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June 2017

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ffmpeg-devel.log.20170609
by burek 10 Jun '17

10 Jun '17

[00:21:43 CEST] <J_Darnley> Should I make some sort of note in the commit messages that I have sort of cherry-picked these commits? [01:07:21 CEST] <J_Darnley> Ah. I didn't intend for those to have CCs. Oh well. [01:34:20 CEST] <nevcairiel> git does that by default i think, can be a bit unexpected sometimes [01:46:49 CEST] <J_Darnley> Yeah. I forgot about them plus I have configed it so that I don't CC myself. [13:26:52 CEST] <J_Darnley> Rats. I should get my Laptop's ssh key added to the git server [13:29:06 CEST] <cone-627> ffmpeg 03raymondzheng 07master:9f20cc5c8458: libavformat/http: return EIO when ffurl_read return 0, but s->off < target_end [13:31:40 CEST] <atomnuker> J_Darnley: better yet use 1 ssh key [13:33:07 CEST] <J_Darnley> Maybe, but putty doesn't support these "curve" algorithms [13:34:04 CEST] <nevcairiel> then you should upgrade putty [13:34:04 CEST] <nevcairiel> :) [13:34:17 CEST] <RiCON> J_Darnley: it does [13:34:22 CEST] <nevcairiel> 0.68 added support for elliptic curves [13:38:17 CEST] <J_Darnley> Ha [13:38:28 CEST] <J_Darnley> I really should [13:50:45 CEST] <cone-627> ffmpeg 03Henrik Gramner 07master:406e0ddc0b9b: x86inc: Fix call with memory operands [13:50:46 CEST] <cone-627> ffmpeg 03Henrik Gramner 07master:88dcdfad0964: x86inc: Make REP_RET identical to RET in SSSE3+ functions [13:50:47 CEST] <cone-627> ffmpeg 03Henrik Gramner 07master:cd4ca8245963: x86inc: Prefer r14/r15 over r12/r13 on x86-64 [13:50:48 CEST] <cone-627> ffmpeg 03Anton Mitrofanov 07master:d991b3e8a87a: x86inc: Remove argument from WIN64_RESTORE_XMM [13:53:05 CEST] <BBB> what happened to the 5th patch? [14:00:05 CEST] <J_Darnley> I want to speak to x264dev first [14:05:15 CEST] <BBB> ok [14:05:24 CEST] <BBB> is the idct patch waiting for any input from me? [14:09:30 CEST] <J_Darnley> Not presently [14:09:57 CEST] <J_Darnley> I have not managed much work on it since your rounding contibution [14:10:10 CEST] <J_Darnley> My nasm troubles have been in the way. [14:10:14 CEST] <BBB> :( [14:10:17 CEST] <BBB> sorry about that [14:10:25 CEST] <BBB> so are we moving to nasm now? [14:10:47 CEST] <J_Darnley> I am going to submit a patch that will require 2.11 [14:10:49 CEST] <BBB> the nasm on my system still cannot actually compile ffmpeg (the rodata in vp9 mc filters fails to work, I think) [14:10:50 CEST] <JEEB> yasm's raison d'etre after all faded aways [14:10:53 CEST] <RiCON> we should, isn't yasm abandonned? [14:11:14 CEST] <JEEB> and yes, mostly abandoned since yasm-ng didn't go anywhere [14:11:17 CEST] <JEEB> (yes, that was a thing) [14:11:20 CEST] <J_Darnley> RiCON: Yes it seems to be [14:11:45 CEST] <J_Darnley> I blame them for want to rewrite in c++ with boost [14:11:58 CEST] <J_Darnley> (I think that was the plan) [14:12:11 CEST] <wm4> wat [14:13:31 CEST] <JEEB> yes, yasm-ng was a C++ rewrite [14:13:34 CEST] <JEEB> because of course [14:14:06 CEST] <RiCON> J_Darnley: i think with future libav merges nasm will be preferred too, iirc [14:14:15 CEST] <wm4> even JS would have been a saner choice [14:14:28 CEST] <RiCON> something to do with generating dependencies information(?) [14:46:18 CEST] <iive> hate boost [15:34:43 CEST] <BBB> J_Darnley: is there a downside to continuing support for yasm for devs that happen to have that installed? [15:36:36 CEST] <J_Darnley> I guess not, it just requires more thought to be put into detection [15:38:31 CEST] <J_Darnley> If you want to keep yasm support please send a reply stating that. [16:17:48 CEST] <J_Darnley> Ah ha! I might have cracked my idct problem [16:18:23 CEST] <J_Darnley> I was setting the new functions for the wrong algorithm selection. [16:46:49 CEST] <cone-627> ffmpeg 03Vittorio Giovara 07master:c12e8f5f0b41: vf_colorspace: Add a pixdesc API alias name for bt2020nc color space [17:30:15 CEST] <J_Darnley> What the? The idct block is supposed to be 16 byte aligned. [17:31:29 CEST] Action: J_Darnley afk [18:16:06 CEST] <BtbN> "Run a RTMP server using libavformat" that's a thing? [18:18:17 CEST] <wm4> let's hope it's not [18:19:36 CEST] <kierank> of course it is [18:19:46 CEST] <kierank> because ffmpeg is everyone's toilet of crap [18:21:21 CEST] <BtbN> you can run ffmpeg standalong as rtmp server? oO [18:21:36 CEST] <wm4> we don't know yet [18:21:44 CEST] <JEEB> no, but you can feed rtmp servers with lavf [18:22:17 CEST] <BtbN> that guy seems to be running lavf as server though [18:22:41 CEST] <JEEB> :D [18:24:20 CEST] <cone-627> ffmpeg 03Sasi Inguva 07master:93db5e3fc41a: lavf/mov.c: offset index timestamps by the minimum pts to make first pts zero [18:24:54 CEST] <wm4> so what's up with those email addresses anyway ("isasi-at-google.com(a)ffmpeg.org") [18:25:56 CEST] <DHE> ffserver back from the dead? (joke) [18:26:07 CEST] <JEEB> wm4: our server has no right to send email as google.com. modern anti-spam validation tech blocks it [18:27:32 CEST] <BtbN> super stupid bullshit, breaks basic "Reply To" if you want to mail them directly [18:27:50 CEST] <BtbN> They should introduce some special thing to handle mailing lists and the like [18:28:12 CEST] <DHE> they did. it was supposed to be DKIM [18:28:18 CEST] <DHE> but it still kinda sucks [18:28:39 CEST] <wm4> ffserver isn't dead yet [18:28:45 CEST] <BtbN> Something like making the mail-server sign the entire mail, and as long as that signature is intact, it can be