST] <andross> thanks for helping btw [18:42:43 CEST] <andross> is it /usr/local/lib/pkgconfig ? [18:43:13 CEST] <andross> since it has stuff like libavcodec.pc [18:43:32 CEST] <furq> nvm i just checked and lame doesn't bother installing a .pc [18:43:36 CEST] <furq> search for lame.h instead [18:44:20 CEST] <andross> furq: http://hastebin.com/caxodipahu.rb [18:45:20 CEST] <furq> --extra-cflags=-I/home/andross/build/include --extra-ldflags=-L/home/andross/build/lib [18:45:23 CEST] <furq> should do it [18:45:40 CEST] <andross> on the configure command? [18:45:42 CEST] <furq> yeah [18:45:47 CEST] <andross> great thanks [18:45:49 CEST] <furq> add the ldflags to your existing --extra-ldflags [18:46:00 CEST] <furq> if you specify it twice the second one will override the first [18:47:04 CEST] <furq> also thank you for reminding me that i still need to sort out this mingw build system ;_; [18:47:17 CEST] <furq> stupid libgme [18:48:08 CEST] <N_1ck> hi there, i have a n00bie question : i'm trying to do this : https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20inp… However when i run the provided example script i end up with : " -i : command not found " ... I understand this is because the file is multi-line... what should i do ? [18:48:25 CEST] <andross> furq: what's the syntax to add flags to the existing option? [18:48:26 CEST] <furq> remove the newlines? [18:48:32 CEST] <N_1ck> to keep it 'readable' [18:48:48 CEST] <furq> --extra-ldflags="-static-libgcc -L/home/andross/build/lib" [18:49:57 CEST] <andross> how does this look furq: [18:50:00 CEST] <andross> ./configure --cross-prefix=i686-w64-mingw32- --arch=i686 --target-os=mingw32 --disable-static --enable-shared --pkg-config='pkg-config --static' --extra-ldflags="-static-libgcc -L/home/andross/build/lib" --extra-cflags=-I/home/andross/build/include --enable-libmp3lame --prefix=/home/andross/ffmpeg-win [18:50:19 CEST] <furq> looks fine [18:51:43 CEST] <andross> i think it worked :O [18:51:58 CEST] <andross> now doing make install [19:01:40 CEST] <andross> this is taking longer than i thought [19:01:47 CEST] <andross> i think it is compiling... [19:01:57 CEST] <furq> you can cancel it and run make -j4 (or however many cores you have) [19:02:01 CEST] <furq> it'll pick up where it left off [19:02:22 CEST] <andross> well this is a virtual machine and i think its just using one core [19:02:45 CEST] <furq> i'm getting deja vu here [19:03:07 CEST] <andross> by the way will this delete the compiled files in the source folder or will there be two copies now [19:03:57 CEST] <furq> it'll overwrite them [19:04:57 CEST] <andross> woah i think its doen [19:05:02 CEST] <andross> that was quicker than i thought [19:05:32 CEST] <andross> so i now have a 25.8mb folder called ffmpeg-win [19:05:50 CEST] <andross> and i just copy this to my win pc? [19:05:57 CEST] <furq> sure [19:06:11 CEST] <furq> and then hope it doesn't segfault [19:06:27 CEST] <furq> or more likely hope it doesn't complain about missing dependencies [19:06:39 CEST] <furq> it should be fine though [19:13:20 CEST] <andross> aaaright [19:16:10 CEST] <andross> so i notice it builds both .a and .lib files [19:16:15 CEST] <andross> which do i want to include? [19:17:08 CEST] <furq> i don't think you want either if you're linking against the shared libs [19:17:17 CEST] <furq> iirc mingw should spit out .dll.a import libraries [19:17:26 CEST] <andross> yeah thats what i mean [19:17:32 CEST] <andross> so i want to include the .dll.a files? [19:17:34 CEST] <furq> yeah [19:17:46 CEST] <furq> i think .lib is msvc's name for those, but i don't know why mingw would produce those [19:28:35 CEST] <andross> hmm, seems using ffmpeg in Qt might not be as straightforward as i thought [19:31:08 CEST] <andross> maybe its simpler if i use ffmpeg binaries and run them as an external process heh [19:32:34 CEST] <furq> if you just need to encode pcm to mp3 then use lame directly [19:32:38 CEST] <furq> the library, not the binary [19:32:50 CEST] <andross> want to do aac as well [19:32:58 CEST] <andross> and who knows what else [19:40:52 CEST] <andross> how come the files in bin have these number suffixes [19:40:58 CEST] <andross> e.g. avcodec-57.dll [19:42:32 CEST] <JEEB> it's a way of telling which API version you're linking against [19:42:50 CEST] <JEEB> every time a change in API is done that might require a re-linking the number gets +1'd [19:43:01 CEST] <andross> ah [19:43:57 CEST] <JEEB> on *nix usually development packages have f.ex. avcodec.so (read: like dll) which is a symbolic link to the versioned one [19:44:22 CEST] <JEEB> so you can link against avcodec but the final binary will depend on avcodec-57 f.ex. [19:45:02 CEST] <JEEB> (that way you can also have multiple versions of the library in your system in case of global library locations - on windows the DLLs tend to be located next to the binaries) [20:10:40 CEST] <andross> god maybe ive made a huge mistake [20:12:53 CEST] <grotesk> Hi @ all, maybe someone of you could help me, i'm trying to write an ios app in swift, to get the live stream from my garmin virbxe and stream it to youtube live, the camera is sending an rtsp stream with h264 video and aac audio. [20:13:06 CEST] <grotesk> here is my output http://pastebin.com/uJKdWEaS [20:13:14 CEST] <andross> furq: when i do the command av_register_all() [20:13:17 CEST] <andross> it just crashes [20:14:08 CEST] <andross> error code 0xc0000135 [20:25:30 CEST] <andross> furq: is there an easy way to verify the libraries built correctly? as right now im unsure if the problem is with the libraries or with QtCreator [20:25:55 CEST] <furq> run the binary? [20:26:36 CEST] <andross> interesting [20:26:45 CEST] <andross> cant start because libmp3lame-0.dll is missing [20:27:29 CEST] <furq> if you built a shared lib then you'll need that in the same directory [20:27:39 CEST] <furq> if you built a static lib of lame then shrug [20:27:51 CEST] <andross> yeah its share [20:29:24 CEST] <andross> okay, the executable seems to work as in, it displays the help text [20:30:18 CEST] <andross> i dont know how much i need to test, or if that alone confirms it built properly? [20:30:53 CEST] <furq> if it gets that far then it should be fine [20:31:04 CEST] <andross> ill be be back in a bit need to get food [20:48:07 CEST] <grotesk> Is there an easy way to figure out, how to use this command line as c code? ffmpeg -i rtsp://url -acode copy -vcode copy c:/name.flv [21:09:31 CEST] <andross> alright im back [21:10:00 CEST] <andross> furq: tbh i think i jumped the gun and im now way in over my head prob [21:10:17 CEST] <andross> it somehow never even occured to me that ffmpeg is in c while my project is c++ [21:11:22 CEST] <JEEB> yes, the way to correctly use the C interfaces in C++ is noted in the FAQ [21:13:54 CEST] <andross> well yeah im using extern "C" already [21:14:07 CEST] <JEEB> yup [21:14:26 CEST] <JEEB> I used to make a simple thing to decode the first frame of a matroska file and have explorer show it as a thumbnail [21:14:46 CEST] <JEEB> the funny part was that it worked for a while on win7, until some random windows update made it start crashing [21:14:58 CEST] <JEEB> this was circa 2013 [21:18:01 CEST] <kristian_on_linu> hi [21:18:19 CEST] <JEEB> ohai [21:18:40 CEST] <kristian_on_linu> I have an mp3 file that is 30 minutes long, made up of ... 12 individual pieces or so. I know how long each song is. Can I somehow cut it out? [21:18:50 CEST] <kristian_on_linu> I mean, in a smart way ... [21:19:25 CEST] <JEEB> ffmpeg -ss [seekpoint timestamp] -i input.mp3 -t [for how long] -c copy out.mp3 [21:19:40 CEST] <JEEB> also see ffmpeg-all.html for documentation on these options on ffmpeg.org [21:23:42 CEST] <kristian_on_linu> so I have to do it X number of times? [21:23:50 CEST] <JEEB> yes [21:24:08 CEST] <JEEB> if you know the spots you can script [21:25:37 CEST] <kristian_on_linu> thanks [21:31:18 CEST] <andross> already having trouble with the example [21:31:31 CEST] <andross> where is CODEC_ID_MP2? [21:31:48 CEST] <andross> its not recognized [21:32:24 CEST] <JEEB> ./include/libavcodec/avcodec.h: AV_CODEC_ID_MP2 [21:32:34 CEST] <JEEB> you're looking at something rather old if you don't have the AV_ prefix [21:32:41 CEST] <JEEB> I recommend looking at the docs/examples dir [21:32:47 CEST] <JEEB> that has examples for demuxing, decoding etc [21:32:56 CEST] <andross> hmm [21:33:16 CEST] <andross> yeah i was looking here https://ffmpeg.org/doxygen/trunk/encoding-example_8c-source.html [21:33:36 CEST] <furq> andross: if you don't need to decode a wide variety of formats then i'd probably just use libmp3lame/libfdk-aac directly [21:33:56 CEST] <JEEB> uhh [21:33:59 CEST] <furq> the only advantage of using libavcodec for this is if you need to decode anything other than pcm [21:34:00 CEST] <JEEB> fdk-aac can't be distro'd [21:34:40 CEST] <andross> furq: my understanding is that ffmpeg has lots of powerful wrapper functions that makes things easier or something [21:34:41 CEST] <JEEB> unless he or his employer complies to the software license's binary redistribution part [21:34:43 CEST] <andross> or so i was told [21:35:17 CEST] <JEEB> so for AAC I'd almost just go with the latest internal libavcodec aac encoder unless you need HE-AAC) [21:35:29 CEST] <JEEB> that way you're clearly in LGPL territory [21:35:45 CEST] <furq> oh yeah i forgot lame is lgpl [21:36:01 CEST] <JEEB> I actually just thought of that too [21:36:15 CEST] <JEEB> yeah, it is [21:37:02 CEST] <JEEB> andross: funny how that example still uses the non-AV-prefixed thing [21:37:08 CEST] <furq> andross: it's simpler for a lot of things but for pcm->mp3 using lame directly would be much easier [21:37:27 CEST] <JEEB> true [21:37:36 CEST] <JEEB> if that's all you do, just using LAME's API could be simpler [21:37:41 CEST] <furq> likewise with fdk but you can't distribute a binary that links to lame and fdk [21:37:51 CEST] <furq> so you'd be stuck with libfaac which sucks [21:37:54 CEST] <JEEB> uhh [21:37:56 CEST] <furq> actually is that even gpl-compatible [21:38:00 CEST] <JEEB> faac breaks its own license [21:38:03 CEST] <JEEB> it's supposed to be GPL [21:38:05 CEST] <furq> yeah i remembered something like that [21:38:07 CEST] <drv> where did you find this doxygen link? it's from 2011 (not sure why it's still on the web server) [21:38:09 CEST] <JEEB> but it's not really because it uses non-GPL-compatible code [21:38:10 CEST] <furq> it'd be libvo-aacenc then [21:38:14 CEST] <furq> which is even worse [21:38:18 CEST] <JEEB> well, no [21:38:25 CEST] <JEEB> in that case you build a limited libavcodec [21:38:29 CEST] <JEEB> with the AAC encoder [21:38:32 CEST] <JEEB> and use that [21:38:35 CEST] <furq> i meant if you were doing this without using libav* [21:38:47 CEST] <JEEB> well that makes sense if there are good alternatives to libavcodec [21:38:51 CEST] <JEEB> like LAME itself [21:38:52 CEST] <furq> exactly [21:38:59 CEST] <furq> but since fdk is out, there's no good aac encoder [21:39:04 CEST] <JEEB> uhh [21:39:09 CEST] <furq> other than the ffmpeg one obv [21:39:11 CEST] <JEEB> even the current lavc internal one? [21:39:13 CEST] <JEEB> yes [21:39:21 CEST] <JEEB> which is when it becomes the "alternative" [21:39:39 CEST] <JEEB> you can still use LAME straight and lavc separately [21:39:44 CEST] <JEEB> or use lavc for both [21:39:49 CEST] <furq> you might as well use the same code path for both though [21:39:57 CEST] <JEEB> yup [21:41:05 CEST] <furq> i really ought to figure out why ffmpeg keeps stopping for no reason [21:41:10 CEST] <furq> i bet it's mintty's fault [21:41:11 CEST] <andross> drv: i just googled ffmpeg encoding example or something like that and it was the first thing that came up [21:41:33 CEST] <drv> anyway, that is just a listing of a source file from the ffmpeg tree, so you can find an up-to-date copy there [21:42:05 CEST] <JEEB> the github mirror has probably the best web view https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples [22:35:23 CEST] <vade> what is the latest on hardware encode for ffmpeg via libavcodec API? is still mostly up to you to enable partcular paths in code, or does avcodec_encode_video2 handle this for you now-a-days if you pass right codec ? [22:35:42 CEST] <andross> sorry im [22:35:43 CEST] <andross> back [22:35:50 CEST] <andross> got interrupted by a phonecall [22:37:53 CEST] <andross> god damn it [22:38:00 CEST] <andross> as soon as i try to use any function furq it just crashes [22:38:22 CEST] <andross> during program startup [22:38:43 CEST] <andross> im sure theres something wrong with the linker [22:39:28 CEST] <andross> in my .pro i have unix|win32: LIBS += -L$$PWD/../../ffmpeg-win/lib/ -lavcodec -lavformat -lavutil [22:39:42 CEST] <andross> INCLUDEPATH += $$PWD/../../ffmpeg-win/include [22:57:55 CEST] <andross> was just missing the dlls in the build folder apparently [23:06:53 CEST] <andross> all seems to be working now JEEB furq just need to work through the examples i guess, thanks again [23:09:38 CEST] <andross> good way to learn c a bit more too i guess [23:19:54 CEST] <DHE> vade: while there's a new API available, the avcodec_encode_video2 API is still working. and yes, for hardware support it's just a matter of selecting the right encoder (eg: nvenc rather than libx264) [23:20:15 CEST] <vade> DHE - any documentation on this new API? [23:20:22 CEST] <vade> apologies if Ive missed it :) [23:24:49 CEST] <DHE> vade: http://ffmpeg.org/doxygen/trunk/group__lavc__decoding.html#ga9395cb802a5feb… it's still pretty new though. it's not in the 3.0 release [23:25:56 CEST] <vade> ah. interesting. Any options for encode too ? [23:26:18 CEST] <DHE> avcodec_(send|receive)_(frame|packet) as required [23:26:24 CEST] <vade> ah I see [23:26:28 CEST] <DHE> but that's all API stuff. if you use the ffmpeg tool who cares? [23:27:29 CEST] <vade> Well, im using the API stuff :) [23:27:38 CEST] <DHE> well then [23:30:11 CEST] <andross> welp [23:30:17 CEST] <andross> the following code appears to cause a segfault: [23:30:20 CEST] <andross> const enum ffmpeg::AVSampleFormat *p = codec->sample_fmts; [23:30:47 CEST] <andross> ffmpeg:: is because i wrapped the header include directives within an ffmpeg namespace [23:31:29 CEST] <vade> is codec null? have you registered your formats and codecs via register_all or whatever it is [23:31:55 CEST] <DHE> that sounds like something easily identified with a debugger [23:32:32 CEST] <andross> codec = ffmpeg::avcodec_find_encoder(ffmpeg::AV_CODEC_ID_MP2); [23:33:26 CEST] <andross> you're right tho i havent done avcodec_register_all(); [23:34:05 CEST] <JEEB> which means nothing is initialized [23:34:11 CEST] <andross> got it [23:35:25 CEST] <andross> i guess that fixed it [23:36:22 CEST] <vade> DHE: is h264_videotoolbox /videotoolboxenc.c scheuled for the next release? [23:37:05 CEST] <vade> and btw JEEB I figured out my resampling / timing issues. Everything is pretty much dead on now. Wooop woop :) [23:37:26 CEST] <andross> what is 's16' exactly [23:37:36 CEST] <DHE> I would imagine so, but I don't do releases. I'm just another API user [23:37:39 CEST] <vade> short 16 bit [23:37:40 CEST] <JEEB> signed, 16 [23:37:42 CEST] <DHE> andross: signed [23:37:44 CEST] <vade> oops sorry [23:37:47 CEST] <andross> oh duh [23:41:24 CEST] <andross> tbh i dont think i've ever played an mp2 file in my life lol [23:41:27 CEST] <vade> is the new recieve / send packet frame functionally the same as encode video2 audio 4, but just a unified path for audio / video ? [00:00:00 CEST] --- Thu May 26 2016