sent as "from that domain" [18:29:01 CEST] <BtbN> so re-distributing a valid mail is ok [18:30:20 CEST] <DHE> BtbN: the intention of DKIM was that the original signature from the original sender could be forwarded by the mailing list and all would be well with the world. (in reality that still doesn't happen) [20:14:51 CEST] <kierank> atomnuker: https://twitter.com/marcan42/status/873241816434221056 [20:15:22 CEST] <wm4> aw [20:15:37 CEST] <BtbN> Probably tested ffmpeg 2.8 or something [20:16:11 CEST] <wm4> yeah I'd hope some sort of misunderstanding/configuration [20:16:28 CEST] <J_Darnley> Curses. It looks like people on the ML are not as keen to drop yasm as IRC is. [20:17:05 CEST] <MrZeus1> Parallelism in C++ using arrays and gcc: https://www.youtube.com/watch?v=Pc8DfEyAxzg [20:18:06 CEST] <wm4> you should know by now that the ML never agrees to drop anything [20:18:22 CEST] <J_Darnley> :) [20:57:17 CEST] <atomnuker> kierank: given audiophile-level explanation like "notes sound less clean", "there's weird fuzziness", etc. I'm not willing to give it any credence [20:57:56 CEST] <kierank> That guy is very capable [20:58:02 CEST] <kierank> He reverse engineered the ps4 [20:58:31 CEST] <BtbN> Doesn't seem capable at blind listening tests though [20:58:34 CEST] <atomnuker> he's also annoying, wrong and refuses to accept new technology on the basis of "I heard rumors/read it on the internet" [21:00:48 CEST] <wm4> where is he rejecting new technology [21:11:48 CEST] <iive> atomnuker: there did you see these explanations? [21:13:15 CEST] <atomnuker> #mpv [21:31:11 CEST] <iive> i guess there are no public logs of it... [22:00:58 CEST] <kiroma> Passing -mavx2 flag implies all -msse flags, right? [22:01:14 CEST] <thebombzen> I'm not sure what he's doing but like the new AAC encoder was a big deal and a nice accomplishment [22:02:37 CEST] <thebombzen> so the fact that he just tries it and says "nope it definitely sucks" without coming on #ffmpeg and trying to figure out why rather than just blabber on twitter [22:02:47 CEST] <thebombzen> kind of obnoxious [22:03:04 CEST] <wm4> he merely pointed out a case where it failed [22:03:09 CEST] <wm4> and atomnuker reacted aggressively [22:03:32 CEST] <thebombzen> well rightfully so tbh, that was an obnoxious response "nope, all that testing and hard work, they're just lying" [22:03:51 CEST] <thebombzen> the people here are pretty keen to admit when something isn't up to standard on something and needs work [22:04:07 CEST] <thebombzen> "ffmpeg.c is not ideal for realtime streaming, here's a different tool" etc. etc. [22:05:11 CEST] <kiroma> Who says that? [22:05:55 CEST] <thebombzen> it's known that ffmpeg.c's buffers aren't highly configurable and it's difficult to get very-low-latency streaming with it. at least it was a few months ago [22:22:34 CEST] <wm4> IMO latency on demuxing is mostly the fault of libavformat's automagic utils.c stuff [22:22:42 CEST] <wm4> that tries to probe codec parameters etc. [22:26:20 CEST] <BtbN> that only adds startup delay, not latency [22:26:26 CEST] <BtbN> all the buffers everywhere add the latency [22:31:33 CEST] <BtbN> coverity build failed because this sofa-lib already has a build dir... great. [22:32:14 CEST] <BtbN> I wonder if the coverity build can be split up, and submited from multiple sources [22:33:49 CEST] <JEEB> hmm [22:34:14 CEST] <JEEB> seems like I'm having issues with h264_mediacodec with the latest FFmpeg [22:34:39 CEST] <durandal_1707> mediacodec? who cares... [22:34:48 CEST] <wm4> BtbN: well, if you discard all data once started up, it doesn't affect latency I guess [22:34:58 CEST] <wm4> which is why there's a dumb flag that explicitly discards this data [22:36:15 CEST] <wm4> durandal_1707: many [22:37:03 CEST] <JEEB> http://up-cat.net/p/c9c0800b [22:37:11 CEST] <durandal_1707> wm4: couple at most [22:37:24 CEST] <JEEB> this file used to work earlier... :/ [22:38:04 CEST] <wm4> seems like it consumes all packets with no output [22:38:19 CEST] <JEEB> yea, it's all black [22:38:25 CEST] <JEEB> if I switch to sw decoding it works [22:38:37 CEST] <JEEB> I'll try to reboot just in case [22:41:34 CEST] <wm4> I could adjust the mpv fallback to fall back if it feeds more than 32 packets or so with no return [22:42:24 CEST] <JEEB> ok, same after reboot [22:42:35 CEST] <JEEB> when was mediacodec switched to the new style? [22:45:01 CEST] <JEEB> ok, so since my last FFmpeg was from end of Feb pretty much all changes since the march stdatomic thing are new [22:45:04 CEST] <mateo`> durandal_1707: I do care [22:45:44 CEST] <durandal_1707> i dont care that somebody care for nonopensource solution [22:46:30 CEST] <JEEB> well ASICs by their nature are not OSS [22:46:52 CEST] <JEEB> and unfortunately ARM bullshit devices need hwdec [22:47:28 CEST] <mateo`> durandal_1707: ok [22:51:23 CEST] <JEEB> I guess I'll have to test the changes one by one [22:52:02 CEST] <mateo`> JEEB: is it with a specific sample ? [22:52:17 CEST] <JEEB> all of my 8bit AVC stuff seems to be affected [22:53:26 CEST] <JEEB> and the one file with which I was testing hwdec->swdec fallback doesn't fall back [22:54:35 CEST] Action: JEEB starts with 005da88c1ee231eddd9924ad8173aeeab6366165~1 [22:55:08 CEST] <mateo`> JEEB: let me know if I can help [22:56:18 CEST] <JEEB> another guy's device seems to have it working with local files but not external URLs [22:56:27 CEST] <JEEB> so it seems awfully device dependant [23:00:23 CEST] <JEEB> -_- [23:00:52 CEST] <wm4> is that maybe avc vs. annexb [23:01:01 CEST] <wm4> *avcc [23:02:25 CEST] <mateo`> the decoder inserts the h264_mp4toannexb bitstream filter [23:02:29 CEST] <JEEB> doesn't seem so [23:02:37 CEST] <JEEB> @ avcc | annex b [23:02:52 CEST] <JEEB> also 005da88c1ee231eddd9924ad8173aeeab6366165~1 is also borked it seems so it's either mpv or something else [23:06:47 CEST] <mateo`> JEEB: what is the issue exactly ? is it about draining the remaining frame ? [23:07:03 CEST] <JEEB> I open stuff and I only get a black screen [23:08:00 CEST] <mateo`> What is the output of the decoder ? CPU buffers or a surface ? [23:08:14 CEST] <mateo`> Can you share the sample you are trying to play ? [23:08:41 CEST] <JEEB> it happens on all of my samples, let me grab one of them [23:10:49 CEST] <JEEB> https://kuroko.fushizen.eu/videos/HidamariHoneycombED.mp4 [23:10:51 CEST] <JEEB> one example [23:14:07 CEST] <JEEB> I did switch NDK, though... [23:14:24 CEST] <JEEB> let me wipe my prefix and re-build with an older NDK [23:14:34 CEST] <JEEB> r13b => r15 [23:14:44 CEST] <JEEB> since older versions of FFmpeg/libmpv didn't seem to help [23:14:57 CEST] <mateo`> i'll test on my end (on a nexus 5x) [23:26:11 CEST] <JEEB> building older versions with older NDK to make sure [23:28:10 CEST] <mateo`> It works on my side, ffmpeg master as of a32a6b4201dca46c54247194bd5249dfb7c64874, buffer and surface output works as expected [23:28:44 CEST] <JEEB> which NDK? [23:29:08 CEST] <mateo`> i'm using ndk r14b [23:29:14 CEST] <JEEB> ok [23:32:49 CEST] <JEEB> mateo`: are you using gcc or clang btw? [23:33:12 CEST] <mateo`> clang [23:33:17 CEST] <JEEB> ok, so same as me [23:35:16 CEST] <J_Darnley> Oh my god! Does NASM not give a proper warning if you use %9 in an 8 parameter macro? [23:35:44 CEST] <J_Darnley> oh you POS! [23:36:10 CEST] <JEEB> mateo`: HAH [23:36:12 CEST] <JEEB> r13b works [23:37:47 CEST] <mateo`> I will test with r15 but it will take some time as I'll have to rebuild the whole stack I use at work [23:38:31 CEST] <jamrial> JEEB, mateo`: speaking of annexb bitstream filter, you could try porting mediacodec to use the new decoder level autobsf feature [23:39:00 CEST] <jamrial> see 8fb4210ad8 for an example [23:41:18 CEST] <JEEB> mateo`: r15 for some reason utilizes an SVN build of clang [23:41:22 CEST] <JEEB> as in, not a release [23:42:11 CEST] <JEEB> $ ~/ownapps/ndk-toolchain-r15/bin/clang++ --version [23:42:12 CEST] <JEEB> Android clang version 5.0.300080 (based on LLVM 5.0.300080) [23:42:27 CEST] <JEEB> trying to play apple I guess? [23:42:34 CEST] <JEEB> who also seem to use random SVN revisions for their stuff [23:45:01 CEST] <BtbN> wow, this libmysofa is horribly bad and broken [23:45:16 CEST] <BtbN> It comes with a pre-populated ./build dir [23:45:28 CEST] <BtbN> and if you don't "out-of-tree" build in there, the build fails. [23:46:00 CEST] <mateo`> jamrial: I was thinking porting the code to it but that would prevent the future support of encrypted streams through MediaCrypto (a dev from kodi suggested it). We would need to disable the bitstream filtering according to an option of the hwaccel [23:48:51 CEST] <wm4> lol [23:48:58 CEST] <wm4> DRM garbage preventing nice code? check [23:49:26 CEST] <wm4> whyx doesn't google just get their shit together and define avcc support for mediacodec anyway [23:49:51 CEST] <wm4> not to mention that d3d11va can do DRM, despite being a hwaccel [23:56:19 CEST] <mateo`> wm4: I'm personally against the DRM thing, but on the other hand I don't want to restrict users by me personal choice [23:59:04 CEST] <wm4> I can understand that, but in this case there's surely something severely wrong [23:59:34 CEST] <wm4> such as for example what the heck would the mediacodec wrapper code do anyway if the packet data is somehow encrypted enough that annexb repacking doesn't work [00:00:00 CEST] --- Sat Jun 10 2017

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ffmpeg.log.20170609
by burek 10 Jun '17

10 Jun '17

[03:35:51 CEST] <Obliterous> Anyone well versed with ffserver available? [05:06:48 CEST] <waqas> Anyone here up for small paid job for ffmpeg? [05:07:26 CEST] <johnjay> did someone say money??? [05:07:45 CEST] <johnjay> I have a special script that alerts me anytime someone mentions money [05:08:13 CEST] <waqas> lol yes [05:10:07 CEST] <johnjay> well I only used ffmpeg one time for streaming [05:10:23 CEST] <johnjay> but hey maybe I should learn all the command line switches if it means teh moneyz [05:10:32 CEST] <johnjay> what kind of job is it? [05:10:54 CEST] <waqas> Just need to create a slide show that has pan and zoom effect. [05:11:28 CEST] <johnjay> oh like from a given video? [05:11:53 CEST] <waqas> so take brand video, text, images and create a slide show [05:12:31 CEST] <waqas> here is a example video [05:12:32 CEST] <waqas> https://www.youtube.com/watch?v=9dvmyuJpt6k&t=1s [05:14:07 CEST] <johnjay> that audio is wicked loud in my speakers [05:15:11 CEST] <johnjay> that is creepy as hell lol [05:15:21 CEST] <johnjay> the sound makes it seem like a horror movie [05:15:29 CEST] <waqas> lol [05:15:41 CEST] <johnjay> like there's zombies behind those doors when it zooms in all slowly [05:16:38 CEST] <johnjay> so minus the ghostly horror audio of doom [05:16:51 CEST] <johnjay> you're talking about that part where the logo fades in and out followed by text fading in and out? [05:17:23 CEST] <waqas> and images [05:17:38 CEST] <johnjay> oh. so basically turning images into a slide show [05:17:47 CEST] <waqas> yes sir [05:17:56 CEST] <waqas> with