1 0

ffmpeg-devel.log.20160524
by burek 25 May '16

25 May '16

[00:06:41 CEST] <cone-609> ffmpeg 03ZhouXiaoyong 07master:05a546181ff1: avcodec/mips/blockdsp_mmi: Version 2 of the optimizations for loongson mmi [00:06:42 CEST] <cone-609> ffmpeg 03ZhouXiaoyong 07master:377e5db3db4a: avcodec/mips/pixblockdsp_mmi: Version 2 of the optimizations for loongson mmi [00:06:43 CEST] <cone-609> ffmpeg 03ZhouXiaoyong 07master:8392794c9227: avcodec/mips/idctdsp_mmi: Version 2 of the optimizations for loongson mmi [02:01:47 CEST] <philipl_> exit [04:40:38 CEST] <cone-513> ffmpeg 03Michael Niedermayer 07master:4c82ccad2b3c: avformat/mov: Avoid "Unintended sign extension" [13:21:52 CEST] <ln-> oh, ffmpeg has a tech support and customer service phone in canada [13:26:53 CEST] <iive> pages deleted, ip blocked. [13:28:22 CEST] <thardin> they're tenacious [13:28:42 CEST] <BtbN> It might be time to entirely block public editing of the wiki. [13:30:34 CEST] <iive> i think we might try some tweaking of the anti-spam engine [13:31:08 CEST] <thardin> putting something simple like "what's the name of this project?" as a text input usually goes a long way [13:31:08 CEST] <BtbN> I don't think it's too much of a problem to require manual activation of new accounts [13:31:22 CEST] <thardin> that's what the mxflib forums do [13:32:23 CEST] <thardin> spammers are only going to put so much effort into a custom antispam thing, if any [13:49:43 CEST] <michaelni> if someone has a patch for trac to make registration require specifying the project name that would be welcome [14:53:35 CEST] <Compn> finally someone gave me vqc1 sample :) [14:54:09 CEST] <durandal_170> Compn: what it is? [14:56:08 CEST] <Compn> old codec [14:56:13 CEST] <Compn> doesnt look to be mjpeg [14:56:33 CEST] <Compn> https://samples.ffmpeg.org/drivers32/new/VQC1/ [14:57:17 CEST] <farfel_> not sure if this is the right place - I have a question about libavcodec API [14:58:04 CEST] <Compn> farfel_ : are you working on ffmpeg or libavcodec? or are you working on your own project ? [14:58:15 CEST] <farfel_> yes, own project [14:58:20 CEST] <Compn> then probably ask in #ffmpeg [14:58:25 CEST] <farfel_> ok, thanks [14:58:28 CEST] <Compn> no problem [15:23:02 CEST] <Compn> durandal_170 : want sample ? [15:25:02 CEST] <BBB> I really dont think the trac spam is that much of an issue [15:25:16 CEST] <BBB> I mean, it doesnt hurt us in any way, we just manually delete some pages [15:25:57 CEST] <BtbN> As long as they don't start editing legitimate pages, yeah [15:28:47 CEST] <Compn> says the person not on spam/trac patrol... [15:28:56 CEST] <Compn> ask llogan or carl about it [15:46:06 CEST] <omerjerk> Hi everyone [15:46:25 CEST] <omerjerk> Carl had previously suggested to use softfloat api instead of this - https://github.com/omerjerk/FFmpeg/blob/float/libavcodec/alsdec.c#L289 [15:46:55 CEST] <omerjerk> but softfloat api doesn't seem to have the sign bit data - https://www.ffmpeg.org/doxygen/2.2/structSoftFloat.html [15:50:02 CEST] <omerjerk> and if I do av_float2int(float value) >> 31, that'd defeat the whole purpose I guess. [15:54:45 CEST] <jkqxz> What is wrong with sign bit = mant < 0? (Or value < 0.0 while it was a float.) [16:07:30 CEST] <omerjerk> umm okay. that's also correct. [16:08:01 CEST] <omerjerk> jkqxz: So I'll go ahead with your solution. [16:09:00 CEST] <omerjerk> I'll send my patch to the mailing list by tomorrow, to get it reviewed further. [16:15:43 CEST] <andrey_turkin> I wonder what's happening with my nvenc patches. Nobody looked at them yet? Or nobody's going to because of some reason? [16:29:23 CEST] <BBB> I dont think theres any specific reason [16:36:57 CEST] <jkqxz> andrey_turkin: I think BtbN was working on related things (CUDA/NVENC), and might be able to make some comment on it here. [16:49:17 CEST] <andrey_turkin> mmkay... guess I'll keep waiting for some reaction then [17:00:00 CEST] <BtbN> Did I miss a patch on the ml? [17:01:03 CEST] <andrey_turkin> http://ffmpeg.org/pipermail/ffmpeg-devel/2016-May/194502.html and thread replies [17:01:28 CEST] <BtbN> Anyway, I'm about to replace the nvenc encoder in ffmpeg with a merged libav/ffmpeg one. So patches aren't exactly easy at the moment. [17:02:11 CEST] <andrey_turkin> ah shoot. So i guess I did that for nothing [17:02:37 CEST] <andrey_turkin> if you are going to merge in libav features then these patches are useless [17:02:57 CEST] <BtbN> https://git.khirnov.net/libav.git/log/?h=nvenc that's the current status. Patching is done so far, "just" need to review and test for regressions from the ffmpeg implementation. [17:03:30 CEST] <BtbN> And I have to figure out how to do the actual merge. I guess just plainly replacing it would be the easiest. [17:04:24 CEST] <andrey_turkin> hmm, maybe you still can you my first 3 patches to bring ffmpeg version a bit closer to libav [17:05:04 CEST] <BtbN> I'm already short on time for just one review. [17:06:25 CEST] <andrey_turkin> anyway, as long as CUDA frames support lands in ffmpeg I'd be happy [17:06:37 CEST] <BtbN> Isn't that mostly useless at the moment? [17:06:49 CEST] <andrey_turkin> not for me [17:06:55 CEST] <nevcairiel> the libav version supports cuda frame input [17:07:04 CEST] <BtbN> yes, but nothing else supports cuda frames. [17:07:21 CEST] <andrey_turkin> I know but libav doesn't support things from ffmpeg like e.g. frame metadata [17:08:56 CEST] <BtbN> I'll try to get the merge on its way this weekend. [17:10:07 CEST] <BtbN> Not sure how to handle it though, can't realy send a merge to the ML. [17:10:15 CEST] <andrey_turkin> why not? [17:10:26 CEST] <BtbN> I could send a link to the merged branch [17:13:14 CEST] <andrey_turkin> what's wrong with sending a patch to the ML? Of course it would look scary but still [17:13:51 CEST] <nevcairiel> some people care to finish their work before sending it =p [17:14:24 CEST] <BtbN> There is no simple patch [17:14:35 CEST] <andrey_turkin> I don't see much difference between sending a patch to ML and pointing to a branch with the same commitdiff [17:14:39 CEST] <BtbN> I'm pretty sure format-patch on the merge commit won't produce anything sensible [17:22:46 CEST] <andrey_turkin> so let me reiterate; you got all relevant stuff from ffmpeg and merged it to libav (so libav now has best of both worlds when it comes to nvenc). Now you want to bring it all back to ffmpeg and the plan is to make a commit which basically replaces nvenc implementation with libav version. Sounds about right? [17:22:46 CEST] <michaelni> ive added some simple custom anti spam check to ffmpeg trac, tests & comments welcome [17:23:28 CEST] <pfelt> greatest of morning everyone! [17:23:42 CEST] <BtbN> andrey_turkin, i want to merge the development branch we used. [17:23:50 CEST] <BtbN> So a merge commit, two parents and stuff. [17:25:46 CEST] <nevcairiel> any distinct changes should ideally be submitted as such [17:26:36 CEST] <nevcairiel> having one huge merge change everything causes headaches for bug finding later [17:27:37 CEST] <andrey_turkin> I didn't think libav and ffmpeg actually merged to each other (as in git merges) [17:29:59 CEST] <BtbN> well, we took the libav nvenc encoder, and put all missing features and fixes from ffmpeg into it. [17:30:07 CEST] <BtbN> So there aren't realy any distinct changes on the ffmpeg side. [17:46:54 CEST] <andrey_turkin> BtbN: so this branch you pointed to - is this that's slated to get into ffmpeg? I mean, is it going to be a merge with --strategy=their so whole existing content will be thrown out and replaced with what's in this branch? [17:51:23 CEST] <BtbN> not sure how to do that yet [18:01:39 CEST] <michaelni> BtbN, IIUC you didnt work on the ffmpeg codebase ? if so theres no magic its just not in ffmpeg [18:02:32 CEST] <BtbN> yes, it's a libav tree. Just copying over the nvenc files and adjusting the build system works though. [18:02:59 CEST] <andrey_turkin> it works, but that commit would be hell to review [18:03:32 CEST] <michaelni> BtbN, thats possible but isnt that a maintaince nightmare ? [18:03:51 CEST] <BtbN> yes, that's why I'm not sure how to do the merge yet. [18:04:33 CEST] <andrey_turkin> if you (anyone) wants, I can try to break out some of the difference between the two codebases into (semi)readable chunks, so that remaining diff is more approachable [18:07:34 CEST] <michaelni> BtbN, without looking at the diff, i suspect that spliting the diff manually into self contained changes is the only clean option [18:07:52 CEST] <andrey_turkin> as I said I'd be happy to do that [18:08:10 CEST] <BtbN> that sounds close to impossible, as the implementations aren't common at all. [18:08:55 CEST] <michaelni> i still dont understand why you didnt do the work on the ffmpeg codebase ? [18:09:06 CEST] <michaelni> in the first place that is ... [18:09:26 CEST] <BtbN> Less stuff to change on the libav side to reach a common set of features. [18:10:16 CEST] <BtbN> We could also just merge your patches on the ML, and would have achived pretty much the same. Which might be the better thing to do now, as you already wrote them. [18:10:19 CEST] <andrey_turkin> BtbN: I think it's not that hard to bring diff down a lot. Remaining churn will be harder to deal with but it would be at least redable [18:12:46 CEST] <andrey_turkin> It would be great to have same (or at least minimally divergent) code at least in some parts of the two combined codebases [18:16:15 CEST] <michaelni> BtbN, you are the maintainer of that code, you have to decide what you prefer / what is best. I dont object to replace the code if you prefer to maintain the resulting code and history [18:22:24 CEST] <BtbN> andrey_turkin, your patches bring the ffmpeg encoder quite close to the libav one, I think the remaining differences(primarily the diffrent dts/pts calculation from what I can see) can be merged individually. [18:27:30 CEST] <andrey_turkin> my thoughts exactly [18:28:00 CEST] <andrey_turkin> and then it's just 2-3 more mechanical patches to rename some fields and we are done [18:32:50 CEST] <andrey_turkin> well, if we are going this way then my initial question remains - please someone review and apply my patches ) [18:33:42 CEST] <BtbN> I hope next weekend is still early enough? [18:37:54 CEST] <andrey_turkin> I don't really care about timeframes as long as someone eventually gets to them. I'll be doing some more of these in the mean time [20:23:23 CEST] <BtbN> michaelni, the antispam redirects me to a page that trac has no handler for. [20:24:04 CEST] <BtbN> oh, nevermind. Didn't notice that I was on the mplayer trac... [20:37:01 CEST] <cone-557> ffmpeg 03Michael Niedermayer 07master:35b37591c034: avfilter/af_hdcd: Replace complexer expression by litteral constant [20:37:01 CEST] <cone-557> ffmpeg 03dsmudhar 07master:04d35004c3d0: libavfilter/vf_hwdownload: show error when ff_formats_ref fails [20:59:05 CEST] <cone-557> ffmpeg 03dsmudhar 07master:49640ae315ab: libavcodec/dv_tablegen: fixed wrong printf format [21:39:00 CEST] <BBB> anyone know what type of algebra Im looking for? I have a list of N numbers and am trying to create M (where M <= N) averages of subgroups such that the sum of some kind of error over each number and its subgroup average is minimized [21:39:14 CEST] <BBB> brute-force is somewaht easy but subideal [21:42:09 CEST] <fritsch> BBB: can you write the optimization problem in tex? [21:42:20 CEST] <fritsch> BBB: so that I am sure what you are looking for? [21:42:21 CEST] <BBB> I dont know tex :-p [21:42:38 CEST] <fritsch> M averages of subgroups [21:42:52 CEST] <fritsch> example: {1,2,3,4,5} -> [21:42:55 CEST] <fritsch> tell me :-) [21:43:27 CEST] <fritsch> i want to get the definition of subgroups, is it elements of the power set? [21:43:31 CEST] <fritsch> non disjunct? [21:43:37 CEST] <fritsch> or exclusiv and non overlapping? [21:44:04 CEST] <fritsch> or in other words: the way you formulate your problem does not ring a bell :-) [21:47:49 CEST] <BBB> lets say you have {1,2,3,4,5,6,7,8,9,10} (N=10) and M=2 [21:48:54 CEST] <BBB> if I create two averages of 3 and 7, then my sum_error (lets assume my error metric is sse) is (3-1)^2 + (3-2)^2 etc. up until 6, where its (7-6)^2 up to 10 [21:49:25 CEST] <BBB> I Can just try two averages and set them iteratively between anything in between 1 and 10 and try everything [21:49:27 CEST] <BBB> but thats slow [21:49:54 CEST] <BtbN> andrey_turkin, do you happen to have a git branch somewhere i can pull the nvenc patches from? [21:50:15 CEST] <andrey_turkin> I can upload them [21:52:06 CEST] <fritsch> BBB: is the list sorted? [21:53:26 CEST] <fritsch> what you are basically doing is a k-means with k=M [21:53:48 CEST] <fritsch> so you need a clustering algorithm [21:54:13 CEST] <fritsch> see: https://en.wikipedia.org/wiki/K-means_clustering [21:54:43 CEST] <fritsch> it's np hard :-) [21:54:55 CEST] <fritsch> so if you find a better solution tell me, hehe [21:55:50 CEST] <andrey_turkin> BtbN: https://github.com/tea/FFmpeg/commits/nvenc-merge . There is one extra patch at the front of the queue not related to nvenc per se [21:55:57 CEST] <BBB> ok so it is k-means [21:56:09 CEST] <fritsch> from what you describe, yes [21:56:10 CEST] <BBB> k-means kind of sucks [21:56:15 CEST] <BBB> and yes the list can be sorted [21:56:27 CEST] <fritsch> find M clusters of N elements [21:57:07 CEST] <BBB> yes [21:57:16 CEST] <fritsch> so that all elements in a cluster M' of M have a distance d' which is minimal compared to other clusters M'' [21:59:17 CEST] <fritsch> BBB: the results needs to be "fully correct"? [22:02:00 CEST] <BBB> no [22:02:07 CEST] <BBB> just something that is not entirely stupid [22:02:24 CEST] <BBB> when does anything ever have to be correct, right? I mean, its not like Im shooting rockets at Pluto :-p [22:04:00 CEST] <fritsch> fast approximate nearest neighbour [22:14:35 CEST] <fritsch> BBB: if you get the array sorted it's much easier, if it's sorted and disjunct, e.g. no doublettes, even more and if it's sequential it's trivial :-) [00:00:00 CEST] --- Wed May 25 2016

1 0

ffmpeg.log.20160524
by burek 25 May '16

25 May '16

[02:17:42 CEST] <Mista_D> Is there a way to detect duplicate video frames without ImageMagick? [02:50:30 CEST] <throckmorton> I am trying to add jpeg 2000 still image support to VLC [02:51:09 CEST] <throckmorton> when I try to open a file, I get get_buffer failed error from avcodec [02:51:59 CEST] <throckmorton> [jpeg2000 @ XXXXX] get_buffer() failed [02:52:16 CEST] <throckmorton> what is the best way of troubleshooting this ? [03:00:06 CEST] <spooooon> I'm not an expert, but can you breakpoint and see the callstack [03:00:12 CEST] <spooooon> or grep ffmpeg for the failure string [04:04:21 CEST] <yagiza> Hello! [04:46:23 CEST] <gxm1221> ffmpeg is awesome!! [04:46:25 CEST] <gxm1221> really like it :) [09:13:28 CEST] <YellowOnion> Could someone help me with an error, I just finished compiling ffmpeg on windows with opencl support -opencl_bench works, but unsharp is giving me an error (which I'll paste shortly) [09:16:40 CEST] <YellowOnion> http://pastebin.com/At4YnFEb [09:23:36 CEST] <YellowOnion> actually never mind figured it out. [10:24:42 CEST] <YellowOnion> is unsharp broken for anything larger than a 7 matrix? the contrast goes out the window, you get strange banding. [10:37:54 CEST] <YellowOnion> And if your matrix is really high, you actually get the opposite of sharpening for positive values. ha, and if you use a png as a target the picture goes completely haywire. [11:09:57 CEST] <Pisa_Dev> Hi All. I need to install ffmpeg in a Phi Server without root permission, so i cannot use apt. I need all the dependences, like lib264, for install FFMpeg. Is there any possibility to download all dep and install it? I followed a Guide in FFMpeg site, but most of the packages are not avaiable. [11:32:36 CEST] <YellowOnion> Pisa_Dev, you're probably better off downloading the .deb's manually and extracting them [11:37:14 CEST] <Pisa_Dev> Hi YellowOnion! Thanks for the reply [11:37:31 CEST] <Pisa_Dev> i'm in a Red Hat [11:38:03 CEST] <Pisa_Dev> yum don't work properly nor apt source [11:38:36 CEST] <YellowOnion> can you do it on another machine and push the files there? [12:05:22 CEST] <yagiza> Do anyone know anything about resampling delay? [12:06:12 CEST] <YellowOnion> yagiza, vaguely, why is that? [12:07:01 CEST] <yagiza> YellowOnion, ??? [12:07:59 CEST] <yagiza> YellowOnion, I just want to know, what do I have to do with it? [12:11:12 CEST] <YellowOnion> yagiza, Well that kinda depends on what you're doing in the first place. [12:11:30 CEST] <yagiza> YellowOnion, I'm encoding audio. [12:12:35 CEST] <YellowOnion> just audio? no video? samplerates? what resampler are you using? [12:12:39 CEST] <yagiza> YellowOnion, and I need to resample audio samples I'm getting from audio device to format, supported by audio codec. [12:12:52 CEST] <yagiza> YellowOnion, swr [12:16:27 CEST] <YellowOnion> yagiza, unless I'm misunderstanding the situation, delay from resampling isn't going to be your primary issue with delay when you're using an audio codec. [12:17:33 CEST] <yagiza> YellowOnion, I don't have any issues with delay at all so far. [12:18:11 CEST] <yagiza> YellowOnion, but I just wanna know if I have to take that delay into account during encoding. [12:18:25 CEST] <YellowOnion> plus I'm not sure what the delay is compared to? if its just audio then who cares? [12:19:42 CEST] <yagiza> YellowOnion, right now I just add it to number of processed samples when calculating audio frame PTS. [12:20:09 CEST] <yagiza> YellowOnion, but I'm not sure if that's what I should do. [12:22:20 CEST] <YellowOnion> Are you worried about track desync? [12:23:55 CEST] <yagiza> YellowOnion, yes [12:27:27 CEST] <YellowOnion> I would assume ffmpeg handles this itself, assuming the sources are in sync to begin with. [12:44:45 CEST] <YellowOnion> yagiza, though if your sample rate is very low (like 8k) you might end up with smearing of of the transients which is caused by the group the group delay :P [12:51:08 CEST] <wallbroken> hor to rotate a video of 90 degrees? [13:12:23 CEST] <Mrc_munir> Hello i have some problem linker -fPIC with clang compiler for x86_64 in android platform but we use --enable-pic how fix i.e libswscale.a(rgb2rgb.o): requires dynamic R_X86_64_PC32 reloc against 'ff_w1111' which may overflow at runtime; recompile with -fPIC? [14:19:23 CEST] <Hababam> hi [14:19:57 CEST] <Hababam> does any of your guys have experience with rstp input to ffmpeg ? [14:20:05 CEST] <Hababam> i get allot of missing rtp packets [14:20:11 CEST] <Hababam> but in vlc [14:20:15 CEST] <Hababam> the stream shows nicely [14:48:06 CEST] <farfel> hello - question about jpeg 2000 support in ffmpeg. I see there is a native decoder and optional openjpeg decoder [14:48:54 CEST] <farfel> I have built ffmpeg with openjpeg enabled. I have another project which uses avcodec [14:49:18 CEST] <farfel> ffmpeg decodes j2k files using openjpeg, but the other project uses the native decoder [14:49:46 CEST] <farfel> so, with two decoders, how does an avcodec user choose which one to use ? [14:50:12 CEST] <farfel> other project is videolan [14:50:42 CEST] <farfel> both projects use the same libavcodec library [14:51:04 CEST] <farfel> i.e. ffmpeg and videolan use the same libavcodec library [14:51:42 CEST] <JEEB> videolan is a group of people and projects under an umbrella [14:51:49 CEST] <farfel> vlc project [14:52:20 CEST] <farfel> I want to add j2k support to vlc for still images, via avcodec [14:52:38 CEST] <farfel> and openjpeg seems to support more kinds of j2k images [14:53:02 CEST] <wallbroken> hor to rotate a video of 90 degrees? [14:53:06 CEST] <JEEB> there's a way to force a decoder, but unfortunately I don't remember how to use that API [14:53:19 CEST] <farfel> ok, thanks anyways [14:53:46 CEST] <c_14> wallbroken: https://ffmpeg.org/ffmpeg-filters.html#transpose [14:53:55 CEST] <farfel> JEEB: so, if I don' [14:54:04 CEST] <farfel> t force, it will default to native ? [14:54:26 CEST] <farfel> both avcodec and openjpeg are built as shared libraries, if this makes a difference [14:54:38 CEST] <farfel> I thought maybe vlc wasn't loading openjpeg, since it was shared [14:55:42 CEST] <farfel> is this an appropriate question for ffmpeg-devel ? [15:41:42 CEST] <odinsbane> I encoded a video using ffmpeg, but I cannot play it back using quicktime. VLC works, I'm wondering if you have any suggestions. Also I need to use the h.264 encoding. [15:41:46 CEST] <odinsbane> ffmpeg -i "out%03d.png" -c:v libx264 -qp 0 lf200-low-res.mp4 [15:44:06 CEST] <DHE> just off the top of my head, try a higher qp value like 15? 0 is codec-lossless mode [15:54:51 CEST] <odinsbane> No matter which way I do it, quicktime won't play the videos. The best part is when it sends me to a webpage explaining how great quicktime is for playing all sorts of videos. [15:55:19 CEST] <relaxed> odinsbane: add -pix_fmt yuv420p [15:55:19 CEST] <JEEB> quantizer zero won't work [15:55:26 CEST] <JEEB> it's lossless [15:55:29 CEST] <JEEB> as was mentioned [15:55:44 CEST] <JEEB> otherwise see the output pixel format in the ffmpeg cli logs and force it in case it is not yuv420p [15:55:56 CEST] <odinsbane> I tried specifing it with a different value, I also tried omitting it. [15:56:13 CEST] <odinsbane> (qp value that is. I am trying the px value now.) [15:56:23 CEST] <JEEB> don't change it if it's already yuv420p [15:56:27 CEST] <JEEB> see your damn logs [15:56:41 CEST] <JEEB> aka "don't throw things in without reason" [15:57:00 CEST] <JEEB> and if you can't parse your logs, post them on a pastebin or gist [15:59:01 CEST] <DHE> if you use "-profile:v main" would it reject non-4:2:0p content? [16:05:24 CEST] <odinsbane> It was defaulting to yuv444p, If I specify yuv440p and set the qp value to 15 it is playable. I am supposed to use lossless compression though. Why doesn't qp 0 work? [16:05:50 CEST] <odinsbane> er 420p [16:11:26 CEST] <furq> odinsbane: does -qp work with yuv420p [16:11:31 CEST] <DHE> different decoder profiles are required. 420p is broadcast standard, 444 requires the decoder to be more "sophisticated" to work [16:11:33 CEST] <furq> -qp 0, that is [16:11:56 CEST] <DHE> yes, but it's not true lossless. 4:4:4 -> 4:2:0 degrades the image quality even before H.264 gets it [16:12:03 CEST] <furq> yeah what he said [16:12:21 CEST] <odinsbane> I swear the person giving these specs is making some sort of joke. Thanks for the help. For some odd reason yuv420p with crf 0 works. yuv420p and qp 0 does not work. [16:12:33 CEST] <kepstin> -qp 0 works with yuv420p, but the lossless mode isn't supported by the normal 'high' profile. [16:13:26 CEST] <kepstin> so a high profile only decoder can't handle it, only a high444pp decoder can iirc. even at 4:2:0 sampling. [16:13:59 CEST] <odinsbane> scratch that. [16:14:41 CEST] <odinsbane> I cannot get qp 0 or crf 0 to produce a video that quicktime will play. [16:15:03 CEST] <kepstin> it's only the predictive lossless coding that's not supported, you might be able to do a lossless encode that's intra-only which would work :/ [16:15:12 CEST] <kepstin> not sure tho [16:15:27 CEST] <furq> odinsbane: do the specs say that it has to work with quicktime [16:15:45 CEST] <furq> if so then yeah you should go and shout at someone [16:16:03 CEST] <odinsbane> furq: no, but everybody that I work with wants to see the video and they use quicktime. [16:16:38 CEST] <odinsbane> Ill just use qp 18 so everybody can see it and complain about how tiny the output video is. [16:16:55 CEST] <furq> you should use crf over qp 99% of the time [16:18:04 CEST] <kepstin> (most of the remaining 1% of the time is lossless mode, where -qp 0 is preferred) [16:19:07 CEST] <odinsbane> crf 18 works too, so Ill use that. [16:21:32 CEST] <kepstin> crf 18 will give a pretty high quality (visually) encode, but if this is an intermediate file that might be re-encoded a few more times, I'd probably use a lower setting (higher quality). [16:22:35 CEST] <odinsbane> The same file will probably not be re-encoded. I really suspect something is wrong with these specs. [16:22:53 CEST] <odinsbane> Ill keep it in mind though when we are on the final version. [16:25:45 CEST] <kepstin> (crf numbers are similar scale to qp numbers, except they take psy optimizations into effect - so the actually selected qp will vary per frame - or per macroblock even - based on frame complexity and motion) [16:25:54 CEST] <farfel> good morning [16:26:29 CEST] <farfel> if avcodec supports two codecs for the same file type, is there a way of forcing it to use a particular codec ? [16:27:44 CEST] <kepstin> farfel: sure, using the -codec option (e.g. -codec:v or -c:v to select video codec) [16:28:36 CEST] <farfel> kepstin: thanks a lot. In my case, I have vlc linking to avcodec. Is there an API for doing this from another program ? [16:29:11 CEST] <kepstin> farfel: are you using vlc or developing it? [16:29:23 CEST] <farfel> a bit of both :) [16:29:30 CEST] <kepstin> if using, you'll want to talk to the vlc folks to see if they provide a ui or interface to do it [16:30:01 CEST] <farfel> thanks, I have been talking to them [16:30:16 CEST] <kepstin> when using libavcodec, you can just select an encoder or decoder using an exact name (e.g. 'libvpx-vp9' instead of the generic tag ('vp9') [16:30:52 CEST] <farfel> ok, cool. so for jpeg 2000, it would be libopenjpeg-jp2k ? [16:30:57 CEST] <farfel> I think jp2k is the fourcc [16:31:50 CEST] <kepstin> the generic name for that in ffmpeg is 'jpeg2000'. I don't have multiple encoders enabled in my build, so I'm not sure the name. You can try running "ffmpeg -codecs | grep jpeg2000" to see what you have. [16:32:36 CEST] <farfel> kepstin: thanks, I really appreciate the help here [16:33:51 CEST] <farfel> ok, so the name of the openjpeg decoder is libopenjpeg, and the native one is called jpeg2000 [16:34:30 CEST] <kepstin> yeah, the native one usually has the same name as the generic format name, so it's preferred by default in most cases. [16:47:29 CEST] <farfel> kepstin: so, in vlc, if it uses the generic tag for jpeg 2000, would that be jp2k ? [16:48:00 CEST] <kepstin> no, it would be 'jpeg2000'; 'jp2k' isn't a codec name (although it might be an alias, i'm not sure) [16:48:18 CEST] <farfel> well, in vlc code, I can't find any reference to jpeg2000 [16:48:25 CEST] <kepstin> what's the container format? [16:48:40 CEST] <kepstin> it might be just using the "default codec" pulled from the container [16:48:58 CEST] <farfel> ok, in my case, I am trying to add support for still j2k images to vlc [16:49:07 CEST] <farfel> so, a bit unusual for a video player, I guess [16:49:24 CEST] <kepstin> btw, encoding or decoding? [16:49:36 CEST] <farfel> decoding [16:51:07 CEST] <kepstin> hm. as a still image, i'd guess it's probably using the image2 demuxer, which recognizes the j2k extension, and gives the codec AV_CODEC_ID_JPEG2000, which would cause it to use the default jpeg2000 decoder. [16:51:35 CEST] <kepstin> is there some particular problem you're having with the internal jpeg2000 decoder? [16:52:41 CEST] <farfel> Thanks. native decoder seems to have problems opening files from the openjpeg conformance suite [16:53:33 CEST] <kepstin> you'd have to add some code to vlc to load a particular decoder (only if available in the ffmpeg build) rather than the default for that codec id if you want to do that. [16:54:13 CEST] <kepstin> obviously it would be best to get the internal jpeg2000 decoder fixed to decode the test files correctly ;) [17:03:29 CEST] <farfel> kepstin: yes, absolutely [17:03:42 CEST] <farfel> these are conformance images, so not too pathological [17:05:32 CEST] <farfel> kepstin: actually, I see that vlc uses AV_CODEC_ID_JPEG2000 [17:05:43 CEST] <farfel> so, what would I need to change it to to use libopenjpeg ? [17:06:19 CEST] <farfel> oh, sorry, never mind [17:06:40 CEST] <farfel> as you said, this loads the default [17:07:23 CEST] <kepstin> farfel: probably have to switch from avcodec_find_decoder to avcodec_find_decoder_by_name("libopenjpeg") - and make sure to fallback if that codec's not available. [17:07:41 CEST] <farfel> awesome, thanks a lot [17:08:16 CEST] <farfel> I see in avcodec that both libopenjpeg and native decoder share the same id [17:08:27 CEST] <farfel> AV_CODEC_ID_JPEG2000 [17:08:42 CEST] <kepstin> yeah, that identifies the codec itself, not any particular implementation [17:08:56 CEST] <farfel> I see, thanks [17:10:01 CEST] <kepstin> the avcodec_find_decoder finds more or less the "best" decoder for that codec, depending on some relative ordering the ffmpeg devs did and experimental flags. [17:11:36 CEST] <farfel> ok, thanks [17:11:57 CEST] <farfel> this is very helpful [17:19:04 CEST] <farfel> kepstin: as an experiment, is it possible to re-order the codecs to put libopenjpeg at the front? [17:21:34 CEST] <farfel> actually, never mind, need to fix this in vlc [17:21:41 CEST] <farfel> if [17:37:24 CEST] <farfel> kepstin: so it looks like vlc is failing to process these images for other reasons, it is not the codec [17:37:28 CEST] <farfel> so that is good news [17:37:48 CEST] <farfel> get_buffer() failed for some reason [17:38:13 CEST] <grrk-bzzt> Hello [17:38:40 CEST] <grrk-bzzt> To have a lossless x265 video, should I use -qp 0 or -x265-params lossless=1 ? [17:39:31 CEST] <farfel> I forced vlc to use libopenjpeg, but still errors out [17:39:54 CEST] <kepstin> farfel: did it return an error code or print a message? [17:40:08 CEST] <kepstin> farfel: ffmpeg has some arbitrary limits on image size and memory, iirc [17:40:38 CEST] <farfel> yes, prints an error message [libopenjpeg 0xXXXXX] get_buffer() failed [17:41:26 CEST] <farfel> ahhh, yes, but, ffmpeg can read these images [17:41:27 CEST] <kepstin> grrk-bzzt: I think you have to use the -x265-params option. [17:41:37 CEST] <kepstin> grrk-bzzt: the x265 site says "Note that QP 0 does not cause lossless encoding, it only disables quantization." [17:41:38 CEST] <farfel> no problem [17:41:46 CEST] <grrk-bzzt> thank you kepstin [17:41:59 CEST] <farfel> perhaps ffmpeg doesn't use the same code path [17:42:01 CEST] <farfel> as vlc [17:42:16 CEST] <kepstin> the ffmpeg cli tool is definitely a different code path than vlc. [17:42:35 CEST] <kepstin> but I'm not sure why it would change a get_buffer failing inside the decoder. [17:43:51 CEST] <farfel> there are four error messages, thread_get_buffer() error, get_buffer() error, thread_get_buffer() error, get_buffer() error [17:44:10 CEST] <farfel> for each image load attempt [17:46:57 CEST] <farfel> unfortunately, the error code is not logged [17:48:13 CEST] <farfel> I can add more debugging information to see what is going on [17:48:39 CEST] <kepstin> farfel: Hmm. is vlc overriding the get_buffer(2) callbacks on the codec context? [17:49:14 CEST] <kepstin> could be something in the vlc code, if that's the case :/ [17:49:17 CEST] <farfel> overriding ? not sure [17:49:20 CEST] <farfel> let me check [17:49:45 CEST] <farfel> what code should I look for? [17:49:48 CEST] <farfel> for override ? [17:50:07 CEST] <kepstin> something setting the get_buffer or get_buffer2 member on an AVCodecContext somewhere [17:50:44 CEST] Action: kepstin is heading out; good luck. [17:51:36 CEST] <farfel> ok, thanks very much for your help! [18:00:10 CEST] <dbugger> Hello everyone [18:01:30 CEST] <dbugger> Can anyone please take a look at my command? http://hastebin.com/ehimevalap.erl I am taking 3 videos, putting them on top of one another, and then putting them next to another media, but for some reason the first columns is 872px, and should be 874px... what did I do wrong? [18:14:33 CEST] <Jookia> Extracting audio out of a 1:17:26 matroska stream gives me a 1:17:17 desynced track- are there tools I can use to extract it accurately? [18:36:50 CEST] <Jookia> Okay, I just needed to use the resampling filter! Yay [19:28:44 CEST] <dbugger> Can someone please take a look at my question? http://stackoverflow.com/questions/37418785/composition-on-ffmpeg-does-not-… [20:26:01 CEST] <lilibox> hi [20:27:01 CEST] <lilibox> does anybody could point me at page where is describet the best setting for this: i have got sequence of .dpx files, one .wav and i need to mix it together to one nice .h264 2k standard file [20:27:23 CEST] <lilibox> unfortunaly i don't have time to experimenting :( [20:27:41 CEST] <lilibox> any well known preset for this avaiable around? [20:27:47 CEST] <lilibox> thank you [20:29:23 CEST] <BtbN> .h264 isn't exactly a standard file. [20:29:30 CEST] <BtbN> Also, it's raw h264 video, so no audio. [20:30:41 CEST] <lilibox> well, my video is targeted to one electro store, it will be played from USB stick directly on TV set probably [20:31:46 CEST] <lilibox> i do graphics, but i put all my outputs as sequence of 10bits .dpx files, rest is on guys from post/broadcast department [20:32:03 CEST] <JEEB> mp4 with level 4.1 H.264, AAC [20:32:04 CEST] <JEEB> done [20:32:06 CEST] <lilibox> so, i have not much experiences with video formats on tvs :( [20:32:19 CEST] <lilibox> i will let it on yours guys, i belive you :) [20:32:36 CEST] <lilibox> JEEB ok, level 4.1 means what? [20:33:06 CEST] <JEEB> it's setting a limit to the encoder to not make a stream that requires more memory/bandwidth from the decoder than defined in that given level in the H.264 spec [20:33:21 CEST] <JEEB> 4.1 is usually used with HD for most HW decoders, as it tends to be supported [20:33:53 CEST] <JEEB> profiles = features, levels = (max) requirement of RAM/bandwidth from the decoder for the stream [20:33:56 CEST] <JEEB> in H.264 [20:34:42 CEST] <lilibox> ok, soo, please could you point me or write it out directly here, which command sequence i should use? as i said, i have got sequence of .dpx in one folder - named video and in second folder named audio is one stereo .wav file [20:34:53 CEST] <JEEB> the audio part is simple [20:35:02 CEST] <JEEB> the multiple dpx files is not gonna be [20:35:41 CEST] <JEEB> there's an image2 input "format" which lets you give a string format thing for multiple images, but I'm definitely not sure if dpx is supported with that [20:35:52 CEST] <JEEB> other than that, it should be rather simple [20:36:56 CEST] <JEEB> ffmpeg [stuff for the video input] -i audio/input.wav -c:v libx264 -c:a aac -level 41 -vf scale=-2:1080,format=yuv420p out.mp4 [20:37:09 CEST] <lilibox> ok, thank you for small edu lessons, i am going to finish shot and then will tell you [20:37:21 CEST] <lilibox> JEEB great, it is exactly, what i wanted to see [20:37:22 CEST] <lilibox> :) [20:37:47 CEST] <JEEB> ffmpeg cli by default selects one video and audio stream, so if you just put two -i (inputs) with one having video and the other audio, it should "just work" [20:38:01 CEST] <lilibox> ok [20:38:20 CEST] <JEEB> rate control for libx264 by default is the equivalent of "-crf 23", so if it looks bad you set a lower CRF value, and if it looks good and you want to compress it more you push it upwards [20:38:35 CEST] <JEEB> (the idea with CRF is to find the highest CRF value that still looks good to you) [20:38:53 CEST] <JEEB> not sure about the default rate control for the internal aac encoder, but it should be sane enough [20:39:12 CEST] <JEEB> (if not, do -b:a 192k which sets audio bit rate to 192 kilobits per second) [20:39:40 CEST] <JEEB> buut yea, the image input is probably the hardest part for you :P [20:39:49 CEST] <JEEB> which I conveniently left out [20:40:01 CEST] <lilibox> ok, i am off for a while, thank you much again [20:40:08 CEST] <JEEB> as I haven't used the image2 stuff and I don't know if it works with your image format [20:41:43 CEST] <furq> do you really want -2:1080 over 1920:-2 [21:08:46 CEST] <kbarry> So, I'm trying to 1) detect silence on a stream, and 2) Alert. [21:09:38 CEST] <kbarry> In my search, I came across SilentJack, but am not sure that its rea''y what I need. It might be that I am just too much a newb to really understand how JACK/SilentJack work, [21:27:49 CEST] <lilibox> back [21:28:45 CEST] <lilibox> and please does ffmpeg do not support imagefilename.0012.dpx name convention? [21:31:42 CEST] <JEEB> image2 has a hack for setting an input file name template [21:31:45 CEST] <JEEB> in general, no [21:32:08 CEST] <JEEB> https://www.ffmpeg.org/ffmpeg-all.html#Examples-7 [21:32:30 CEST] <JEEB> if setting the sequence string correctly doesn't work then sorry, nope [21:32:38 CEST] <lilibox> JEEB your link ended with this error: error opening filters [21:32:50 CEST] <JEEB> wat [21:32:53 CEST] <lilibox> i just put ffmpeg.exe to windows folder [21:33:02 CEST] <lilibox> no other files used ad mine side [21:33:27 CEST] <JEEB> full command line and terminal output to pastebin or gist, please [21:33:32 CEST] <JEEB> otherwise you are making no sense [21:33:43 CEST] <JEEB> also the page is long so make sure you have the browser load the page [21:33:45 CEST] <lilibox> i got something like this: Size value less than -1 are not acceptable. [21:34:03 CEST] <pandaologist> hey, trying to find a volume meter filter that shows the volume on a logarithmic scale [21:34:06 CEST] <pandaologist> https://ffmpeg.org/ffmpeg-filters.html#showvolume [21:34:18 CEST] <pandaologist> this appears to be a linear scale [21:34:36 CEST] <pandaologist> is anyone aware of another way to display in log? [21:34:51 CEST] <JEEB> lilibox: stop guessing if any random part you happened to notice is useful for debugging or not. you clearly have no idea what you're doing so please post full command line and terminal output on a pastebin or gist [21:34:56 CEST] <JEEB> whichever is simpler for you [21:34:57 CEST] <lilibox> and second line [Parsed_scale_0 @ 020987e0] Failed to configure output pad on PArsed_scale_0 Error opening fileters! [21:35:10 CEST] <JEEB> ok, it's clear you don't want to be helped. have a nice evening [21:35:50 CEST] <lilibox> oh, wait, i just retyped what other window shows, sorry [21:36:11 CEST] <lilibox> i didn't read what are you typeing :(, sorry [21:36:31 CEST] <JEEB> you can select text in cmd.exe by right clicking the terminal [21:36:45 CEST] <JEEB> both complete command line and terminal output [21:36:48 CEST] <kepstin> pandaologist: as far as I can tell, that filter is doing the volume in decibels, which is a logarithmic scale. [21:36:49 CEST] <lilibox> yes, i will paste it to pastebin and send you link in a while [21:36:55 CEST] <JEEB> thank you [21:37:09 CEST] <pandaologist> kepstin: the value displayed is in decibel correct [21:37:27 CEST] <pandaologist> but the scaling of the volume bars (the visual representation) is definitely linear [21:38:58 CEST] <kepstin> hmm, the code in that filter is not very obvious at all :/ [21:39:28 CEST] <lilibox> JEEB: http://pastebin.com/s7dkBq0Q [21:39:44 CEST] <pandaologist> yeah, just looking to see if it's a simple change too [21:40:28 CEST] <furq> lilibox: that's a really old version of ffmpeg [21:40:36 CEST] <furq> i assume it predates the ability to use -2 in the scale filter [21:40:39 CEST] <JEEB> lilibox: that's old enough to not have -2 which is a shorthand [21:40:45 CEST] <JEEB> just grab a new one from zeranoe's [21:40:53 CEST] <JEEB> since the AAC encoder is vastly superior now too [21:41:00 CEST] <lilibox> ok [21:41:06 CEST] <JEEB> https://ffmpeg.zeranoe.com/builds/ [21:44:38 CEST] <vade> hey JEEB - so converting my codebase to use an intermediary AV_TIME_BASE representation fixed like 90% of my problems. Only thing remaining that I cant quite hone in on is whom / what is responsible for altering packets if I use an AVFilter which uses a source and a sink that have different sample rates. [21:45:30 CEST] <vade> if I feed in 44.1 audio samples to my filter chain, and write out 48Khz audio, I get a shorter audio track [21:45:41 CEST] <vade> but its *not* shorter by the ratio of 48/44.1 [21:47:16 CEST] <lilibox> JEEB thanks, seems like .mp4 works fine [21:47:38 CEST] <lilibox> gotta put it on usbstick and check it on tv set [21:48:07 CEST] <JEEB> and you checked that the quality was good enoguh? no need to poke the CRF value? [21:48:21 CEST] <JEEB> but yeah, checking with a TV set makes sense [21:48:31 CEST] <JEEB> also I guess you set frame rate accordingly [21:48:49 CEST] <JEEB> seems like you didn't at least in the pastebin example [21:48:54 CEST] <JEEB> default just happens to be 25Hz [22:09:54 CEST] <lilibox> quality was fine, well, i forgot to tell you, that this clip had added Kodak film grain [22:10:34 CEST] <furq> if you want to preserve the grain then try -tune grain [22:10:41 CEST] <JEEB> not required [22:10:59 CEST] <JEEB> if the quality is good enough for him, then no further tweaking is required [22:11:29 CEST] <JEEB> lilibox: just make sure your frame rate is correct (without setting anything it's 25Hz/fps) [22:11:32 CEST] <furq> well i mean technically nothing is required on account of the universe is dying and all human endeavour is pointless [22:11:40 CEST] <JEEB> -_- [22:11:46 CEST] <furq> but we have to while away the time somehow [22:11:48 CEST] <lilibox> yes, PAL region here [22:11:51 CEST] <JEEB> ok [22:12:31 CEST] <JEEB> so yeah, check the clip on your hw device(s) and then you're done [22:27:23 CEST] <Ricardo_> Hi all. Can anyone point me to docs that explain how I can reconstruct a RFC6381 compliant codec string based on the ffprobe output? (e.g., "avc1.42E01E") [22:28:54 CEST] <thebombzen> I have a question - I'm trying the new OCR filter that uses tesseract. Is there any way to get it to dump the contents of OCR as soon as it encodes them? [22:29:14 CEST] <Ricardo_> based on what i'm seeing, I can easily determine the profile (via a profile name) and the profile level... but I can't seem to find the byte containing the contraint_set flags. [22:30:32 CEST] <wallbroken> ffmpeg -i input -vf "hflip,vflip,scale=1024:-2" [22:30:33 CEST] <wallbroken> ffmpeg -i input -vf "transpose=1,scale=576:-2" [22:30:44 CEST] <wallbroken> is there some difference between those commands? [22:30:53 CEST] <wallbroken> asi i see, the first, rotates 180 [22:31:00 CEST] <wallbroken> the seconds, 90 [22:31:16 CEST] <wallbroken> but why the first uses hflip, vflip but the second uses "transpose" ? [23:39:49 CEST] <kbarry> Are there tools to detect silence/dB levels in ffmpeg/play/probe/etc?) [23:40:19 CEST] <kbarry> I can't find anything specifically, but thought maybe it just wasn't being done, but was certainly possible. [23:40:50 CEST] <furq> http://ffmpeg.org/ffmpeg-filters.html#silencedetect [23:42:15 CEST] <spooooon> I have decode AVFrames, I wish to display them using opengl [23:42:51 CEST] <spooooon> how can I pull the data out of the AVFrame, it looks like the data* is actualy 3 data* and it doesn't look very helpful [23:43:46 CEST] <spooooon> am I forced to use sws_scale? [23:44:54 CEST] <kepstin> well, in most cases you'll have planar YUV video... [23:45:41 CEST] <kepstin> depending what you want to do with the video, you could use libswscale to turn it into a different pixel format [23:45:41 CEST] <spooooon> I was planning to convert to RGB in the shaders with opengl [23:46:25 CEST] <kepstin> for planar yuv, you could do something like upload it as 3 separate textures, and have the shaders sample from all of them separately. [23:46:25 CEST] <spooooon> I just wanted to pass it to an opengl texture [23:46:32 CEST] <spooooon> oh I see [23:46:38 CEST] <spooooon> planar YUV is 3 textures [23:46:41 CEST] <spooooon> that makes sense [23:46:54 CEST] <spooooon> sorry I'm new to video processing, thanks! [23:47:42 CEST] <kepstin> yeah, unless you look into how different pixel formats work and are laid out in memory, you're gonna have a hard time with this. [23:48:09 CEST] <spooooon> can't I always have the output from the decoder output into AV_PIX_FMT_YUV410P? [23:48:17 CEST] <spooooon> then assume that is always the same [23:48:27 CEST] <spooooon> I'm only supporting H264, MJPEG and MPEG4 [23:48:38 CEST] <furq> h.264 can be rgb [23:49:02 CEST] <furq> and they can all be yuv422 or yuv444 iirc [23:49:31 CEST] <spooooon> so I'm in for a bad time unless I use sws_scale? [23:49:34 CEST] <furq> you could just have swscale convert it to 4:2:0 but in that case you might as well just have it output rgb [23:49:46 CEST] <kepstin> most of the time you will get yuv420p from those formats, but they could be in something else :/ [23:50:16 CEST] <spooooon> The problem is I could have lots and lots of video feeds, I'm trying my best to increase performance [23:50:25 CEST] <spooooon> and one way was to do conversions and scaling on the gpu [23:50:45 CEST] <kepstin> why are you writing your own opengl output anyways? you should be able to grab the one e.g. from mpv/mplayer [23:50:55 CEST] <furq> i believe zscale does higher-performance format conversions than swscale [23:50:58 CEST] <kepstin> which I think does support several pixel formats [23:50:59 CEST] <furq> it's still on the cpu though [23:51:17 CEST] <spooooon> It's part of a much larger applciation [23:51:53 CEST] <spooooon> I would be happy with doing the colour conversion on the CPU [23:52:09 CEST] <spooooon> is sws_scale the correct function to be looking at to only do colour conversion [23:52:43 CEST] <kepstin> yes, it can do color conversions without any scaling (has optimized codepaths for some common conversions, iirc) [23:52:52 CEST] <spooooon> cool thanks [23:53:08 CEST] <kepstin> or look into the zscale filter that furq mentioned, it can do that too, I think. [23:55:26 CEST] <furq> i think someone mentioned it's multithreaded, which swscale isn't [23:55:54 CEST] <spooooon> I have already seperated the workload into threads [23:56:06 CEST] <furq> it's not a noticeable difference when you're doing it alongside an x264 conversion though [23:56:20 CEST] <furq> which is the only time i've ever used it [00:00:00 CEST] --- Wed May 25 2016