pan and zoom effect [05:18:28 CEST] <johnjay> hmm. idk there was any zoom in that vid [05:18:31 CEST] <johnjay> although i only skimmed it. [05:19:24 CEST] <waqas> images are changing with zoom in and out i think [05:20:53 CEST] <johnjay> like the 3d effect at the start? [05:21:15 CEST] <waqas> thats just video [05:21:26 CEST] <waqas> i will provide the video [05:21:34 CEST] <johnjay> oh yeah i see the zoom effect now [05:21:43 CEST] <johnjay> it's what made that part with the door so creepy [05:21:51 CEST] <johnjay> like someone's gonna jump out of that door any second [05:21:59 CEST] <SpicySalt> is it just me or x265 is slow? [05:22:19 CEST] <johnjay> idk why but when it zooms, it's always into something like a door or shower [05:22:26 CEST] <johnjay> giving a creepy "this is the burglar cam" vibe lol [05:24:24 CEST] <johnjay> anyways waqas good luck with your project [05:24:41 CEST] <johnjay> i can't even compile ffmpeg let alone do things with it. [05:29:52 CEST] <SpicySalt> i have i7 ivy bridge yet x265 is incredibly slow. what do i need? [05:33:37 CEST] <c3r1c3-Win> A ton more cores... Maybe some better settings. [05:58:03 CEST] <Obliterous> Anyone well versed with ffserver available? [06:02:02 CEST] <johnjay> why Obliterous [06:02:04 CEST] <johnjay> what is it you need? [06:04:13 CEST] <Obliterous> Need to figur out how to embed a stream from ffserver into a webpage and have it actually play. [06:05:10 CEST] <Obliterous> the documentation is somewhat less than stellar [06:11:26 CEST] <c3r1c3-Win> Obliterous: that more depends on your video player on the web page. [06:12:23 CEST] <Obliterous> Recomendations? [06:12:53 CEST] <c3r1c3-Win> For what? A player? Stream format? [06:13:05 CEST] <Obliterous> both [06:13:42 CEST] <c3r1c3-Win> Obliterous: Google for open-source html5 video players. Pick one. The one you pick will decide which streaming formats you can use. [06:13:48 CEST] <Obliterous> nothing I have right now is set in stone except for the server hardware. :-) [06:14:44 CEST] <c3r1c3-Win> To answer you question a bit more pointly, No. I don't have any recommended players. I use JWplayer, and want to move away from them. [06:15:21 CEST] <Obliterous> Okay. that gives me a hint right there. [06:18:00 CEST] Last message repeated 1 time(s). [06:18:53 CEST] <c3r1c3-Win> Nothing wrong with the JW-people (nice crew), but I don't want to pay $300+/yr for a player. [06:19:02 CEST] <Obliterous> My target goal is to take a few webcams from around my house and stream them so that my wife can watch birds etc. [06:23:07 CEST] <Obliterous> probably going to either use mp4 or flv & video.js [06:29:04 CEST] <kepstin> SpicySalt: x265 is incredibly slow, yes. if it's too slow for you, you should probably be using x264 instead. [06:29:19 CEST] <SpicySalt> x265 has better compression [06:29:24 CEST] <kepstin> that said, make sure your x265 build has assembly optimizations enabled and working [06:29:35 CEST] <SpicySalt> kepstin how do i do that [06:29:41 CEST] <kepstin> if you tweak x265 to run as fast as x264, it no longer has better compression :/ [06:30:06 CEST] <c3r1c3-Win> ^ [06:30:15 CEST] <kepstin> the way x265 works is that if you wait longer than x264, you can get better compression than the slowest x264 preset [06:30:46 CEST] <kepstin> (well, ignoring placebo because nobody should use placebo) [06:50:48 CEST] <Obliterous> Hrmmmmm [06:51:04 CEST] <Obliterous> new error now that I'm using video.js [06:51:42 CEST] <Obliterous> ' Error writing output header for stream 'pine.mp4': Invalid argument' [07:47:37 CEST] <JohnDoe_71Rus> https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20inp… if i have only 2 sources. is the way set dummy stream? [08:16:04 CEST] <JohnDoe_71Rus> ok/ 2 sources. what is wrong? http://paste.ubuntu.com/24813694/ [08:56:05 CEST] <JohnDoe_71Rus> fix some error mesage. command http://paste.ubuntu.com/24813825/ log http://paste.ubuntu.com/24813827/ no translation ( [09:03:13 CEST] <JohnDoe_71Rus> without -y option no error, but no translation too [09:16:57 CEST] <Remi73> Hi there, I have a question concerning h264 decoder & hw acceleration. Is it possible to build libavcodec without any software decoding capability (for h264), but with the hardware acceleration capability ? [09:17:22 CEST] <Remi73> My conclusion after trying different configuration is that is not possible to use hwaccel with h264 decoder disable, is it right ? [14:08:32 CEST] <Tatsh> :) alias ffhwsupported='for i in encoders decoders filters; do echo $i:; ffmpeg -hide_banner -${i} | egrep -i "npp|cuvid|nvenc|cuda|vaapi|vdpau|vda|dxva2|nvdec|qsv"; done' [14:09:02 CEST] <Tatsh> with nvenc, use constqp and -qp <some number>, like 18 [14:09:17 CEST] <Tatsh> it's not extremely predictable especially across different resolutions [14:09:34 CEST] <Tatsh> but it is nice to process progressive 480p at 60x [15:02:29 CEST] <Tatsh> furq, where can i find an nnedi ASIC and use it :P [15:33:58 CEST] <Fyr> guys, I see that Zeranoe's ffmpeg doesn't support Intel QSV. [15:34:05 CEST] <Fyr> does it? [15:34:41 CEST] <Fyr> I have core i7, it would be great if I could use hwaccel here. [15:36:02 CEST] <furq> i'm pretty sure it does [15:36:21 CEST] <furq> that whole fuss about xp support being dropped was because zeranoe builds with libmfx [15:37:38 CEST] <Fyr> furq, how do you make FFMPEG discover Intel CPU? [15:37:45 CEST] <furq> shrug [15:37:52 CEST] <furq> i don't have a cpu with quicksync [15:40:32 CEST] <BtbN> It should just work. [15:40:39 CEST] <BtbN> Everything it needs comes with the Intel Driver [15:41:41 CEST] <Fyr> BtbN, standard Linux driver omits it? [15:42:00 CEST] <BtbN> QSV is mostly a Windows thing [15:42:19 CEST] <BtbN> There is some horrible wrapper to use it on Linux as well, but there is not really a point [15:45:20 CEST] <Tatsh> it's too bad nvidia's deinterlace -deint is pretty limited [15:45:33 CEST] <BtbN> limited? [15:45:39 CEST] <Tatsh> for content coming from VHS it's tolerable for things to be broadcast immediately [15:45:53 CEST] <BtbN> It deinterlaces the video, what else do you want it to do? [15:45:56 CEST] <Tatsh> BtbN, yea it's for really really predictable content [15:46:04 CEST] <Tatsh> similar to yadif [15:46:19 CEST] <Tatsh> otherwise, i'm sticking to nnedi [15:46:32 CEST] <Tatsh> but it is cool that you can crop and resize in the hardware [15:46:48 CEST] <BtbN> last time I tested nnedi it was ridiculously slow to the point of being useless [15:46:55 CEST] <Tatsh> not for me; i can wait [15:46:56 CEST] <Fyr> BtbN, is there a way to combine hwaccels? [15:47:06 CEST] <BtbN> like, 10 seconds per frame slow [15:47:08 CEST] <Tatsh> yea [15:47:10 CEST] <Fyr> for instance, using opencl, h264_nvenc etc. [15:47:11 CEST] <Tatsh> it's 15 fps for me [15:47:15 CEST] <Tatsh> for 480p [15:47:32 CEST] <Tatsh> but that's the difference between really jumpy video and cleaned up video [15:47:47 CEST] <Tatsh> with typical deinterlacing i get extremely jumpy video and that's not what it's like prior to deinterlacing [15:48:30 CEST] <BtbN> There is basically no opencl accel in ffmpeg. [15:48:37 CEST] <Tatsh> nope [15:49:02 CEST] <BtbN> There are two filters that support it, and due to the way it's implemented its usefulness is limited [15:49:04 CEST] <Fyr> BtbN, ok, dxva2, cuvid and h264_nvenc. [15:49:14 CEST] <Tatsh> basically if -deint can't work for your content, use nnedi :) [15:49:18 CEST] <Tatsh> just be prepared to wait [15:49:23 CEST] <Fyr> BtbN, deload the CPU, it's very useful. [15:49:37 CEST] <BtbN> they all use the same decode hardware [15:49:37 CEST] <Tatsh> i'm getting 20X using my GPU's capabilities [15:49:40 CEST] <BtbN> there is no point in mixing APIs [15:49:41 CEST] <Fyr> when I convert via h264_nvenc for my phone, my CPU load ~10%. [15:49:48 CEST] <Tatsh> with -deint adaptive -crop -resize [15:49:52 CEST] <Tatsh> no filters her [15:49:53 CEST] <Fyr> it's very handy. [15:49:53 CEST] <Tatsh> here* [15:49:58 CEST] <Tatsh> no need for scale_npp [15:50:01 CEST] <BtbN> If you are not in any kind of real-time time pressure, there is no point in using a hwaccel [15:50:05 CEST] <Tatsh> it's really nice [15:50:09 CEST] <Tatsh> BtbN, but i do like saving time [15:50:34 CEST] <BtbN> if you are ok with paying with a horrible quality or massive increase in bitrate [15:51:23 CEST] <Fyr> BtbN, sometimes there is a real-time pressure. [15:51:27 CEST] Action: JEEB would rather use some lolfast preset than hwaccel [15:51:29 CEST] <Tatsh> i imagine the function that takes the arguments uses this x format [15:51:40 CEST] <Fyr> right now, I'm converting all the episodes of Desunoto. [15:51:40 CEST] <Tatsh> for crop [15:51:53 CEST] <Tatsh> ffmpeg -y -c:v h264_cuvid -deint adaptive -crop 10x10x20x36 -resize 640x480 [15:52:00 CEST] <Tatsh> adaptive is far better than bob [15:52:05 CEST] <Tatsh> but it's not as fast [15:52:41 CEST] <Fyr> Tatsh, hardware decoding is very fast. [15:52:47 CEST] <BtbN> keep in mind you have to manually specify the input framerate for adaptive deint [15:52:47 CEST] <Tatsh> yup [15:52:48 CEST] <Fyr> and really helpful. [15:52:56 CEST] <Tatsh> BtbN, i do? [15:52:57 CEST] <BtbN> Decoders in ffmpeg are unable to double the framerate [15:53:02 CEST] <BtbN> So you have to do so manually [15:53:03 CEST] <Tatsh> it's encoding to 29.97 [15:53:13 CEST] <BtbN> for 25i content, it's 50 fps [15:53:37 CEST] <Tatsh> the original content is 29.97 but unfortunately not marked correctly in the metadata [15:53:44 CEST] <Tatsh> Stream #0:0: Video: h264 (High), yuv420p(progressive), 720x480 [SAR 8:9 DAR 4:3], 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc (default) [15:53:52 CEST] <BtbN> just pass the correct doubled framerate as input option to cuvid [15:53:57 CEST] <BtbN> or to ffmpeg itself rather [15:53:58 CEST] <Tatsh> -r ? [15:54:02 CEST] <Tatsh> before the -i [15:54:04 CEST] <Nacht> When using -maxrate. Is that the maximum bitrate you'd like to have ? or is it the maximum difference to that of your set bitrate with b:v ? [15:54:43 CEST] <DHE> Nacht: it's the bitrate of the entire container (mpegts, right?). including the overhead [15:54:48 CEST] <Tatsh> BtbN, what is the correct way to capture interlaced content to h264? [15:54:54 CEST] <Tatsh> my capture card is doing this [15:55:02 CEST] <Tatsh> i want it to correctly mark this content [15:55:05 CEST] <BtbN> what? [15:55:20 CEST] <DHE> using libx264, add parameter -flags ildct to force interlaced encoding [15:55:33 CEST] <Tatsh> i did that [15:55:34 CEST] <Tatsh> -flags +ilme+ildct [15:55:44 CEST] <Tatsh> and the line is yuv420p(progressive) [15:55:51 CEST] <Nacht> DHE: I see. So it differs per container ? (TS vs MP4) ? [15:55:52 CEST] <BtbN> doesn't the card just give it to you correctly? [15:55:53 CEST] <Tatsh> it doesn't matter realy [15:56:18 CEST] <Tatsh> this is the card [15:56:19 CEST] <Tatsh> Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x480, 165722 kb/s, 29.97 fps, 29.97 tbr, 1000k tbn, 1000k tbc [15:56:43 CEST] <DHE> Nacht: only mpegts supports muxrate. it is designed for real-time