1 0

ffmpeg-devel.log.20160523
by burek 24 May '16

24 May '16

[02:33:12 CEST] <cone-354> ffmpeg 03Petru Rares Sincraian 07master:89806e0b9add: fate: add aecho test [05:42:54 CEST] <cone-965> ffmpeg 03dsmudhar 07master:5836a5037ecb: libavfilter/af_hdcd: fixed negative signed value shift [05:42:54 CEST] <cone-965> ffmpeg 03dsmudhar 07master:84417593b9da: libavcodec/cfhd: fixed wrong printf format [11:54:33 CEST] <iive> the spam/scammers are back [12:11:55 CEST] <iive> and no admins around... what a shame. [12:12:01 CEST] <iive> michaelni [12:34:50 CEST] <michaelni> iive, i think i deleted them and the spammer, whats your account on trac ? (i can give you delete rights if you want to help) [12:35:06 CEST] <michaelni> in the future [12:36:02 CEST] <iive> should be the same as my nick [12:36:32 CEST] <iive> yes it is. [12:39:31 CEST] <michaelni> iive, done, be carfull with delete, its not easy to undo [13:22:18 CEST] <BBB> isnt ~((unsigned) -1 << 8) just 0xff? [13:22:29 CEST] <BBB> or do I have my order of operations mixed up? [13:23:08 CEST] <ubitux> 0xffff..ffff00 i'd say [13:23:23 CEST] <ubitux> oh i missed the ~ [13:23:25 CEST] <nevcairiel> sounds like it would be 0xff [13:23:32 CEST] <nevcairiel> seems odd either way [13:24:10 CEST] <ubitux> yes it's 0xff [13:44:34 CEST] <cone-609> ffmpeg 03dsmudhar 07master:6b852a3fd942: libavutil/opencl: fixed uninitialized var warning [19:59:49 CEST] <kierank> michaelni: taking some stuff offline [21:17:50 CEST] <durandal_1707> kierank: why? [21:18:06 CEST] <kierank> was upgrading switch [21:23:32 CEST] Action: michaelni was wondering too but when i tried ping it was up so i thought hmm whatever ... [21:31:17 CEST] <kierank> was only 20-30mins [21:50:02 CEST] <cone-609> ffmpeg 03Paul B Mahol 07master:5b8faaad6c71: avfilter/vf_blend: fix incorrect Y variable when threading is used [22:56:51 CEST] <cone-609> ffmpeg 03foo86 07master:9bd1d79584c6: avcodec/dca_exss: always update downmix flags [00:00:00 CEST] --- Tue May 24 2016