streaming and has a "null" packet concept which allows it to pad the stream to meet the muxrate [15:57:51 CEST] <Nacht> DHE: I was talking about maxrate, not muxrate [15:58:32 CEST] <DHE> oh, oops my bad... [15:58:41 CEST] <Tatsh> BtbN, when i use -r 30000/1001 before -i after -deint, the video is extremely jumpy [15:58:51 CEST] <DHE> okay. so maxrate specifies the bitrate at which the bitrate buffer is filled [15:58:51 CEST] <Nacht> np, I can understand the error :) [15:59:14 CEST] <Tatsh> but without it, the video looks really good [15:59:19 CEST] <Tatsh> and the framerate is 30 [15:59:22 CEST] <Tatsh> 29.97 [15:59:42 CEST] <BtbN> if you deinterlace it, the framerate doubles. [15:59:53 CEST] <Tatsh> something else is setting the framerate back to 29.97 [15:59:56 CEST] <Nacht> Hm, yeah I read something similar on: https://trac.ffmpeg.org/wiki/Limiting%20the%20output%20bitrate [16:00:03 CEST] <Nacht> I just can't picture yet how that works. [16:00:06 CEST] <Tatsh> ffmpeg -y -c:v h264_cuvid -deint adaptive -crop 10x10x20x36 -resize 640x480 -i capture/raw2/to-cut-willboro-unknown.mkv -c:v h264_nvenc -rc constqp -qp 25 -pixel_format yuv420p -ss 00:06:00 -profile:v high -level 4.1 -aspect 4/3 out2.mp4 [16:00:12 CEST] <BtbN> 30000/1001 looks like 29.97 to me [16:00:14 CEST] <DHE> in VBV mode, there's a buffer (of size -bufsize) that all encoded frames are pulled from. if an encoded frame is 80 kilobytes, it drains 640,000 bits from the buffer. if it's at 30fps and the mitrate is 3 megabits, the buffer will be filled by 100,000 bits. [16:00:28 CEST] <DHE> s/mitrate/maxrate/ [16:00:30 CEST] <Tatsh> for some reason if i specify -r 30000/1001 the output is different [16:00:33 CEST] <DHE> (max bitrate) [16:00:50 CEST] <Tatsh> like this [16:01:02 CEST] <Tatsh> ffmpeg -y -c:v h264_cuvid -deint adaptive -crop 10x10x20x36 -resize 640x480 -r 30000/1001 -i ... [16:01:18 CEST] <Nacht> I see [16:03:09 CEST] <Nacht> Then another related question, but have you ever seen a videoplayer crashing due to the video having high spikes in bitrate even tho the average is still normal ? (15Mbit vs 305Mbit spikes) [16:03:48 CEST] <JEEB> I mean, if it's some embedded plastic thing it might just not have the buffer for the spike if it goes over its VBV/HRD capabilities [16:03:51 CEST] <DHE> I can't say I've heard of that specifically, but that is a HUGE spike... [16:04:13 CEST] <Nacht> Yeah, it sure is [16:04:55 CEST] <Nacht> It's a player running on a GearVR, which crashes saying it doesn't have enough resources, and that wouldn't suprise me with those numbers [16:04:58 CEST] <Tatsh> i'm pretty comfortable with nvenc i can switch over from libx264 now :) [16:05:05 CEST] <Nacht> Sad thing is, it doesn't always crash. Just occasionally [16:05:16 CEST] <Tatsh> still have to encode and encode to check quality though [16:05:40 CEST] <Nacht> I'm experimenting with 2pass encoding icm maxrate/bufsize [16:05:54 CEST] <DHE> how long does this spike last? is it just one frame? keyframes can be large [16:05:55 CEST] <BtbN> 2pass only makes sense for cbr [16:06:24 CEST] <Tatsh> BtbN, if you are trying to stream live how do you maintain the bitrate then? [16:06:34 CEST] <BtbN> not with 2pass [16:06:41 CEST] <BtbN> How would that even work with live content? [16:06:42 CEST] <DHE> I use x264 with a cranked read-ahead when 2-pass isn't possible [16:06:56 CEST] <Tatsh> that's what i'm thinking [16:06:59 CEST] <DHE> it's loosely like 2-pass, but only sees a few seconds into the future (at best) [16:08:31 CEST] <furq> 14:46:48 ( BtbN) last time I tested nnedi it was ridiculously slow to the point of being useless [16:08:41 CEST] <furq> when people say nnedi they normally mean with vapoursynth [16:08:41 CEST] <Nacht> We're using a HLS VOD container. [16:08:50 CEST] <furq> which is still slow but is at least frame multithreaded [16:09:21 CEST] <Nacht> But judging from your reactions, I should really read more upon the technique of 2 pass [16:09:37 CEST] <BtbN> you do the entire encode twice [16:09:44 CEST] <BtbN> so you know in advance where the video will need more bitrate [16:10:05 CEST] <BtbN> it's only ever useful if you target a specific filesize [16:10:08 CEST] <Tatsh> so like -deint adaptive is okay but nnedi is still better [16:10:23 CEST] <Tatsh> not jumpy but still blocky in some areas of this content i have [16:10:40 CEST] <Nacht> Yeah I was hoping to use 2pass to reduce the giant spikes in bitrate. But hearing what you say wouldn't really help with that, it could even create more spikes [16:10:50 CEST] <DHE> Nacht: 2-pass helps, but it depends on the situation. using -b and -maxrate at the same value will help a bit. [16:10:56 CEST] <BtbN> if you don't want spikes, use a small vbv buffer size [16:10:59 CEST] <DHE> Nacht: select a sane bufsize [16:11:04 CEST] <furq> Nacht: are you setting -bufsize [16:11:06 CEST] <Tatsh> https://i.imgtc.com/nOGQoNX.png see the airplane figure on the right [16:11:10 CEST] <Tatsh> all blocky [16:11:16 CEST] <Nacht> I was experimenting with both [16:11:31 CEST] <BtbN> that's not blocky, that just not properly deinterlaced. [16:11:37 CEST] <DHE> I would start with a bufsize of about 1/2 the bitrate. also make -b and -maxrate identical [16:11:42 CEST] <Nacht> I luckely found an idle server with lots of power, cause my poor laptop was slowly dying. It's in HEVC as well :/ [16:11:44 CEST] <DHE> see how that turns out [16:11:50 CEST] <DHE> oh... oh dear... [16:12:04 CEST] <Nacht> Pushing a load of 30 on my server is funny tho :) [16:12:34 CEST] <Nacht> 2048x2048 HEVC, always fun :) [16:12:53 CEST] <DHE> pfft. my video encoder server was at load average of 120.00 most of yesterday [16:12:59 CEST] <Tatsh> if i encode with bob it will be intolerable [16:14:20 CEST] <Nacht> Jeez, how much cores does it have ? [16:14:26 CEST] <Tatsh> but at least with bob it is not so blocky [16:14:54 CEST] <DHE> Nacht: 2 sockets * 20 cores each * 2 (hyperthreaded) = 80 total threads [16:15:31 CEST] <Nacht> Niiice :D [16:15:49 CEST] <furq> not bad [16:15:56 CEST] <furq> you might be able to run nnedi in realtime on that [16:16:32 CEST] <DHE> maybe. but I have cgroups set up to limit any given process to 1 socket for memory latnecy reasons [16:16:42 CEST] <DHE> ... [16:17:25 CEST] <Tatsh> BtbN, there's no perfect live deinterlace filter [16:17:28 CEST] <Tatsh> :| [16:17:33 CEST] <furq> s/live// [16:17:49 CEST] <Tatsh> nnedi does wonders at the extreme cost of time [16:17:54 CEST] <Nacht> Another question, but do you lads know why, when I use FFprobe on HEVC, I don't get any codec_picture_numbers when using -show_frames ? Is it's a HEVC thing ? [16:18:00 CEST] <Tatsh> -deint bob messes up colours for me not sure why [16:18:11 CEST] <Tatsh> -deint adaptive works well most of the time but i gave an example where it didn't work [16:18:30 CEST] <Tatsh> yadif is similar in this regard [16:18:41 CEST] <furq> are you sure that's not bff being treated as tff or something [16:18:59 CEST] <Tatsh> well with -deint on nvenc i have no way to specify bff or tff [16:19:03 CEST] <furq> that floating line at the bottom of the plane looks pretty suspect [16:20:12 CEST] <Tatsh> idet output not so great on this input [16:20:15 CEST] <Tatsh> [Parsed_idet_0 @ 0x10d2280] Repeated Fields: Neither: 8600 Top: 119 Bottom: 136 [Parsed_idet_0 @ 0x10d2280] Single frame detection: TFF: 562 BFF: 5519 Progressive: 56 Undetermined: 2718 [Parsed_idet_0 @ 0x10d2280] Multi frame detection: TFF: 777 BFF: 7993 Progressive: 0 Undetermined: 85 [16:20:33 CEST] <Tatsh> basically it seems it's BFF [16:20:36 CEST] <furq> that sure looks like bff [16:22:46 CEST] <Tatsh> yea and unfortunately cuvid.c is hard-coded to use tff if it encounters an interlaced frame [16:22:57 CEST] <Tatsh> https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/cuvid.c#L624 [16:23:12 CEST] <BtbN> no it's not. [16:23:23 CEST] <furq> yeah that's hardcoded to use whatever the frame flags say [16:23:51 CEST] <furq> i think you can remux and rewrite those? not sure [16:23:53 CEST] <Tatsh> so is it possible the original video has frames badly marked? [16:24:28 CEST] <Tatsh> i encoded it with -flags +ilme+ildct [17:25:54 CEST] <kepstin> Tatsh: you can try using -vf setfield=bff - if that fixes it, then yeah, the frames are incorrectly marked as tff rather than bff. [17:26:31 CEST] <kepstin> but yeah, I don't think ffmpeg can edit that without re-encoding [17:31:12 CEST] <furq> isn't the deinterlacing done by cuvid/nvdec/whatever it's called this week [17:31:19 CEST] <furq> so presumably before it gets to the filterchain [17:36:23 CEST] <kepstin> no idea. I mean, I suppose you should be able to tell the hwaccel not to deinterlace and do it in software, maybe? [17:38:54 CEST] <furq> i think he specifically wants nvdec's deinterlacing for speed [17:38:55 CEST] <Filystyn> so anywone knows the answer? [17:39:06 CEST] <Filystyn> to my question [17:39:21 CEST] <furq> yes, but i don't know what the question is [17:39:46 CEST] <Filystyn> ok [17:39:50 CEST] <Filystyn> Im going to ask now [17:39:57 CEST] <Mavrik> We're ready! [17:41:09 CEST] <Filystyn> When i use the C api ffmpeg function to count the metadata : printf( "%d", av_dict_count( av_formcont->metadata ) ); I get return 0; [17:41:19 CEST] <Filystyn> Everythign would be ok if there was no metadata the thing is [17:41:44 CEST] <Filystyn> this returns on stderror the metdata and general info av_dump_format( av_formcont, 0, PLAYFILE, 0); [17:42:07 CEST] <Filystyn> on mp3 For example i see the data but on my ogg file i don't like there is some trick to it [17:42:10 CEST] <Filystyn> i can't see [17:42:34 CEST] <Filystyn> the second function always see all metadata [17:43:00 CEST] <Filystyn> the first one doe snot same goes with the reading with [17:43:03 CEST] <Filystyn> while( ( tag = av_dict_get( av_formcont->metadata, "", tag, [17:43:31 CEST] <Filystyn> i did look at source but i miss to udnerstand what it is doing ( dumo.c ) [17:43:52 CEST] <Filystyn> it uses the function i use and does some extra things i totaly don't get [18:08:57 CEST] <Guest1318> We are using the libav api to encode a series of microscope pictures using ffv1 into an mkv container. Question: Is there any support for mkv segments in the api? Where should i start to look? Thanks! [18:24:39 CEST] <DHE> if you mean making multiple small segments, you can either do it yourself or use the 'segment' output format to assist you. [18:26:39 CEST] <Filystyn> where is the fucking suport ;-) [18:29:53 CEST] <Filystyn> ok i got it ;-) [18:31:37 CEST] <Guest1318> thanks for the support =), by small segments i mean chunks of ~1000 images that i would consider logically connected (and fit into RAM when decoded). From https://matroska.org/technical/diagram/index.html i see that there are segments in the container format, that can be chained (as far as i understood). Where should i start looking in the libav for this? [18:32:19 CEST] <Guest1318> Or from the other side: what do you (DHE) mean by "you can do it yourself"? [18:35:00 CEST] <DHE> well if you're