1 0

ffmpeg.log.20160523
by burek 24 May '16

24 May '16

[00:00:06 CEST] <furq> tlvb: it shouldn't make any difference for mpegts [00:04:59 CEST] <tlvb> furq: thankyou :) [00:10:47 CEST] <tlvb> hm, apparently the .MTS files are not likely exactly mpegts but rather the m2ts container (an extension of mpegts it seems) but I can't see any information on wikipedia that would likely change the answer [00:24:48 CEST] <FlorianBd> Hi there! [00:25:16 CEST] <FlorianBd> Is there any known issue of uploading a 6fps video file on youtube? (e.g. some devices would not like it?) [00:27:33 CEST] <m1dnight_> Hrrm, this streaming stuff is confusing. I have an rpi streaming over http, so that would the transfer protocol, right? But from the command I cant make up the codec and format? [00:27:48 CEST] <m1dnight_> vlc v4l2:///dev/video0 --v4l2-width 1920 --v4l2-height 1080 --v4l2-chroma h264 --sout '#standard{access=http,mux=ts,dst=0.0.0.0:12345}' <- That is what is streaming the video over http. [00:28:22 CEST] <m1dnight_> h264 is the codec, http is the protocol, but what is the format? I want to take this stream in on my server, and restream it over.. http I guess? [00:29:20 CEST] <c_14> >mux=ts [00:29:55 CEST] <m1dnight_> Disclaimer: Ive been spitting through all kinds of posts/docs the last day but it's just really really overwhelming :) [00:30:44 CEST] <c_14> mux = muxing (the format), ts = transport stream (short for mpeg-ts i.e. MPEG Transport Stream) [00:32:25 CEST] <m1dnight_> Hrm, so If I get it correctly this command `avconv -i http://192.168.1.167:12345 -f mpegts http://192.168.1.130:1234` expects a server to be listening on port 1234 for the incoming stream? [00:32:34 CEST] <m1dnight_> Because I get a connection refused on it. [00:32:46 CEST] <DHE> FlorianBd: youtube reprocesses videos, so if there's any playback issues youtube should fix it for the version actually played to users [00:33:13 CEST] <FlorianBd> DHE: right, they're quite advanced after all :) ok thanks [00:36:18 CEST] <furq> FlorianBd: in my experience youtube won't actually change the framerate [00:37:05 CEST] <furq> not from 6fps, anyway [00:37:16 CEST] <furq> i've uploaded a bunch of stuff at 5fps and it shows up at 6fps on youtube [00:37:45 CEST] <furq> i can't imagine it causing any playback issues though [00:42:02 CEST] <m1dnight_> Ah, I think Ive got something going, just can't see any video. I have avserver running with the default config (according to the docs) and I have, on the same machine, this running as well: avconv -i http://192.168.1.167:12345 -f mpegts http://localhost:8090/feed1.ffm [00:42:15 CEST] <m1dnight_> But VLC wont show me anything. When I try to connect I do see the GET commands etc on the logs. [00:44:30 CEST] <FlorianBd> furq: right, my point was to force them to use a low frame rate. Great then :) thanks [00:46:36 CEST] <furq> now if only there was some way to convince youtube to stop telling me "we detected your video may be shaky" about a static image [00:56:17 CEST] <m1dnight_> l/go info [01:01:10 CEST] <DHE> furq: ever actually said yes, deshake it? I did with a video game video. the new video made me a bit motion sick [01:02:41 CEST] <furq> i tried it once and it just zoomed it in a bit [01:02:50 CEST] <furq> and screwed up the white balance [01:03:38 CEST] <furq> https://www.youtube.com/watch?v=ahN7cXDX6oA [01:03:44 CEST] <furq> look at how shaky it is [01:07:56 CEST] <DHE> oh wow, I was expecting at least some scanline noise... yeah, good job youtube.. [01:08:11 CEST] <DHE> SOMETHING to "justify" the alert [01:08:30 CEST] <furq> i can at least understand it bugging me about bad cropping when i have a logo on a black background [01:08:40 CEST] <furq> but that one's just baffling [01:55:08 CEST] <m1dnight_> Another question.. I switched to ffmpeg/ffserver from avconv/avserver and now I keep getting the "503 server too busy" error from ffserver. [01:55:19 CEST] <m1dnight_> Ive tried all the google results from the first page but none really had a fix. [01:55:36 CEST] <m1dnight_> I have increased the tempfile size to 1g, moved it to my home directory and whatnot. Nothing seems to work. [01:56:22 CEST] <m1dnight_> And the stats page of ffserver shows that no bandwidth is in use at all.. [01:56:41 CEST] <m1dnight_> http://call-cc.be:8090/stat.html <- see there [02:05:06 CEST] <m1dnight_> Hrm, when I try to use mpegts it just gives a broken pipe error. [02:05:17 CEST] <m1dnight_> Ill give it another shot tomorrow :) [02:51:39 CEST] <Ben321> I notice FFMPEG seems to have a bug in it. I'm using the following command line with FFMPEG in order to set it up to stream raw BGRA format frames via pure TCP/IP (not HTTP) from a webcam. FFMPEG acts as a listening server, while another program I wrote is the TCP client. [02:52:10 CEST] <Ben321> ffmpeg -rtbufsize 5M -f dshow -framerate 5 -video_size 320x240 -i video="Philips SPC 900NC PC Camera" -f rawvideo -pix_fmt bgra -listen 1 tcp://127.0.0.1:1234 [02:54:59 CEST] <Ben321> But there's a problem with it. FFMPEG seems to feel the compulsive need serve the next piece of data, the INSTANT that the client software received the previous piece of data. Now my TCP client software receives the data as soon as the server makes it available. This causes a problem. Between frames that the webcam produces, there's no data for FFMPEG to serve. The result is that FFMPEG... [02:55:01 CEST] <Ben321> ...DUPLICATES the previous frame, in order to to keep up with the rule it tries to follow "as soon as the client has received the data, send new data". [02:55:57 CEST] <Ben321> How do I PREVENT FFMPEG from exhibiting this behavior? How to I keep FFMPEG synced with the webcam, so that it ONLY attempts to serve pixel data, when the webcam makes it available, and between frames sits idle waiting for the next frame from the webcam? [02:56:40 CEST] <Ben321> I am trying to AVOID having duplicate frames being sent over TCP. Please help. [03:05:56 CEST] <__jack__> Ben321: maybe -re [03:06:42 CEST] <__jack__> Ben321: not sure, didn't really understood your stuff :p [03:55:42 CEST] <lesderid> I'm having this strange error with initialising a codec context for PNG encoding: https://ghostbin.com/paste/k8muw [03:57:46 CEST] <lesderid> can't find much information about the error, but I've found code snippets online that are very similar to my code, so I'm not sure why mine isn't working correctly [05:52:46 CEST] <kzisme> I'm currently trying to take some photos and turn them into a video I keep getting "ould not find file with path DSC_%d.jpg and index in the range 1-5 - no such file or dir [05:53:03 CEST] <kzisme> I'm on windows and I opened a command prompt in the folder where I have all of my photos [05:53:31 CEST] <kzisme> Any idea how I could fix this? [05:58:15 CEST] <white> hi yes hello [05:58:32 CEST] <white> why on earth is this not working: [05:58:32 CEST] <white> ffmpeg -i stop.png -i stop.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.5:0.0[ckout];[0:v][ckout] overlay [out]" -map "[out]" -codec:a copy output.mp4 [05:58:37 CEST] <white> it does things like this: [05:58:47 CEST] <white> frame= 3 fps=2.2 q=-1.0 Lsize= 72kB time=00:00:00.04 bitrate=14821.8kbits/s dup=0 drop=96 [05:58:52 CEST] <white> it drops all the frames except for 3 [07:18:57 CEST] <dedondesta> hello [07:19:42 CEST] <dedondesta> anyone here? [07:22:29 CEST] <dedondesta> can anyone please help me converting this command: ffmpeg -f avfoundation -i ":0" -c:a libvo_aacenc -b:a 64k -ac 2 -y 1.wav to version of ffmpeg 3.0 and higher? [08:00:42 CEST] <carli> Hi, have one question, if someone get the same problem and know how to solve it. I stream to RTMP server, all works just fine, I only copy mp4 to flv stream. The problem is that sometime stream finises before the end of file? [08:02:20 CEST] <carli> Last time stream finished 20-30 seconds before the end of video, just droped. Are there any signals that I need to check, or it's maybe just a stream provider thing? [09:12:25 CEST] <dedondesta> can anyone please help me converting this command: ffmpeg -f avfoundation -i ":0" -c:a libvo_aacenc -b:a 64k -ac 2 -y 1.wav to version of ffmpeg 3.0 and higher? [09:16:15 CEST] <Fyr> guys, I need to plot bitrate (bitrate (Mbps)/time (s) plot). [09:16:21 CEST] <Fyr> what tool should I use? [09:16:29 CEST] <Fyr> can I use FFMPEG for that purpose? [09:17:42 CEST] <Fyr> right now I'm encoding a TS and it's gving output: XXXXXX kB time=1:00:00 bitrate = 500 kbps dup,drop,speed [09:18:02 CEST] <Fyr> I need this, but as a XY file. [09:19:03 CEST] <Fyr> something like the following: [09:19:03 CEST] <Fyr> http://smg.photobucket.com/user/Anamnesis/media/GINGER_SNAPS_2_zps47f20b2a.… [10:34:21 CEST] <m1dnight_> FINALLY GOT IT WORKING :) [10:34:27 CEST] <m1dnight_> Not its time to tweak :p [10:38:53 CEST] <dedondesta> can anyone please help me converting this command: ffmpeg -f avfoundation -i ":0" -c:a libvo_aacenc -b:a 64k -ac 2 -y 1.wav to version of ffmpeg 3.0 and higher? [10:39:03 CEST] <dedondesta> m1dnight_: can you help? [10:39:21 CEST] <dedondesta> in newer version they removed libvo_aacent [10:41:08 CEST] <m1dnight_> I have no idea, sorry [10:41:12 CEST] <m1dnight_> Im a total noob :D [10:41:22 CEST] <m1dnight_> What is the error you get..? [11:00:05 CEST] <dedondesta> m1dnight_: The native FFmpeg AAC encoder has been stable since FFmpeg 3.0. If you [11:00:08 CEST] <dedondesta> were using libvo-aacenc or libaacplus, both of which have been dropped in [11:00:10 CEST] <dedondesta> FFmpeg 3.0, please consider switching to the native encoder (-c:a aac), [11:00:13 CEST] <dedondesta> fdk-aac (-c:a libfdk_aac, ffmpeg needs to be installed with the [11:00:15 CEST] <dedondesta> --with-fdk-aac option), or faac (-c:a libfaac, ffmpeg needs to be [11:00:18 CEST] <dedondesta> installed with the --with-faac option). [11:06:34 CEST] <BtbN> so do what the output tells you. [11:06:49 CEST] <dedondesta> BtbN: i need wav as an output [11:07:15 CEST] <BtbN> and why are you trying to use an aac encoder then? [11:07:29 CEST] <dedondesta> am i? [11:07:33 CEST] <BtbN> yes. [11:07:48 CEST] <dedondesta> BtbN: thats my command ffmpeg -f avfoundation -i ":0" -c:a libvo_aacenc -b:a 64k -ac 2 -y 1.wav [11:14:14 CEST] <dedondesta> m1dnight_: the exact error is: Unknown encoder 'libvo_aacenc' [11:14:44 CEST] <dedondesta> it works in versions prior to 3.0 [11:15:10 CEST] <dedondesta> i need the same result but in ffmpeg 3.0 and higher [11:15:27 CEST] <dedondesta> it = this command [11:16:52 CEST] <dedondesta> BtbN: i'm a total noob in this so i'm not sure if i were using acc or not [11:17:26 CEST] <BtbN> this command never worked. And even if it didn't error, it certainly never did what you expected. [11:17:30 CEST] <dedondesta> I just know that encoder 'libvo_aacenc' is droped and i need to use something else [11:17:42 CEST] <BtbN> you don't need to use any encoder for wav. [11:17:47 CEST] <BtbN> It's uncompressed pcm. [11:20:46 CEST] <dedondesta> BtbN: So probably i just need to convert this command: ffmpeg -f alsa -ac 1 -ar 44100 -i hw:0 -t 30 out.wav to use Mac OS's avfoundation, right? [11:21:16 CEST] <BtbN> If that's what you want to do, yes. [11:22:28 CEST] <dedondesta> BtbN: something like: ffmpeg -f avfoundation -i ":0" -t 30 out.wav ? [11:22:43 CEST] <BtbN> No idea how avfoundation works, I don't use Macs. [11:31:38 CEST] <dedondesta> is it possible to set bitrate for this command? ffmpeg -f alsa -ac 1 -ar 44100 -i hw:0 -t 30 out.wav [11:33:00 CEST] <BtbN> wav is still uncompressed [11:36:33 CEST] <dedondesta> BtbN: something in this command: ffmpeg -f avfoundation -i ":0" -c:a libvo_aacenc -b:a 64k -ac 2 -y 1.wav made it so that 10 seconds of recording was only 84 kb and not 2 mb as with ffmpeg -f alsa -ac 1 -ar 44100 -i hw:0 -t 30 out.wav [11:36:49 CEST] <BtbN> yes, because you encoded it as aac.. [11:44:53 CEST] <dedondesta> BtbN: figured out, thanks for your help [12:10:53 CEST] <m1dnight_> My guess is the obvious codec that is misaing [12:11:15 CEST] <m1dnight_> Oh somebody is already on it. Nvm. [12:21:07 CEST] <flying_toaster> hi there [12:21:38 CEST] <flying_toaster> Do someone know here how I can use ffmpeg executable to pass options to the s302m decoder ? [12:22:10 CEST] <c_14> ffmpeg -h decoder=s302m [12:22:39 CEST] <flying_toaster> ok, sorry for aksing such a dumb question [12:22:42 CEST] <flying_toaster> thanks a lot [12:40:26 CEST] <jonascj> Does it make sense to reencode ACC mp4a to mp3? It is an audiobook I want to store, originally it comes as "Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 22050 Hz, stereo, fltp, 62 kb/s" [12:49:50 CEST] <ritsuka> I see no reason to reencode it [12:51:07 CEST] <jonascj> so just copy the audio if I want to split it into chapters? [12:51:44 CEST] <c_14> The only reason to reencode would be if whatever you're going to be playing it one doesn't support AAC (which these days won't be very much) [12:51:56 CEST] <c_14> So, yes [13:11:24 CEST] <c_14> *on [14:34:25 CEST] <lesderid> fwiw, I think I found what was wrong with my code. a mismatch between the libraries and the headers >.> [15:03:07 CEST] <Fyr> why does frame_bitrate equal to frame_size * frame_rate, while frame_size == pkt_size * 8 / 1000 ? [15:03:30 CEST] <Fyr> I don't understand actually pkt_size * 8 / 1000. [15:04:05 CEST] <BtbN> looks like bytes to kilobit to me. [15:04:22 CEST] <Fyr> and not 1024? [15:04:52 CEST] <furq> that's a kibibit [15:05:05 CEST] <Fyr> I thought if one needs to convert bytes into kbytes, he should divide it into 1024. [15:05:18 CEST] <furq> ffmpeg uses si prefixes [15:05:56 CEST] <iive> bitrate is always in power of 10 [15:07:13 CEST] <Fyr> ok, how to convert information from ffprobe output, pkt_pts_time and pkt_size into bitrate/time? [16:06:45 CEST] <Adrien2002> Hello there [16:11:35 CEST] <Adrien2002> I'm trying to set up a ffserver on a distant machine but when I stream to it, it suddently stops with a conversion failed. I have no idea why, I'm not video expert at all. I can paste server and client config file if you wish to check. I use the same script as the one I use for Twitch with address changed of course [17:57:28 CEST] <yagiza> Hello! [17:58:16 CEST] <Fyr> hey! [17:59:40 CEST] <yagiza> Can any1 tell me meaning of value, returned by swr_get_delay() routine? What should I do with it? [18:02:04 CEST] <pfelt> DHE: you around? [18:33:45 CEST] <Zelmoon> Hi, is the Sample_AES encryption possible to decrypt with ffmpeg yet? [18:35:11 CEST] <c_14> Don't think so [18:35:48 CEST] <Zelmoon> Ah that's pretty sad then. Got a few files I would like getting, but seems like the encryption is ruining that party then [19:20:57 CEST] <DHE> pfelt: ??? [19:27:42 CEST] <pfelt> DHE: had a question on your mpeg packet metering app if i could? [19:28:11 CEST] <DHE> sure [19:28:49 CEST] <pfelt> you read in the data and buffer it for output. are you doing that multithreaded so you can ensure you aren't blocking either end? [19:29:31 CEST] <pfelt> mine right now is just hard coded on rate and reads from a file at the right speed, but i'll need to modify it for live stream [19:32:32 CEST] <DHE> I ended up doing that, yes. just for ease of coding. one input thread, one output thread [19:34:34 CEST] <pfelt> ok. that's what i was headed towards too. simple mutex based fifor [19:34:51 CEST] <pfelt> cool. thx DHE ! [19:35:25 CEST] <DHE> heh. yeah I have a small collection of MPEG-TS applications [21:40:37 CEST] <fusevagio> why would I be getting `Unrecognized option 'tile-columns'.`? https://bpaste.net/show/bed3f74883fd [21:43:21 CEST] <wallbroken> hi [21:43:38 CEST] <wallbroken> how to rotate a .mov video without scaling it? in lossless mode? [21:44:50 CEST] <fusevagio> wallbroken: you're probably looking for the rotate filter combined with -codec:video copy [21:46:39 CEST] <fusevagio> but it's most likely going to fail because rotating without reencoding is possible only when using some particular codecs [21:46:44 CEST] <fusevagio> see this: http://stackoverflow.com/questions/25031557/rotate-mp4-videos-without-re-en… [21:51:53 CEST] <Fyr> guys, is there a big difference between AAC and AC-3? I've downloaded a torrent which name contains AC-3, but found out that mediainfo says that it has AAC instead. [21:52:04 CEST] <wallbroken> fusevagio, and if i want to scale, do i need to reencode? [21:52:37 CEST] <fusevagio> I can see libvpx listed in `ffmpeg -encoders` and also it was built with --enable-libvpx but it says libvpx-vp9 is not an recognized encoder, anyone's got a clue what's going on? [21:52:46 CEST] <fusevagio> wallbroken: probably [21:53:08 CEST] <fusevagio> well, actually I'm pretty sure you have to [21:53:16 CEST] <wallbroken> fusevagio, do you know why iphone does not allow to set resolution when recording videos? [21:53:34 CEST] <fusevagio> never had an iphone, sorry [21:54:21 CEST] <wallbroken> maybe the camera sensor gets the stream at the same costant resolution, and reducing it maybe will need an huge load on the CPU [21:55:34 CEST] <fusevagio> I would say that's unlikely, have you checked the settings thoroughly? [21:56:03 CEST] <fusevagio> I've got an android phone and it allows me to choose from a couple of different resolutions, iphones should be no different [21:56:26 CEST] <wallbroken> yes i just checked [21:56:28 CEST] <wallbroken> and is not possible [21:56:44 CEST] <wallbroken> i am not sure that i can rotate the video without reencode [21:56:50 CEST] <wallbroken> maybe i should just try? [21:56:51 CEST] <fusevagio> http://www.tech-recipes.com/rx/56843/change-video-recording-resolution-ipho… [21:57:09 CEST] <TD-Linux> Fyr, AAC is much better than AC3 [21:57:19 CEST] <fusevagio> first try only altering the rotation related metadata [21:57:30 CEST] <Fyr> TD-Linux, ok but is there a difference in byte sequence? [21:57:42 CEST] <TD-Linux> yes they are totally different f ormats [21:58:20 CEST] <Fyr> alright [21:58:51 CEST] <furq> fusevagio: it's possible to build libvpx without vp9 support, so maybe that's happened [21:59:09 CEST] <furq> i seem to remember ffmpeg not compiling when i tried to disable vp8 in libvpx though [21:59:17 CEST] <TD-Linux> also the ABI broke again recently [21:59:36 CEST] <TD-Linux> make sure you have libvpx 1.5.0 at least [21:59:59 CEST] <wallbroken> fusevagio, thank you [22:00:09 CEST] <fusevagio> furq: that seems to be the case, I'm getting these builds at Zeranoe and a different build from March has libvpx-vp9 [22:00:22 CEST] <fusevagio> wallbroken: but have you succeeded? [22:00:26 CEST] <furq> weird [22:00:39 CEST] <furq> i can't imagine why he'd disable vp9 [22:00:41 CEST] <furq> not on purpose, anyway [22:00:50 CEST] <wallbroken> fusevagio, yes, i solved, but right now i need to rotate the past videos [22:00:59 CEST] <wallbroken> withut scaling [22:01:08 CEST] <wallbroken> the only thing i need is without reencoding [22:02:01 CEST] <Fyr> guys, when uploading video onto Youtube, does it always re-encode? or it can keep if it's already encoded h264/aac? [22:02:08 CEST] <furq> it always reencodes [22:02:11 CEST] <fusevagio> Fyr: always reencode [22:02:18 CEST] <Fyr> =( [22:02:23 CEST] <fusevagio> this is why I always upload my audio as flac [22:02:30 CEST] <Fyr> fusevagio, is there a reason to? [22:02:37 CEST] <furq> youtube didn't support flac last time i tried it [22:02:38 CEST] <fusevagio> but won't be going for lossless video because the bandwidth won't allow me [22:02:42 CEST] <furq> that was ~4 years ago though [22:03:05 CEST] <fusevagio> Fyr: I mostly do that to keep the audio quality as good as possible in the future [22:03:23 CEST] <fusevagio> youtube can produce new encodes of old clips [22:03:42 CEST] <furq> yeah i believe they keep the original source clip around [22:07:05 CEST] <fusevagio> in case anyone needs this: lowest bitrate for libopus in cbr mode is 2533.5 bits per second and in the vbr and cvbr modes it's 2400 bits per second [22:07:19 CEST] <fusevagio> so you don't have to search yourselves [22:14:32 CEST] <wallbroken> can you suggest to me the line to use to rotate of 90 degrees without reencoding ? [22:18:28 CEST] <furq> without reencoding, use -c copy -metadata:s:v:0 rotate=90 [22:19:10 CEST] <furq> if you want to reencode then use -vf transpose=1 [22:19:22 CEST] <furq> or -vf transpose=0 to rotate 90 degrees anticlockwise [22:19:51 CEST] <furq> changing the metadata is only supported in certain containers and by certain players, so you might need to reencode [22:20:23 CEST] <furq> er, that should be -vf transpose=2 to rotate 90 degrees anticlockwise [22:24:32 CEST] <wallbroken> no [22:24:48 CEST] <wallbroken> i haven't asked how to rotate in metadata [22:24:55 CEST] <wallbroken> i need to rotate the entire video [22:25:03 CEST] <furq> that is how you rotate the video without reencoding [22:25:36 CEST] <furq> you set a flag in the metadata which tells the player to flip the video during playback [22:25:42 CEST] <wallbroken> changing the metadata flag does not mean rotate the video [22:25:54 CEST] <furq> then you need to reencode [22:26:01 CEST] <wallbroken> no! [22:26:15 CEST] <wallbroken> have you ever used a jpeg? [22:26:19 CEST] <wallbroken> you can rotate it without reencoding [22:26:26 CEST] <wallbroken> and not using the metadata flag [22:26:52 CEST] <wallbroken> a jpeg is composed by standard squared [22:27:01 CEST] <wallbroken> you can change their position [22:27:08 CEST] <wallbroken> without reencoding [22:27:11 CEST] <TD-Linux> that's not possible for most video formats due to intra prediction [22:27:28 CEST] <furq> well that's great if you have an mjpeg stream [22:27:33 CEST] <furq> it's not possible for most other codecs though [22:27:39 CEST] <wallbroken> ok [22:27:48 CEST] <wallbroken> so the only way is use metadata? [22:28:02 CEST] <furq> i'm not sure if that's supported by mov, you might need to remux to mp4 [22:28:16 CEST] <wallbroken> that's what? [22:28:17 CEST] <wallbroken> metadata? [22:28:19 CEST] <furq> yeah [22:28:24 CEST] <wallbroken> yes is supported [22:28:31 CEST] <wallbroken> but vlc fails reading it [22:28:37 CEST] <wallbroken> video is stretched [22:28:42 CEST] <wallbroken> and not rotated [22:28:53 CEST] <furq> did you record this off a phone [22:29:07 CEST] <furq> it might already have the rotate flag set, in which case you just want to remove it [22:29:26 CEST] <wallbroken> yes, the phone sets it automatically [22:29:33 CEST] <wallbroken> but vlc does not use it well [22:29:39 CEST] <furq> try -metadata:s:v:0 rotate=0 [22:29:45 CEST] <Mavrik> VLC indeed doesn't listen to rotation flag. [22:29:48 CEST] <furq> oh really [22:29:52 CEST] <furq> never mind then [22:29:53 CEST] <Mavrik> Which messes up all videos from phones. [22:30:06 CEST] <Mavrik> We had that issue as well, had to reencode -_- [22:30:13 CEST] <furq> how dumb [22:30:24 CEST] <furq> i was going to say "i expect better from vlc" but that would be a lie [22:30:49 CEST] <wallbroken> Mavrik, looks like it reads the rotation flag [22:30:55 CEST] <wallbroken> but it uses it wrongly [22:54:41 CEST] <vade> can someone describe to me when and why AV_TIME_BASE and AV_TIME_BASE_Q would be used rather than a existing time base of a codec or stream ? Im trying to parse some of the timestamp paths and conversions from stream to codec to stream in FFMPEG :) [23:01:11 CEST] <vade> ah. I see [23:01:28 CEST] <vade> its a high resolution time base one can use to sync audio and video or whatever streams [23:01:49 CEST] <JEEB> was this enlightenment brought by the documentation or reading the fabulous(?) code? [23:02:10 CEST] <vade> haha. Stack Overflow. I didnt see it in the docs [23:06:32 CEST] <vade> honestly JEEB I have a hard time following all of the pts/dts and duration branches and why they occur in FFMPEG.c, which is why I was lost on its use [23:06:49 CEST] <JEEB> ffmpeg.c is a mess [23:07:00 CEST] <vade> I didnt want to say it, but, yea [23:07:18 CEST] <JEEB> like, libavformat supports dynamic finding of streams [23:07:20 CEST] <vade> I mean, its handling a hell of a lot of edge cases [23:07:26 CEST] <JEEB> what does ffmpeg.c do? [23:07:33 CEST] <JEEB> "lemme skip this and just log it" [23:08:04 CEST] <JEEB> yes and it also contains a heckload of hacks because people didn't have time or didn't want to properly implement something [23:08:17 CEST] <vade> is there a better, or shall I say, more modern best practice of time base adjustments from demux, decode, encode and mux ? [23:08:37 CEST] <vade> yea, its tricky with Open Source / time management and also supporting formats that inherently make different assumptions [23:10:21 CEST] <JEEB> usually after decoding you have the correct timestamps for each input packet, container timestamps are DTS-ordered (for obvious reasons) [23:11:05 CEST] <JEEB> with b-frames there's a reordering that happens during the decoding process [23:12:50 CEST] <JEEB> also this reminds me that some demuxers fuck up timestamps and expect you to decode things... ugh my head hurts [23:13:09 CEST] <vade> yea. I totally get that. My use case right now is, im on an amazon G2 instance, I render some stuff / synthesize some frames in GL and send them to an encoder. I also might decode a selection of frames after doing some CV detection for black frames, so I only decode a time-range of one source [23:13:30 CEST] <vade> and then all of that gets sent to an encoder that needs to deduce how to mux it (and also sync audio) [23:13:36 CEST] <vade> fortunately im only MP4, not any muxer [23:14:08 CEST] <JEEB> muxers generally handle timestamps right because you're feeding them [23:14:10 CEST] <vade> but its still tricky for me since depending on frame rate, I get wildly different time bases for decode vs encode and in the streams and codecs too. I also have to do some normalization, etc. [23:14:27 CEST] <vade> so you just feed them PTS monotonically and dont worry about duration? [23:14:47 CEST] <JEEB> if you have a duration available that can be set I guess [23:15:00 CEST] <JEEB> I would guess it then fills the duration field in the container from the different PTS values [23:15:19 CEST] <JEEB> (you can use L-SMASH's boxdumper to dump a text version of what's in the container) [23:15:21 CEST] <vade> well, this is for broadcast, so I need to ensure frame boundaries are exact, and that I dont mess up durations :X [23:15:39 CEST] <vade> yea, I think you mentioned that to me previously. I have it here [23:16:26 CEST] <JEEB> I think martin lately worked on improving various parts of movenc although it could be used before as well [23:16:48 CEST] <JEEB> although he's usually pushing things to libav and then they generally got merged into FFmpeg with the usual merges [23:17:15 CEST] <JEEB> that said, FFmpeg is so awesome that some people decide to bully people out of merging because they dislike merging Libav things [23:17:29 CEST] <JEEB> so I'm not sure if those improvements will ever land in FFmpeg :P [23:18:51 CEST] <vade> :X [23:18:52 CEST] Action: vade sighs [23:19:26 CEST] <vade> Strangly this makes me appreciate AVFoundation even more, for all its lacking, it just works and is super clean and conceptually organized well [23:19:38 CEST] <vade> *strangely [23:21:47 CEST] <JEEB> I've seen DirectShow and I just want to get as far from it as possible [23:22:08 CEST] <vade> hahah. Ive not had the joy yet, but a colleague has said similar things [00:00:00 CEST] --- Tue May 24 2016

1 0

ffmpeg-devel.log.20160522
by burek 23 May '16

23 May '16

[00:17:45 CEST] <Illya> michaelni: does it break the build with patch #1 as well? [00:18:19 CEST] <Illya> SDL patch, I mean. [00:29:47 CEST] <michaelni> Illya, just tried, fails too (see ML) [00:29:59 CEST] <michaelni> i mean patch 1+2 [00:30:14 CEST] <michaelni> ill try the older patch in a moment [00:32:31 CEST] <Illya> if they still dont work: https://gist.github.com/c1b5dbd2882f5580e17dae6f9269612f this should, and if this doesnt, then there's something very wrong (probably on my end, though) [00:37:31 CEST] <michaelni> Illya, just to be sure this isnt missed, this is a ubuntu 12.04 LTS i think (aka a somewhat older OS) [00:38:57 CEST] <Illya> also make sure that you have sdl2 installed (you probably do, just checking). I mean.... it should work, SDL is pretty cross platform [00:42:02 CEST] <michaelni> Illya, i doubt the box has sdl2 but build of plain ./configure broke but then maybe i installed sdl2 at some point and forgot [00:44:00 CEST] <Illya> I didn't see a configure error in your log? Was this afterwards? [00:46:11 CEST] <michaelni> theres no error from configure [00:47:33 CEST] <michaelni> theres also no sdl2-config [00:48:17 CEST] <michaelni> !HAVE_SDL2=yes [00:48:24 CEST] <michaelni> CONFIG_SDL2_OUTDEV=yes [00:48:37 CEST] <michaelni> thats a odd combination [00:49:00 CEST] <Illya> err. oh, yes i see the issue now [00:49:17 CEST] <Illya> I uninstalled sdl2 and sdl2 outdev is still enabled [00:52:33 CEST] <Illya> michaelni: ok, fixing that now. And, with sdl2 installed, how does it do? [00:53:02 CEST] <michaelni> i dont think 12.04 LTS has sdl2 [00:53:17 CEST] <JEEB> it's old enough to barely be in support now [00:53:25 CEST] <michaelni> i could test on another box of course [00:53:26 CEST] <JEEB> will be out next year [00:53:41 CEST] <JEEB> so yeah, you'd have to build SDL2 into another prefix [00:53:43 CEST] <JEEB> to test it [00:59:17 CEST] <Illya> michaelni: this is the updated patch, which doesnt break configure (hopefully) anymore. https://gist.github.com/adc405000493a9f26e568ea3ae0468ef I wasn't checking for the sdl2 outdev's dependencies, must have missed that earlier [01:06:39 CEST] <michaelni> Illya, on old ubuntu ffplay isnt build anymore build seems working thogh for the rest, on new ubuntu ffplay doesnt work, doesnt recognize "q" for example [01:06:42 CEST] <michaelni> "The key you just pressed is not recognized by SDL. To help get this fixed, please report this to the SDL mailing list <sdl(a)libsdl.org> X11 KeyCode 24 (16), X11 KeySym 0x71 (q)." [01:09:53 CEST] <Illya> FFplay won't build without sdl2 since it's been converted from sdl to sdl2, and since you dont have sdl2 on the old ubuntu it cant build, and the first patch is actually Marton's, but mine just requires it (hence me bundling it with mine all the time) so I haven't really delved into ffplay much [01:47:29 CEST] <michaelni> would it be difficult/messy to keep SDL1 support until distros with only SDL1 are dead? [02:18:23 CEST] <Illya> michaelni: probably [02:19:44 CEST] <Illya> For ffplay, most definitely, the patch looks like it changes quite a lot. For the outdevs, not at all. [02:32:43 CEST] <llogan> andrey_utkin: approved. you can ping me next time, but it may take me a while as I was afk for 7 hours. [02:46:15 CEST] <cone-279> ffmpeg 03Michael Niedermayer 07master:ea88dc3e8e2c: avformat/avienc: Rename reshuffles return variable to ensure it is not mixed up [02:46:15 CEST] <cone-279> ffmpeg 03Michael Niedermayer 07master:9d5894b2399e: avformat/avienc: Fix memleaks on errors [02:53:19 CEST] <Illya> michaelni: not sure how to proceed with this. I guess I could still make a patch to add the sdl2 outdev independently to ffplay, idk. Although, it's not like it 100% works either, there is still one major bug which I'm banging my head against the wall about. [03:02:09 CEST] <michaelni> Illya, iam just speaking for myself but wouldnt it be trivial to keep ffplay_sdl1.c ? (though thats something marton would need to decide if he wants that) [03:02:46 CEST] <Illya> oh. well... you could do that, I thought you mean keeping both in the same file, using macros to compile each one [03:03:39 CEST] <michaelni> also either way if its easy to make dev and ffplay more independant that probably doest hurt [03:04:57 CEST] <michaelni> why are they dependant ? [03:05:10 CEST] <Illya> I can submit the dev as a separate thing, but it would require a patch for configure which would suit ffplay as well, although I think I could just squash the configure commits and then remove the ffplay parts [03:05:38 CEST] <michaelni> so its just reordering a few hunks ? if so that seems like it should be done anyway [03:05:50 CEST] <michaelni> reordring between commits that is [03:06:15 CEST] <Illya> michaelni: it was mostly just for keeping things together, I think that Marton didn't want to submit the ffplay patch until there was an sdl2 outdev. [03:06:47 CEST] <Illya> I mean, his sdl2 ffplay patch has been around for a while (2014 iirc) [03:07:42 CEST] <michaelni> i was just thinking why its not 1 patch for build sys and 1 for outdev and one for ffplay [03:08:29 CEST] <Illya> not sure, but it would make sense to do it like that [03:52:27 CEST] <cone-279> ffmpeg 03Michael Niedermayer 07master:2062d51275bd: avformat/matroskaenc: wrap par->format use as sample format under codec_type == AVMEDIA_TYPE_AUDI [03:52:28 CEST] <cone-279> ffmpeg 03Michael Niedermayer 07master:566be4f9e886: avformat/matroskaenc: reindent after last commit [04:32:31 CEST] <rcombs> AVMEDIA_TYPE_AUDI, not to be confused with AVMEDIA_TYPE_CHVY or AVMEDIA_TYPE_TYTA [10:32:20 CEST] <andrey_utkin> thanks for approving email, llogan. [11:58:40 CEST] <j-b> 'morning [18:13:30 CEST] <lilibox> hi [18:15:04 CEST] <Compn> helllloooo [18:15:23 CEST] <Compn> lilibox : we may not be here immediately to take your call, but we will see it later [18:15:33 CEST] <Compn> so please ask your question and stick around for answer [20:33:49 CEST] <Shiz> lol [20:34:39 CEST] <Shiz> why did cehoyos edit the link out [20:50:49 CEST] <nevcairiel> you mean the automatic wiki link? why should that be there? [20:51:05 CEST] <nevcairiel> it links any camelcase word, which is kinda stupid [20:51:14 CEST] <Shiz> no [20:51:17 CEST] <Shiz> he edited it twice [20:51:28 CEST] <Shiz> there was a link to the MPC-BE source on sourceforge first [20:55:26 CEST] <durandal_1707> Shiz: mpc-be supports that files? [20:57:12 CEST] <JEEB> it has a hacked gabest's mpeg-ps demuxer so it might support it [20:57:21 CEST] <JEEB> GPL code of course, so lol [21:02:01 CEST] <kierank> gabest [21:02:05 CEST] <kierank> haven't heard that name for a while [21:02:12 CEST] <JEEB> he pops up in random places [21:02:21 CEST] <nevcairiel> its been like 10 years since he was active in multimedia [21:02:24 CEST] <JEEB> I think a few years back he posted on Avery Lee's blog [21:02:48 CEST] <JEEB> he went to opengl programming with emulators after leaving guliverkli [21:03:03 CEST] <JEEB> and then... he just floats on the internets [21:06:47 CEST] <Shiz> durandal_1707: yes [22:59:29 CEST] <cone-354> ffmpeg 03Paul B Mahol 07master:5ce1d089f01a: avcodec/mvcdec: simplify, no need to use reget buffer [23:06:35 CEST] <cone-354> ffmpeg 03Piotr Bandurski 07master:854c405dab23: avcodec/iff: mention RGB8/RGBN decoder [23:34:49 CEST] <cone-354> ffmpeg 03Paul B Mahol 07master:cd506c279229: avcodec/truemotion2rt: also export key frame info [00:00:00 CEST] --- Mon May 23 2016

1 0

ffmpeg.log.20160522
by burek 23 May '16

23 May '16

[01:58:39 CEST] <kynlem> so, i've got a mov file (h264 video; pcm audio) where each 12th frame is a key frame [01:59:49 CEST] <kynlem> if i reencode just the video part and cut 60.06 seconds (1440 frames) from it, then everything's alright [02:00:38 CEST] <kynlem> but if i reencode audio as well, and then merge the produced 60.06-second chunks, there's a glitch in sound *and* video where the pieces meet. thoughts? [04:53:36 CEST] <hendry> I know one can make a slideshow like so: ffmpeg -framerate 1 -pattern_type glob -i '*.jpg' -c:v libx264 out.mp4 [04:53:50 CEST] <hendry> but I want the filenames of the jpgs in the video itself... how do I achieve that? [07:48:35 CEST] <GimmeSpace> hello sir, how to watermarking videos with ffmpeg for specific frames?? [07:49:36 CEST] <GimmeSpace> like it just up to 10 sec and dissapear for 10 sec and up again for another 10 sec like so... thanks. [07:54:48 CEST] <GimmeSpace> hello guys... [08:13:11 CEST] <twistin> Hello. Can anyone help me use ffmpeg to combine a DashVideo (~489MB) and Dashaudio(~88MB) file fast? The command I am using now is `ffmpeg -i 1.dashAudio -i 2.dashVideo -shortest output.mp4` ... The length of the video is 3:12:29 (hr:min:sec) and same length audio. It has been nearly 2 hours and ffmpeg is still running. [10:53:49 CEST] <cyzz> hi. its a pleasure to play round with your work [10:55:32 CEST] <cyzz> twitch over ffmpeg@nvenc but..... :D it works [13:01:44 CEST] <pandaologist> having some trouble getting ffplay to recognise a rawvideo stream piped from ffmpeg [13:02:33 CEST] <pandaologist> it is NTSC yuv422p, can anyone advise what the ffplay parameters should be? [13:03:08 CEST] <pandaologist> all I get at the moment, no matter what I try is [13:03:11 CEST] <pandaologist> "Invalid data found when processing input" [13:04:01 CEST] <pandaologist> it used to just work until I updated ffmpeg [13:17:59 CEST] <pandaologist> no worries, seems to be working now [13:18:07 CEST] <pandaologist> not even sure what i'm doing differently [17:46:07 CEST] <lilibox> hi [17:48:06 CEST] <ChocolateArmpits> hello [17:52:27 CEST] <lilibox> i would like to know what -f switch is stand for, i have got some complicated .php which calls ffmpeg and uses switch -f sb16le [17:52:53 CEST] <furq> format [17:53:10 CEST] <lilibox> i wanna make static ffmpeg, so which lilbraries add to my build will be able handle sb16le [17:53:14 CEST] <furq> you use it when the format can't be inferred from the filename [17:53:35 CEST] <furq> s16le is pcm, it's built-in [17:53:57 CEST] <lilibox> interesting [17:59:07 CEST] <lilibox> and what is main difference between ffmpeg and ffmpeg_flv? [17:59:55 CEST] <furq> i've never heard of ffmpeg_flv and neither has google [18:00:02 CEST] <furq> so i'm going to assume someone's renamed an ffmpeg binary [18:08:29 CEST] <DHE> sounds like someone's custom build with a narrow purpose [18:08:45 CEST] <DHE> I have a few of those, but I keep them private. confuses people otherwise [18:09:02 CEST] <lilibox> yes, it's Autodesk's chaos [18:19:47 CEST] <ploop> I'm converting videos with the "-target ntsc-dvd" option, but every dvd creating utility I've tried still tries to reconvert them again which destroys the quality. are there any utilities that won't do such a thing if the video is dvd-compliant? [18:32:21 CEST] <c_14> dvdauthor should do it, no? [18:35:04 CEST] <grrk-bzzt> Hello [18:36:18 CEST] <grrk-bzzt> I'm trying to convert one lossless video file to another lossless video codec without color loss [18:36:35 CEST] <c_14> use the same pixel format [18:36:52 CEST] <grrk-bzzt> But the pixel format is pal8 [18:37:31 CEST] <c_14> or something that can accurately transform like zscale or that new filter, whatever it was called [18:38:15 CEST] <c_14> The pixel format of what? The input? Output? [18:39:18 CEST] <grrk-bzzt> The input file uses the zmbv video codec and the pal8 pixel format [18:40:34 CEST] <c_14> Extract a frame from the input (using -c copy or something) convert it to the other codec/pixel format and convert it back and check if they're the same [18:40:54 CEST] <c_14> Doesn't have to be a frame, can be a section or something as well [18:41:38 CEST] <grrk-bzzt> I'm using ffmpeg -i a.avi -i b.avi -filter_complex "blend=all_mode=difference" so far to see if there's any differences [18:41:48 CEST] <c_14> See if ffmpeg does the right thing by default, or if you need to specify a different destination pixel format (higher bit depth or something) or use zscale or the colorspace filter [18:42:19 CEST] <c_14> use -fflags +bitexact and just check the sha hash, or use the framecrc muxer or something [18:42:30 CEST] <grrk-bzzt> Alright [18:42:40 CEST] <c_14> there's also the hash/framehash muxers and md5/framemd5 [18:43:45 CEST] <grrk-bzzt> Thank you for your advices c_14 [19:12:51 CEST] <grrk-bzzt> Seems impossible to obtain the same hash [19:13:10 CEST] <c_14> With what command[s]? [19:13:38 CEST] <grrk-bzzt> c_14, I'm using -flags +bitexact -sws_flags +accurate_rnd+bitexact -fflags +bitexact [19:14:12 CEST] <c_14> what output codec/pixel format? [19:15:02 CEST] <grrk-bzzt> I've tried rawvideo output codec, and so far I've tried with rgb24, pal8 and yuv420p [19:24:07 CEST] <c_14> zmbv (pal8) -> ffv1 (rgba) -> zmbv (pal8) [19:24:09 CEST] <c_14> seems to work here [19:24:15 CEST] <c_14> Checked with -f framemd5 and -f md5 [19:25:00 CEST] <grrk-bzzt> OK, I'll check that c_14 [19:25:18 CEST] <c_14> -f hash -hash sha512 also matches [19:25:30 CEST] <grrk-bzzt> Did your commande included the same -flags -sws_flags and -fflags I've used? [19:25:38 CEST] <c_14> I didn't use the sws_flags [19:25:55 CEST] <c_14> they probably shouldn't hurt though [19:26:07 CEST] <grrk-bzzt> just ffmpeg -i a.avi -c:v ffv1 out.avi ? [19:28:14 CEST] <c_14> -pix_fmt bgra [19:28:23 CEST] <c_14> it chooses bgr0 by default [19:28:34 CEST] <c_14> which might and might not work since pal8 has alpha and bgr0 doesn't [19:28:39 CEST] <c_14> depends on if your source has alpha [19:28:41 CEST] <grrk-bzzt> c_14, you've said rgba earlier [19:28:52 CEST] <grrk-bzzt> No, my source doesn't have alpha [19:28:55 CEST] <c_14> I meant bgra [19:29:04 CEST] <c_14> ffv1 doesn't even support rgba [19:29:08 CEST] <grrk-bzzt> Alright [19:29:26 CEST] <grrk-bzzt> So it'll ignore alpha info? [19:31:11 CEST] <c_14> if you use bgr0, yes [19:31:39 CEST] <c_14> Hashes seem to still match on the video I tested on (which doesn't have alpha) so it should be fine [19:38:26 CEST] <grrk-bzzt> Hum, it doesn't here [19:42:46 CEST] <grrk-bzzt> ffv1 (bgra) -> zmbv (pal8) doesn't work, and it actually degrades the video quality [19:43:05 CEST] <grrk-bzzt> Hum, I'll try with the *flags [19:44:05 CEST] <c_14> https://pb.c-14.de/t/kng.1rTFdm <- if you run that, does it output a diff? [19:44:17 CEST] <c_14> (I'm trying to check if it's something with your video or with your ffmpeg version) [19:47:52 CEST] <grrk-bzzt> c_14, it's doesn't work: Requested output format 'hash' is not a suitable output format in: Invalid argument [19:48:04 CEST] <c_14> Your version of ffmpeg is probably rather old [19:48:07 CEST] <c_14> What version are you running? [19:48:22 CEST] <grrk-bzzt> ffmpeg version 3.0.2 [19:48:37 CEST] <c_14> Ah, that particular change might be only in git? [19:49:26 CEST] <c_14> try that https://pb.c-14.de/t/kng.TMm73v [19:49:42 CEST] <c_14> md5 should be good enough [19:51:33 CEST] <grrk-bzzt> Alright? What should I see? [19:51:52 CEST] <c_14> if you don't see anything it's good. [19:51:57 CEST] <c_14> If it didn't match, you'd see a diff at the end [19:52:03 CEST] <grrk-bzzt> So it's good [19:52:09 CEST] <c_14> you can check in/out and in~ and out~ manually if you want to make sure [19:52:17 CEST] <grrk-bzzt> I did [19:52:19 CEST] <c_14> So it's something with your video probably [19:52:26 CEST] <grrk-bzzt> alright [19:52:32 CEST] <grrk-bzzt> I can send you the sample [19:52:35 CEST] <c_14> maybe it does have alpha? [19:52:46 CEST] <grrk-bzzt> Just a moment [20:02:11 CEST] <c_14> The output looks like it has had dithering applied to it [20:02:25 CEST] <grrk-bzzt> Indeed [20:07:16 CEST] <c_14> the ffv1 output isn't dithered though [20:22:04 CEST] <grrk-bzzt> sorry c_14 I had a network problem [20:22:19 CEST] <grrk-bzzt> I did not receive anything past "the ffv1 output isn't dithered though" [20:22:32 CEST] <c_14> That was the last message I sent. [20:25:05 CEST] <c_14> I can't seem to turn off the dithering during the zbmv conversion, even -sws_dither none doesn't help [20:25:44 CEST] <c_14> *zmbv [20:26:43 CEST] <grrk-bzzt> Is it a ffmpeg bug? [20:27:58 CEST] <c_14> Either a bug, or a deficiency in the zmbv encoder [20:30:39 CEST] <grrk-bzzt> Alright [20:30:50 CEST] <grrk-bzzt> Good to know i've spotted something then [20:31:59 CEST] <grrk-bzzt> To be honnest, I wanted to run some test to see which lossless video codecs to see which one offered the best compression ratio for retrogaming recording [20:32:16 CEST] <grrk-bzzt> But it seems more complicated than that [20:34:54 CEST] <c_14> Well, if it doesn't have to be 100% perfectly lossless (minor loss due to chroma conversion/dithering) it's a lot easier [20:36:12 CEST] <JEEB> well, ffv1 and libx264 both support RGB coding [20:36:29 CEST] <JEEB> so if you want 1:1 lossless to the original capture that's possible [20:42:46 CEST] <Prelude2004c> hey guys.. good day [20:43:55 CEST] <Prelude2004c> i am using ffmpeg to convert somethign into HLS.. and i am using nfs mount to basically drop the files in a remote server.. i may be having some troulbe with that nfs mount for some reason ( not sure why ).. i would like to use webdav with some queue system.. anyone have any idea what i can use to send the files via webdav and then if lets say it doens't respond, it can queue files for say 30 seconds.. and at some point when the destinat [20:43:55 CEST] <Prelude2004c> ion comes bcak online it dumps them all at the same time ? [20:53:40 CEST] <BtbN> why not just run ffmpeg on the target machine? [20:55:46 CEST] <DHE> if it's a live stream then you have a user experience problem anyway. otherwise I would just save locally and rsync either periodically or when it's done [21:24:01 CEST] <grrk-bzzt> <JEEB> so if you want 1:1 lossless to the original capture that's possible How? [21:26:30 CEST] <JEEB> grrk-bzzt: pick ffv1 or libx264_rgb and encode with it. ffv1 is lossless by definition, libx264 has to have -q:v 0 for lossless [21:28:21 CEST] <grrk-bzzt> Which pix_fmt should I use? [21:29:34 CEST] <JEEB> grrk-bzzt: by default you should get the input pix_fmt [21:29:41 CEST] <JEEB> see ffmpeg cli's output [21:43:20 CEST] <erraunt> Hi. I was reading some of the jpeg2000 code and TODO with a seen references to ISO papers, but the document costs 175 CHF, are there any other ways to read it? [21:48:47 CEST] <JEEB> google is generally your friend with such things [22:12:13 CEST] <Prelude2004c> i can't sync.. its a live stream [22:12:16 CEST] <Prelude2004c> has to be done in realtime [22:13:16 CEST] <furq> what's wrong with nfs [22:14:10 CEST] <Fyr> guys, is -filter:v a new option? [22:14:17 CEST] <furq> no [22:14:18 CEST] <Fyr> while -vf - old? [22:14:21 CEST] <furq> oh [22:14:27 CEST] <furq> well yeah it's what you should use in place of -vf [22:14:45 CEST] <furq> i assume the old switches will be deprecated at some point but probably not any time soon [22:15:04 CEST] <Fyr> so, filter:v, so as c:a and other syntax with ":" was introduced recently? [22:15:11 CEST] <furq> comparatively recently [22:15:16 CEST] <Fyr> ok [22:17:51 CEST] Action: penguin42 is a bit confused - I'm converting from an rtsp stream off a cctv camera that I believe is h264 to mp4 (using -preset veryfast -f segment) and I'm seeing with perf top that ffmpeg is spending a lot of time in idct - why? If it's just repackaging why does it need to idct it? [22:18:17 CEST] <furq> because you're reencoding it [22:18:36 CEST] <furq> use -c:v copy if you just want to remux it [22:18:41 CEST] <furq> and also -preset is meaningless if you're remuxing [22:19:25 CEST] Action: penguin42 tries -cv copy - - would it be otherwise the same line? [22:19:46 CEST] <furq> that depends on what the line is [22:19:48 CEST] <furq> but probably [22:20:53 CEST] <penguin42> http://pastebin.com/VTjWSH5A is my current line - lets try that option [22:24:45 CEST] <penguin42> ooh, promising [22:34:47 CEST] <penguin42> furq: OK, that looks pretty good, except for some reason it's changed where segment has decided to segment the video [22:35:01 CEST] <furq> segments have to start on a keyframe [22:35:26 CEST] <furq> obviously if you're just remuxing then you have no control over keyframe placement [22:36:00 CEST] <penguin42> furq: Right, my problem here is the opposite - it seems to be segmenting every second, where as with my world without the -c:v copy it was segmenting about once every minute [22:36:55 CEST] <furq> you can set -segment_time to override that [22:37:00 CEST] <penguin42> ah thanks [22:54:09 CEST] <penguin42> furq: Thank you - that seems to be doing the trick; and I now have two pretty much idle cores :-) [22:54:40 CEST] <penguin42> poor little ULV celerons dont have very fast brains [23:52:55 CEST] <tlvb> hello, I'm examining a script for converting a list of mts files into another format. The script creates a fifo and cats the mts files into that, in the background, the video camera may create multiple mts files even when filming a single event, and apparently cat-ing them together is entirely reasonable, however, I wonder if feeding this fifo file to ffmpeg, as the script does, is any different from using [23:52:58 CEST] <tlvb> the concat demuxer? [23:56:29 CEST] <tlvb> the original script can be found here: http://ubuntuforums.org/showthread.php?t=1630612 , and mine, although changed to convert to a different format as well, here: http://pastebin.com/JyYWZUdW [00:00:00 CEST] --- Mon May 23 2016

1 0

ffmpeg-devel.log.20160521
by burek 22 May '16

22 May '16

[01:13:12 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vrgcd [01:13:12 CEST] <KGB> 13FFV1/06master 1423a953b 15Jérôme Martinez: slice_width and slice_height definition update... [01:18:39 CEST] <Shiz> lol [01:19:11 CEST] <Shiz> can someone get rid of that spam above ^ [01:36:04 CEST] <Compn> i think michaelni needs to give me that power on the trac as well [01:36:10 CEST] <Compn> because i dont have the ability iirc [02:37:11 CEST] <michaelni> Compn, you have wiki delete powers [02:37:17 CEST] <michaelni> now that is [02:38:18 CEST] <michaelni> deleted the page [02:43:07 CEST] <michaelni> marked as spam and user deleted [02:47:52 CEST] <Shiz> \o [04:01:44 CEST] <cone-040> ffmpeg 03foo86 07master:801dbf0269b1: avcodec/dca: simplify 'residual ok' flag tracking [04:14:31 CEST] <cone-040> ffmpeg 03foo86 07master:64fe1eebddbb: avcodec/dca: use LUT for LBR frequency ranges [07:43:56 CEST] <slashdevnull> What is the recommended method for implementing the following change? For the PPC port, replace all the functions in libswscale/input.c with versions that use vector arithmetic hardware. [07:46:03 CEST] <slashdevnull> What I presume is that #ifdef directives in each function cannot be the preferred technique for customizing an entire source file for a particular processor. [11:48:23 CEST] <durandal_1707> review for aic, please [14:42:29 CEST] <cone-443> ffmpeg 03ZhouXiaoyong 07master:a3eb5a3cdd5c: avcodec/mips/h264chroma_mmi: Version 2 of the optimizations for loongson mmi [14:59:18 CEST] <durandal_1707> J_Darnley: why you disagree? [15:52:01 CEST] <cone-443> ffmpeg 03Paul B Mahol 07master:ccc3f4e7104e: avcodec/sgirledec: simplify, no need to use reget buffer [16:24:49 CEST] <durandal_1707> why are there reviews if there are no reviewers? [16:48:59 CEST] <Illya> How do I go about adding another pixel format to FFmpeg? Is it just a case of adding them to the enum, or is it more involved? [16:49:48 CEST] <Fyr> guys, I encountered strange thing: MKVToolnix cannot edit MKV files created by FFMPEG. after it edited, player crashes when playing the final MKV. [16:49:49 CEST] <Fyr> is there different standards of MKV file format? [16:52:19 CEST] <thardin> Illya: you need to add converters for them too [16:52:45 CEST] <atomnuker> you don't actually [16:53:27 CEST] <thardin> right, if you only use that format [16:53:34 CEST] <atomnuker> just an entry in pixdesc.c and endian entries in pixfmt.h [16:54:37 CEST] <atomnuker> and as long as you add support for that format to whatever encoder/decoder you'll be able to feed in a raw file directly [17:31:37 CEST] <Illya> atomnuker: you mean I'd have to add support for every encoder/decoder? [17:32:45 CEST] <Illya> or make the converter [17:33:37 CEST] <nevcairiel> converters are not required, but formats are pretty useless if nothing can use them :) [17:35:11 CEST] <nevcairiel> note that for complicated packed raw formats, we have choosen to use decoders instead, because supporting their unpacking would otherwise be rather annoying [17:36:10 CEST] <cone-443> ffmpeg 03Zhou Xiaoyong 07master:a20646bb247a: avcodec/mips/mpegvideo_mmi: Version 2 of the optimizations for loongson mmi [17:36:11 CEST] <cone-443> ffmpeg 03Zhou Xiaoyong 07master:4a963ee698c0: avcodec/mips: loongson optimize hpeldsp with mmi v1 [17:36:12 CEST] <cone-443> ffmpeg 03Zhou Xiaoyong 07master:c749be9eb349: avcodec/mips: loongson optimize h264pred with mmi v3 [18:02:20 CEST] <Compn> Illya : its easier if you check a commit that added a previous pixfmt [18:02:30 CEST] <Compn> but also if its packed or strange it has to be a decoder :P [18:02:45 CEST] <Compn> what fmt is it ? [18:04:09 CEST] <JEEB> it doesn't have to be a decoder, but it's just simpler to implement because eff swscale [18:04:18 CEST] <Illya> ARGB4444, RGBA4444, ABGR4444, BGRA4444, ARGB1555, RGBA5551, ABGR1555, BGRA5551, and ARGB2101010 [18:04:18 CEST] <JEEB> which is a sad state of affairs [18:16:42 CEST] <cone-443> ffmpeg 03Paul B Mahol 07master:8e069eb525d4: avcodec/aic: add frame threading support [18:20:41 CEST] <Illya> Compn: do you know of one? been trying to find one for the last 20 minutes [18:23:23 CEST] <durandal_1707> Illya: adding just input support? [18:23:42 CEST] <durandal_1707> do you have samples? [18:23:51 CEST] <Illya> no [18:24:18 CEST] <durandal_1707> then why add it ? [18:24:22 CEST] <Illya> I was going to add them for output support, SDL2 takes those formats. [18:24:35 CEST] <durandal_1707> ah [18:25:18 CEST] <Illya> So that, in the future, if inputs are needed, then an output is already there. Since I think it'd be silly to leave out some of the possible SDL2 pixel output formats [18:26:43 CEST] <durandal_1707> improving swscale is not trivial, exp output stuff, so leave that for later IMHO [18:33:50 CEST] <Illya> durandal_1707: ok. Well if it's difficult then I probably won't be able to do it. So I'll just leave them there, as you said [18:39:39 CEST] <jkqxz> Who would ever want to use 4/5-bit video output like that? I don't think there is any value in it: even if something did create one of those formats (which it won't currently, because the formats are not even present in ffmpeg), the cost of converting it to 8-bit is not significant. [18:39:54 CEST] <jkqxz> Maybe the 10-bit one there is interesting, but if it also takes some other 10-bit format which is normally supported then it's hard to justify adding it separately. [18:43:32 CEST] <durandal_1707> yes, adding this would make sense if there is codecs using it, and even then it would not be mandatory [18:49:07 CEST] <cone-443> ffmpeg 03Michael Niedermayer 07master:e7c5dbb4d117: avfilter/avf_ahistogram: raise minimum acmax to 1 [18:54:37 CEST] <cone-443> ffmpeg 03Paul B Mahol 07master:8c17eea46fea: avcodec: add adpcm MTAF decoder [18:54:38 CEST] <cone-443> ffmpeg 03Paul B Mahol 07master:4da4bfcdc900: avformat: add MTAF demuxer [19:14:45 CEST] <pfelt> as i rework my decklink input mode autodetect patch per marton's suggestions, what's the right way to resubmit? do a (v4)? [19:22:34 CEST] <Illya> pfelt: If your previous one was v3, and you feel you've done everything you needed to, then I'd assume so. [19:23:46 CEST] <cone-443> ffmpeg 03Paul B Mahol 07master:defb960a47f6: avfilter/af_loudnorm: fix crash when ebur128 initialization was not successfull/complete [21:51:41 CEST] <andrey_utkin> Sorry to disturb the maillists moderators, I want to ask to approve the posting to libav-user by subject "[JOB] CCTV system developer". Not urgent. Thanks in advance. [00:00:00 CEST] --- Sun May 22 2016

1 0

ffmpeg.log.20160521
by burek 22 May '16

22 May '16

[02:05:01 CEST] <lostquelana> oh, I never remembered to thank you nichego and ln for helping me earlier [02:05:04 CEST] <lostquelana> thanks you guys [02:43:56 CEST] <lvella> I have a DVD source with 5.1 ac3 audio, but I believe original source is either mono or stereo [02:44:19 CEST] <lvella> is there an easy way to know if the tracks were upscaled from mono or stereo? [02:58:55 CEST] <c_14> Listen to/extract the individual channels and compare them to see if they're the same/extremely similar. [03:01:23 CEST] <c_14> https://trac.ffmpeg.org/wiki/AudioChannelManipulation#a5.16mono <- you can use that to extract them [03:23:03 CEST] <qxt> How do I see what the Chroma resolution of a video file is with ffprobe ? Not sure if this file is YUY2 or YU12. [03:25:07 CEST] <durandal_170> qxt: by inspecting returned format [03:25:32 CEST] <qxt> -show_format ? [03:26:49 CEST] <durandal_170> maybe, its pixel format [03:42:18 CEST] <qxt> pix_fmt=yuv420p is that the chroma resolution? [03:42:45 CEST] <qxt> this is some old dv tape from like 1991. [03:43:07 CEST] <qxt> Just dumped it onto my computer using IEEE1394 [03:49:19 CEST] <qxt> I am gonna guess I am retarded and did not see that was YV12. yuv420p is planar 4:2:0 chroma (shared between 2x2 pixels). [04:02:15 CEST] <qxt> Anybody know of a video frameserver that works with vlc? Need to feed some video to avisynth [12:54:50 CEST] <Fyr> guys, what is ffmpeg-10bit in Zeranoe's build? [12:55:12 CEST] <JEEB> probably one linked with 10bit x264 [12:55:30 CEST] <JEEB> since you can only have one bitness built into the encoder (between 8 and 10, although 9 is almost universally unused) [12:55:56 CEST] <Fyr> interesting [12:56:56 CEST] <Fyr> does 10-bit change anything to eye? [12:57:22 CEST] <c_14> depends on the bitrate, if you have relatively low-bitrate content with gradients then yes [12:58:19 CEST] <furq> 10-bit generally compresses better in my experience [12:58:32 CEST] <Fyr> cool [12:58:35 CEST] <JEEB> 10bit is mostly used for the compression benefit. you can think of a compression error of N bit in 8bit width vs 10bit width [12:58:45 CEST] <furq> you shouldn't use it if you have any compatibility requirements at all [12:58:50 CEST] <Fyr> furq, how does it compress in comparison with h264? [12:58:52 CEST] <JEEB> usually people use 8bit for HW decoding compatibility, though [12:58:54 CEST] <Fyr> *265 [12:59:02 CEST] <furq> i've not used x265 much [12:59:10 CEST] <Fyr> bad [12:59:12 CEST] <furq> the compression benefit with 10-bit x264 is basically free in terms of speed though [12:59:24 CEST] <furq> whereas x265 is ~10x slower if you want it to beat x264 [12:59:29 CEST] <Fyr> x265 compresses as twice as better than h264. [12:59:33 CEST] <JEEB> ahahahahaha [12:59:36 CEST] <furq> sure [12:59:43 CEST] <JEEB> HEVC as a format is better than AVC, yes [12:59:59 CEST] <JEEB> but now you're sticking implementations into the game [13:00:05 CEST] <JEEB> and those are much less optimized at the moment [13:00:48 CEST] <furq> which highly scientific study did you get that twice as good figure from [13:01:03 CEST] <JEEB> JCT-VC made PSNR-based studies, that's true. but that's with HM [13:01:09 CEST] <JEEB> + it's theoretical [13:01:18 CEST] <JEEB> also I think they had some viewing-based tests [13:01:28 CEST] <JEEB> but as I said, those tests are done with the reference encoder [13:01:43 CEST] <JEEB> and they are aiming to theoretically set the bar of how well the format can be used [13:02:06 CEST] <JEEB> now the actual human-usable encoders have to find the places they can optimize in both psychovisual as well as speed side [13:02:08 CEST] <furq> it's annoying because i see a lot of people who just take that as gospel and release stuff with x265 set to half the bitrate they'd normally use [13:02:16 CEST] <furq> and then they use preset medium [13:02:19 CEST] <furq> so the result is just garbage [13:03:14 CEST] <JEEB> funny of course is the fact that on such low bit rate x265 probably does somewhat better than x264, due to the more-PSNR-like result (blurring, lines are probably clearer) [13:03:21 CEST] <JEEB> but preset medium is what it is of course :P [13:03:53 CEST] <JEEB> I should do another test, but I want to wait for my 2xE5-2670 rig to arrive [13:04:20 CEST] <furq> even with a slower preset i'd be shocked if it ever beat x264 veryslow at half the bitrate [13:04:22 CEST] <Fyr> I always use the slowest preset and one of the highest crf. =) [13:04:35 CEST] <furq> enjoy your 2fps encodes then [13:04:42 CEST] <JEEB> > highest CRF [13:04:47 CEST] <JEEB> enjoy your highest compression [13:04:53 CEST] <furq> i assume that means lowest [13:05:00 CEST] <Fyr> yeah [13:05:08 CEST] <JEEB> then he was on x264 in lossless mode with a 8bit encoder :P [13:05:11 CEST] <__jack__> cbr is garbage anyway, you should not use it [13:05:18 CEST] <JEEB> crf != cbr [13:05:19 CEST] <furq> who said anything about cbr [13:05:22 CEST] <Fyr> guys, what is hdmv_pgs_subtitle? [13:05:29 CEST] <Fyr> is it srt? [13:05:34 CEST] <JEEB> it's image-based subs from blu-rays [13:05:47 CEST] <Fyr> can I convert it into SRT? [13:05:52 CEST] <furq> not with ffmpeg [13:05:53 CEST] <JEEB> with subtitle edit, yes [13:05:58 CEST] <JEEB> it'll OCR [13:06:00 CEST] <furq> you'd need an ocr tool like subtitle edit [13:06:08 CEST] <Fyr> JEEB, what tool should I use? [13:06:13 CEST] <JEEB> I just noted one [13:06:17 CEST] <JEEB> same that furq noted [13:06:24 CEST] <Fyr> ok [13:06:43 CEST] <Fyr> didn't realize, that it's a name of program. =) [13:07:00 CEST] <furq> it's windows only if that's an issue [13:07:07 CEST] <furq> there's a popular cross-platform equivalent but i forget the name [13:10:36 CEST] <Fyr> can I export PGS subtitles into a temporary MKV file, then compound the final MKV file with PGS subtitles? [13:10:49 CEST] <Fyr> (I'm remuxing a Blu-ray) [13:10:52 CEST] <furq> give it a try [13:12:46 CEST] <Fyr> *highest reasonable CRF [13:12:56 CEST] <Fyr> (= [13:28:21 CEST] <Fyr> does mpegts support chapters? [13:39:38 CEST] <DHE> Fyr: no, mpegts is intended to be streaming-friendly. to the point that you could join it at any single byte or even maybe bit in-transit [13:39:56 CEST] <Fyr> cool, thanks [13:42:34 CEST] <Fyr> DHE, is VOB streaming-friendly? [13:43:59 CEST] <DHE> fairly well so [13:45:43 CEST] <DHE> for reference, mpegts is what's used by digital over-the-air broadcasts [14:25:01 CEST] <BtbN> Isn't VOB mpegps? [14:25:08 CEST] <BtbN> So, the non-streaming variant of it [14:26:33 CEST] <Fyr> guys, how to mux multiple input into one-filed output? [14:26:57 CEST] <JEEB> multiple -i and -map [14:27:02 CEST] <JEEB> read up on ffmpeg-all.html [14:27:23 CEST] <Fyr> I used the example, but ffmpeg uses only the very first audio. [14:27:33 CEST] <Fyr> ignores the other audio streams. [14:28:47 CEST] <furq> yeah that's what -map is for [14:29:28 CEST] <Fyr> I used: [14:29:28 CEST] <Fyr> $ ffmpeg -ss 16:30 -t 60 -i .\II.h264 -ss 16:30 -t 60 -i .\II-1.mkv -ss 16:30 -t 60 -i .\II-2.mkv -ss 16:30 -t 60 -i .\II-3.mkv -map 0:0 -map 1:0 -map 2:0 -map 3:0 -c copy -threads 3 -y 1.mkv [14:29:50 CEST] <Fyr> the output file has only the video and the very first audio stream. [14:31:32 CEST] <__jack__> Fyr: -map 1:0 means "get the first stream from input #2"; you may need more map, like "-map 1:1" etc [14:31:46 CEST] <furq> you presumably want -map 1:a instead of 1:0 [14:31:48 CEST] <furq> likewise for 2 and 3 [14:32:07 CEST] <__jack__> without mapping, ffmpeg get a stream-type at all, that's why you got only a single audio stream [14:33:15 CEST] <Fyr> __jack__, no, the first file - video, the other ones - audio. [14:33:37 CEST] <Fyr> despite the fact that it's MKV. [14:33:50 CEST] <Fyr> it has only the audio stream. [14:47:03 CEST] <onodera> Hi, is it possible to capture my cursor when recording my screen using x11grab [14:50:45 CEST] <onodera> oh it does this by default heh [14:50:46 CEST] <durandal_1707> onodera: it's not captured by default? [14:50:48 CEST] <onodera> sorry [14:50:53 CEST] <onodera> durandal_1707: yeah it is [15:04:53 CEST] <AndrewMock> I'm about setup a test between qaac and ffmpeg's native aac encoder at 96kbps for professionally-recorded speech. [15:05:02 CEST] <AndrewMock> Which one would win today? [15:05:24 CEST] <AndrewMock> I know that ffmpeg's native aac encoder has made good progress. [15:06:42 CEST] <furq> why are you asking us if you're about to test [15:07:07 CEST] <Fyr> AndrewMock, no need, FFMPEG's AAC still sucks. [15:07:20 CEST] <AndrewMock> i was just excited lol [15:07:23 CEST] <Fyr> variable bitrate is not implemented yet. [15:07:46 CEST] <AndrewMock> haha okay i will test in a year or so i guess [15:07:57 CEST] <AndrewMock> thx for the timesaver [15:08:09 CEST] <Fyr> =) [15:08:20 CEST] <Fyr> if you set the bitrate lower than 128 kbps, you will hear Satan's voice. [15:08:59 CEST] <furq> 14:07:23 ( Fyr) variable bitrate is not implemented yet. [15:09:03 CEST] <furq> the docs disagree with you on that [15:09:07 CEST] <Fyr> higher bitrate will sound just bad. [15:09:18 CEST] <Fyr> furq, try it. [15:09:21 CEST] <AndrewMock> oh i mean 96kbps mono male speech only [15:09:38 CEST] <furq> at 96k per channel i doubt you'll be able to tell any difference [15:09:39 CEST] <Fyr> atomnuker said that he will implement it in the end of the year. [15:10:01 CEST] <AndrewMock> either way qaac is stable i will stick with that [15:10:17 CEST] <Fyr> thumb up [15:11:01 CEST] <JEEB> uhh, I don't see the issue in lack of VBR other than possibly using more bits than you want to [15:11:19 CEST] <JEEB> and 96kbps mono is indeed high enough that even a crappy encoder would be able to handle it [15:11:55 CEST] <JEEB> of course VBR would be more fun but telling someone an encoder sucks by just not having it is... [15:12:12 CEST] <AndrewMock> even when played through a few grand in gear? [15:12:13 CEST] <furq> 14:09:18 ( Fyr) furq, try it. [15:12:15 CEST] <furq> i just did [15:12:16 CEST] <furq> it works fine [15:12:27 CEST] <furq> i also checked the logs and the only thing i can find atomnuker saying about it is that it works fine, to you [15:12:30 CEST] <AndrewMock> i.e. sound reinforcement / PA [15:12:31 CEST] <Fyr> furq, what parameters have you used? [15:12:49 CEST] <furq> -q:a 1 and -q:a 2 both work and give different bitrates [15:12:52 CEST] <JEEB> AndrewMock: you have to have something that fails real hard at 96kbps + mono. possibly FFmpeg's encoder circa 2010 when it created audible artifacts [15:13:09 CEST] <furq> and they're both vbr [15:13:18 CEST] <AndrewMock> alright [15:13:19 CEST] <AndrewMock> peace [15:13:23 CEST] <AndrewMock> thx [15:14:11 CEST] <furq> 15:18:01 ( atomnuker) Fyr: yes, the encoder supports VBR, just use -q:a 0.0-1.0 instead of -b:a 128k [15:15:35 CEST] <furq> that -q range seems to be wrong/outdated though [15:15:42 CEST] <JEEB> yeah, most probably [15:16:33 CEST] <Fyr> agreed [15:16:41 CEST] <Fyr> but this range says nothing to me. [15:16:47 CEST] <Fyr> + it's not in the wiki. [15:16:56 CEST] <Fyr> + it's not in the docs. [15:17:20 CEST] <JEEB> welcome to OSS, unless someone cares there's no magically appearing docs [15:17:26 CEST] <JEEB> or actually not limited to OSS [15:17:31 CEST] <Fyr> sure [15:17:39 CEST] <JEEB> I've seen some proprietary projects having fun amounts of docs :P [15:17:52 CEST] <furq> i vaguely remember that vbr mode isn't particularly well-optimised yet [15:18:05 CEST] <furq> but it's definitely working [15:18:17 CEST] <JEEB> even still that shouldn't actually worsen the capability of the encoder to compress [15:18:37 CEST] <JEEB> since that's the same as with CBR. CBR is just a more limited VBR [15:18:50 CEST] <furq> oh fun [15:19:02 CEST] <furq> -q:a 0 gives me 144kbps but -q:a 0.1 is 20kbps [15:19:16 CEST] <JEEB> zero probably goes to some default [15:19:22 CEST] <furq> yeah that comes out the same as 1 [15:19:30 CEST] <furq> 2 is ~2x the bitrate of 1 though [15:19:43 CEST] <furq> although there isn't much point in a 256k stereo aac [15:20:03 CEST] <furq> if i wanted to spend that many bits i'd just use lame [15:48:49 CEST] <Fyr> sorry, I just confused dates in here: https://trac.ffmpeg.org/ticket/2686#comment:379 [15:49:27 CEST] <Fyr> but still VBR is not documented and not implemented. [15:49:49 CEST] <furq> it is implemented [15:49:56 CEST] <furq> and it is partially documented, just the q range isn't [15:52:10 CEST] <Fyr> then it should be. [15:58:13 CEST] <ZeuZ> Guys, I need to turn a JPEG into a frame that will repeat 24 times, is there any libav compatible way of reading a FILE* structure or file descriptor? or do I need to parse it manually? [16:11:30 CEST] <pa> hello! anyone with a dvb-s2 card pointed at Hotbird 13E in here? :-) [16:22:56 CEST] <kynlem> hey [16:38:14 CEST] <kynlem> is there a lossless audio format that can live happily inside a mp4 container? [16:38:26 CEST] <furq> alac [16:38:55 CEST] <JEEB> I think officially only ALS [16:39:11 CEST] <JEEB> but what devices support, probably ALAC with the right types mentioned in the header [16:39:46 CEST] <kynlem> ffmpeg -i canontest.mov -ss 0.00 -t 60.06 -acodec alac -vcodec libx264 chunk-alac.mp4 [16:40:04 CEST] <furq> als and sls are officially supported in mp4 but i'd expect alac is more common than either of them [16:40:09 CEST] <kynlem> doesn't work "[mp4 @ 0x7fef03001000] Could not find tag for codec alac in stream #1, codec not currently supported in container" [16:40:15 CEST] <furq> oh fun [16:40:19 CEST] <kynlem> "Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument" [16:40:25 CEST] <JEEB> yeah, then it requires mov [16:40:26 CEST] <c_14> only works with the mov muxer [16:40:33 CEST] <JEEB> which is just another mode of the same muxer [16:41:01 CEST] <c_14> I think someone wanted to also allow it for mp4, but the thread never advanced. [16:41:21 CEST] <kynlem> but the mov container is less widely supported than mp4, no? [16:41:44 CEST] <kynlem> seems like it [16:41:54 CEST] <furq> do you need specific device support [16:42:05 CEST] <kynlem> no [16:42:37 CEST] <furq> if you don't then just use flac in mkv [16:43:00 CEST] <furq> that way you don't have to pollute the earth with more alac [16:43:14 CEST] <kynlem> got it. on the other hand, what would be the standard command line arguments to get audio of the highest possible quality inside an mp4 container? [16:43:32 CEST] <furq> apparently there's no als encoder in ffmpeg so you'd be stuck with aac [16:43:32 CEST] <JEEB> raw pcm (which is now standardized) or ALS [16:43:37 CEST] <furq> oh is pcm supported now [16:43:43 CEST] <JEEB> those are the two lossless formats in mp4 [16:43:56 CEST] <kynlem> pcm gives me same trouble [16:44:13 CEST] <kynlem> ffmpeg wouldn't put it inside mp4 [16:45:14 CEST] <kynlem> ffmpeg version 2.8.3, that is. [16:45:51 CEST] <JEEB> try latest [16:45:53 CEST] <furq> i get the same thing with a recent git version [16:45:57 CEST] <JEEB> ok [16:45:59 CEST] <furq> recent-ish, about a month old [16:46:05 CEST] <JEEB> then the muxer might have not been updated to the raw PCM specs [16:46:19 CEST] <JEEB> just like the FLV muxed didn't let you mux non-44.1kHz AAC for quite a while :) [16:46:22 CEST] <JEEB> *muxer [16:46:50 CEST] <furq> i take it alac in mp4 is a non-starter until it shows up in some kind of actual spec [16:47:27 CEST] <BtbN> any other rate in flv is a bad idea though, JEEB. [16:47:45 CEST] <BtbN> I have seen weird things with non-44.1Hz in flv [16:47:51 CEST] <BtbN> +k [16:53:59 CEST] <JEEB> BtbN: it works just fine in flash with AAC [16:54:18 CEST] <JEEB> because the FLV spec says that the field is just written as 44.1kHz in the container [16:54:25 CEST] <JEEB> and the actual AAC stream is to be used for any information [16:54:25 CEST] <BtbN> If you upload flv with 48kHz AAC to youtube, it interprets it as 44.1kHz [16:54:35 CEST] <JEEB> then youtube is a dumb fuck [16:54:47 CEST] <JEEB> when they re-transcode your shit [16:55:12 CEST] <BtbN> yep, but It made me switch everything to .ts or .mkv [16:55:19 CEST] <JEEB> not a bad idea [16:56:11 CEST] <JEEB> but in any case, you can always have broken third parties in the flow. But as far as FLV is generally concerned, flash and modern FFmpeg handle AAC in it just fine [16:56:26 CEST] <JEEB> although I haven't tried multichannel AAC in FLV recently with flash [16:56:45 CEST] <BtbN> In general i like the flv container. It's very simple and just works. [16:57:10 CEST] <JEEB> (mostly because flash sucks and I would like standardized outputs read by clients to it) [16:57:54 CEST] <BtbN> Isn't flv primarily kept alive by rtmp now? [16:57:59 CEST] <JEEB> yes [16:58:20 CEST] <JEEB> although it is used by some 3rd parties like rtmp-nginx [16:58:28 CEST] <JEEB> to stream out things in other formats as well [16:58:34 CEST] <JEEB> as the input protocol [16:58:43 CEST] <BtbN> primarily due to the lack of alternatives. [16:58:45 CEST] <furq> that's still using rtmp [16:59:03 CEST] <BtbN> I'm not aware of anything else than can be used as ingest protocol for video streams. [16:59:08 CEST] <JEEB> furq: as input, yes [16:59:19 CEST] <JEEB> BtbN: MSS is used by some servers [16:59:31 CEST] <JEEB> in theory you could use a lot of things as the input feed [16:59:38 CEST] <JEEB> but those are the two I've seen so far [17:00:08 CEST] <BtbN> rtmp supports things like dropping the least important packages in case of congestion. [17:00:32 CEST] <furq> i'd take rtmp all day long over hls or dash [17:00:44 CEST] <JEEB> yes, it's an actual stream protocol [17:00:51 CEST] <BtbN> hls/dash only exist because browsers are shit. [17:01:00 CEST] <JEEB> and because people want to use their HTTP caching solutions [17:01:15 CEST] <furq> and because people apparently don't need to seek backwards any more [17:01:28 CEST] <JEEB> HLS and DASH don't stop you from having a buffer available [17:01:29 CEST] <BtbN> seeking in a hls/dash stream works just fine [17:01:47 CEST] <JEEB> it's just not a streaming protocol [17:01:54 CEST] <furq> dash on youtube breaks about 50% of the time for me when i seek backwards [17:02:07 CEST] <furq> i'm sure that's youtube's fault but i don't regularly use any other dash vod service [17:02:10 CEST] <BtbN> Never seen seeking on YouTube break. [17:02:10 CEST] <JEEB> or well, DASH and HLS are generally used over HTTP which is not a video streaming specific protocol :P [17:02:19 CEST] <furq> i guess it could also be firefox's fault [17:02:31 CEST] <furq> it wouldn't be a great surprise if their dash/mse implementation is broken [17:02:39 CEST] <BtbN> I'm using Firefox. [17:02:52 CEST] <JEEB> they can have various bugs as well, and 'tube generally is probably optimizing for Chrome's JS/MSE implementation [17:03:05 CEST] <furq> has seeking seriously never broken for you on youtube [17:03:09 CEST] <JEEB> recently I've had the buffering in the 'tube player fail with some clips [17:03:27 CEST] <furq> that 50% figure was an exaggeration but not by much [17:03:30 CEST] <JEEB> hasn't broken for me more than any MSE-based thing [17:03:34 CEST] <furq> i'd say it happens at least every third or fourth time i try it [17:03:44 CEST] <furq> the video just hangs and i have to refresh the page [17:03:58 CEST] <JEEB> yeah, deffo a player implementation (or browser implementation) issue [17:03:58 CEST] <furq> seeking backwards, that is [17:04:00 CEST] <BtbN> Is that some old gstreamer based firefox? [17:04:02 CEST] <furq> seeking forwards is usually fine [17:04:05 CEST] <furq> no it's firefox on windows [17:04:17 CEST] <BtbN> Never had that issue. [17:04:39 CEST] <furq> fun [17:04:41 CEST] <JEEB> if you want to see how well DASH works in general you could try Google's Android DASH/HLS/MSS player library's example player [17:04:50 CEST] <JEEB> that thing is actually surprisingly good [17:05:08 CEST] <BtbN> HLS in browsers is kinda... interesting. [17:05:09 CEST] <furq> i did have a bunch of issues just after they switched to dash but i'm pretty sure those were to do with my isp [17:05:10 CEST] <JEEB> ExoPlayer I think [17:05:16 CEST] <BtbN> They remux to mp4 on the fly... [17:05:23 CEST] <furq> yeah i've used hls.js [17:05:36 CEST] <furq> it works surprisingly well for a hideous abomination [17:06:04 CEST] <BtbN> isn't that actually some ffmpeg-like tool, re-compiler to js? [17:06:12 CEST] <JEEB> no [17:06:18 CEST] <JEEB> hls.js is a HLS parser in js [17:06:24 CEST] <JEEB> and then remuxing into ISOBMFF fragments [17:06:29 CEST] <furq> what he said [17:06:31 CEST] <BtbN> I'm pretty sure I have seen some emscripten headers in there. [18:06:32 CEST] <Fyr> is difference in size usual when remuxing? I copied all the streams from m2ts to mkv and found that the final output file is 3 Gb smaller than the source one. [18:09:43 CEST] <furq> ts has more overhead than mkv [18:10:05 CEST] <furq> i've remuxed some 720p movies and got a ~500MB size difference [18:10:40 CEST] <furq> if that's an m2ts from a blu-ray or something then 3GB sounds plausible [19:20:03 CEST] <pandaologist> can't get graticules on the waveform monitors working anymore [19:20:07 CEST] <pandaologist> examples: https://trac.ffmpeg.org/wiki/WaveformMonitor [19:20:23 CEST] <pandaologist> has this been deprecated, or has the syntax changed? [19:35:31 CEST] <durandal_1707> pandaologist: command? [19:38:31 CEST] <durandal_1707> pandaologist: if you use rgb it will not display it [19:41:23 CEST] <pandaologist> guh [19:41:29 CEST] <pandaologist> found the problem [19:41:39 CEST] <pandaologist> only in git at the moment [19:41:44 CEST] <pandaologist> durandal_1707: thanks [20:48:55 CEST] <kynlem> ffmpeg -i canontest.mov -ss 180.18 -t 60.06 -acodec libfdk_aac -b:a 192k -vcodec libx264 -crf 16 -x264opts keyint=12:min-keyint=12:scenecut=-1 -map 0 chunk3.mp4 [20:49:41 CEST] <kynlem> when i split canontest.mov into 4 videos as above, and then merge them back, i get a popping sound at points where they meet [20:50:27 CEST] <kynlem> but when i do the same and use -acodec copy (it's pcm) and mov as the output container, then they merge back fine [20:50:31 CEST] <kynlem> any clues? [21:06:15 CEST] <Fyr> Gibbs phenomenon? [21:08:57 CEST] <kynlem> Fyr: and how can you deal with it? [21:09:23 CEST] <Fyr> just copy? [21:09:31 CEST] <kynlem> not possible [21:09:35 CEST] <Fyr> if copying doesn't give that sound. [21:09:45 CEST] <kynlem> the source is pcm-encoded, the output should be in mp4 container [21:09:57 CEST] <kynlem> mp4 isn't a friend of pcm [21:10:22 CEST] <kynlem> or any lossless audio codec for that matter [21:11:10 CEST] <Fyr> I can suggest: 1) convert 2) cut [21:11:10 CEST] <Fyr> if this doesn't work: 1) cut 2) convert [21:11:20 CEST] <Fyr> I think the latter is what you're talking about. [21:12:04 CEST] <kynlem> yeah, i'm trying the former now [21:12:28 CEST] <Fyr> why is it not possible? [21:13:41 CEST] <furq> ffmpeg doesn't support pcm or alac in mp4 [21:13:49 CEST] <furq> and it doesn't have als or sls encoders [21:14:12 CEST] <Fyr> furq, why does it not support ALAC? [21:14:23 CEST] <furq> because alac isn't officially supported in mp4 [21:14:33 CEST] <Fyr> I all my life converted into/from ALAC with FFMPEG. [21:14:37 CEST] <furq> pcm is but it's a relatively recent addition [21:14:51 CEST] <Fyr> the only problem I encountered - covert art. [21:14:53 CEST] <furq> why would you ever convert to alac [21:15:36 CEST] <furq> kynlem: i assume there's a requirement for the split files to be mp4 and they're not just intermediate files [21:16:28 CEST] <furq> and also that those cut points won't end up at keyframe boundaries if you encode the whole thing and then split it [21:17:08 CEST] <Fyr> ALAC had better compression with faster decompression a few years ago. [21:17:19 CEST] <Fyr> only a year ago FLAC become comparable. [21:17:30 CEST] <Fyr> at least, when converting with FFMPEG. [21:17:50 CEST] <Fyr> though, some perks still have not been implemented in FFMPEG. [21:18:00 CEST] <furq> i've never really used flac with ffmpeg but flac in general has always been better than alac [21:18:11 CEST] <Fyr> in general - yes. [21:18:19 CEST] <Fyr> with FFMPEG - no. [21:20:34 CEST] <durandal_1707> Fyr: why? [21:20:42 CEST] <Fyr> what why? [21:21:11 CEST] <durandal_1707> explain what you just said? [21:21:19 CEST] <Fyr> about ALAC? [21:21:35 CEST] <durandal_1707> FLAC [21:22:43 CEST] <Fyr> when converting audio file with FFMPEG, ALAC had better compression than FLAC. a year ago I noticed that FFMPEG developed to provide good compression in comparison to ALAC. [21:22:49 CEST] <Fyr> sorry, I have to reboot. [21:23:01 CEST] <Fyr> now I keep all my lossless audio library in FLAC. [21:30:04 CEST] <pandaologist> can someone help me? I would like to add an audio volume monitor to my filter; showvolume would work ok [21:30:08 CEST] <pandaologist> https://ffmpeg.org/ffmpeg-filters.html#toc-showvolume [21:30:17 CEST] <pandaologist> current filter is as follows [21:30:19 CEST] <pandaologist> ffplay video.avi -vf "format=yuv422p,split=5[a][b][c][d][e],[b]waveform=c=4:g=1:flags=dots,scale=h=128[bb],[c]waveform=c=2:g=1:flags=dots,scale=h=128[cc],[d]waveform=m=row:r=0:g=1:flags=dots[dd],[e]vectorscope=color5:g=1[ee],[a][bb][cc]vstack=inputs=3[V1],[dd][ee]vstack[V2],[V1][V2]hstack" [21:30:49 CEST] <pandaologist> ideally it would be stacked vertically or horizontally [21:33:18 CEST] <pandaologist> i just can't work out how to display a visual audio monitor alongside the video [21:55:26 CEST] <durandal_1707> pandaologist: using -filter-complex for all of them instead of -vf [22:24:48 CEST] <itt788> i'm trying to extract a part of an mkv video that contains 4 streams being as follwing -- 0 : video, 1 : audio lang 1, 2 : audio lang 2, 3 : subtitles. I would like my extract to contain either or audio streams or if impossible lang 1. My command line is "ffmpeg -ss 0:10:00 -i in.mkv -c:v copy -c:a:0 copy out.mkv". Either with using -c:a:0 or -c:a:1 I'm getting only lang 2 in my extract. [22:27:45 CEST] <pandaologist> durandal_1707: can i use that directly in ffplay? [22:27:59 CEST] <kynlem> furq: do you mean video keyframe? [22:28:05 CEST] <pandaologist> i guess i could filter in ffmpeg and pipe to ffplay [22:28:23 CEST] <durandal_1707> pandaologist: yes, pipe [22:28:33 CEST] <kynlem> furq: in the source video, it's every 12th frame [22:28:44 CEST] <kynlem> furq: and i'm keeping it that way in the cuts using 'keyint=12:min-keyint=12' [22:49:07 CEST] <pandaologist> durandal_1707: got it! thanks :) [23:38:20 CEST] <ploop> is it possible to use ffmpeg to make a video that's just a still image for a certain amount of time? I tried "ffmpeg -i image.png -t 10 video.mkv", but the output seems to last a single frame instead of 10 seconds [00:00:00 CEST] --- Sun May 22 2016

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