getting into the nitty gritty of mkv then that's beyond my level of expertise. I just figured you meant you wanted the video cut into, say, 60 second chunks per file [18:38:01 CEST] <furq> hardly anything supports multi-segment matroska afaik [18:38:27 CEST] <furq> you're probably better off just using the segment muxer [18:54:52 CEST] <Nobliterous> Anyone know the current method to get ffserver to stream webm? its not accepting libvpc as a valid coded even though its installed [18:55:50 CEST] <Guest1318> thanks DHE, i really thought about mkv segments, which i would rather do not want to use (thanks furq). The reason i am thinking about this, is that i would like to implement some sort of random access, that would allow me to decode e.g. images 1000-1100 without decoding the ones before or after. Am i right to assume that the MetaSeek part of the mkv spec could help me there? Is this supported by libav? [18:57:06 CEST] <DHE> mkv-specifics aside, video playback requires starting at a key frame. seek accuracy is only as coarse/fine as how often keyframes come. so set AVCodecContext->gop_size to how many frames you want your seek accuracy to be (worst case) [18:57:23 CEST] <DHE> actually you said ffv1, so that's a nonissue [19:03:40 CEST] <Filystyn> what is this https://ffmpeg.org/doxygen/3.0/structAVFormatContext.html#a58c8c4d0ea974e0f… [19:03:46 CEST] <Filystyn> what is that number of programs? [19:05:14 CEST] <DHE> some formats (notably mpegts) support carrying multiple videos at once. eg: an over-the-air broadcast near where I live carries both the HD and SD version of the same channel on the same frequency. ffmpeg will report 2 programs on that feed [19:05:44 CEST] <Filystyn> ihm [19:07:09 CEST] <BtbN> OTA broadcasts usually have 10+ channels in one mpegts mux [19:23:01 CEST] <Guest1318> What i would like to do is fast (frame accurate) seeking in a fully sequential stream of images. Where should i begin , trying to understand how seeking is implemented in ffmpeg (i do not see an obvious example for it in the doc/examples folder)? [19:29:15 CEST] <DHE> BtbN: depends on the region I guess. around here it's usually MPEG-2 1080i at 15+ megabits. with a QAM throughput of around 19 megabits that's the end of it. [19:34:54 CEST] <kepstin> Guest1318: you have to remember that video codecs are designed so that you don't have to sequentially decode all of the images in a stream - instead you only have to start at a keyframe. A keyframe is the start of a GOP (group of pictures), you can set the length of a gop with the '-g' option in ffmpeg. [19:35:19 CEST] <kepstin> Guest1318: Most file formats (including mkv) automatically index the locations of the keyframes, so a player will jump immediately to the nearest keyframe and start decoding there [19:35:37 CEST] <kepstin> Guest1318: in otherwords you don't need to do anything special, the codecs already do what you want [19:36:23 CEST] <BtbN> DHE, mpeg2 for 1080? wat? [19:36:34 CEST] <JEEB> Japan also does MPEG-2 Video 1080i [19:36:42 CEST] <kepstin> Guest1318: so the only thing you need to do is use an existing player with a frame accurate seek option (e.g. mpv), or make sure to use the ffmpeg seek apis correctly. [19:37:01 CEST] <BtbN> And what kinda small muxes are these, to only contain a single channel? At least on cable/satelite, a single mux is like 50Mbps+ [19:37:03 CEST] <kepstin> satellite providers here in canada are only just finally moving streams from mpeg2 1080i to avc. [19:37:03 CEST] <DHE> BtbN: over the air? yeah.. [19:37:12 CEST] <DHE> ATSC is 18.8 megabit iirc [19:37:16 CEST] <Soni> does ffmpeg strip JPEG EXIF? [19:37:28 CEST] <BtbN> if you tell it to [19:37:42 CEST] <thebombzen> If you want to force it you could use -map_metadata [19:37:52 CEST] <thebombzen> but it should by default map EXIF data [19:38:13 CEST] <Soni> does discord strip EXIF? [19:39:13 CEST] <thebombzen> Yes and no. Discordapp re-encodes a smaller file size preview of uploaded jpegs that afaik don't have the exif metadata. but if a user clicks "open original" they'll get the original file in their browser undoctoered [19:39:17 CEST] <thebombzen> undoctored* [19:39:22 CEST] <Soni> hmm [19:39:27 CEST] <kepstin> Soni: kind of off topic here, but looks like not https://feedback.discordapp.com/forums/326712-discord-dream-land/suggestion… [19:39:29 CEST] <BtbN> does what strip EXIF? [19:39:34 CEST] <Soni> after -vf transpose it seems ffmpeg strips EXIF [19:40:02 CEST] <thebombzen> Soni: what EXIF metadata are you looking for? the data that says how it is rotated? [19:40:17 CEST] <Soni> no [19:40:18 CEST] <Soni> any [19:40:34 CEST] <Soni> I did `exiv2 <output from ffmpeg>` and it comes out clean [19:40:37 CEST] <thebombzen> what if you use -map_metadata 0 (or whatever input # it is) [19:40:45 CEST] <Soni> the input has exif but the output doesn't [19:41:42 CEST] <Soni> with command `ffmpeg -i photo.JPG -vf 'transpose=<something>' photo-rot.JPG` [19:49:53 CEST] <Nobliterous> Why am I getting this erro with an mpeg4 stream? ::: Error writing output header for stream 'pine.mp4': Invalid argument [19:50:42 CEST] <Soni> anyway glad I don't have location services enabled on my phone anyway [19:54:42 CEST] Action: Obliterous beats on it with a hammer [19:58:19 CEST] <BtbN> Nothing4You, is your output seekable? [19:58:38 CEST] <BtbN> Obliterous, ^ [19:58:39 CEST] <BtbN> ... [19:58:53 CEST] <Nothing4You> mine isn't [19:58:55 CEST] <Obliterous> nope. just s tream from a webcam [19:59:06 CEST] <BtbN> that's your input... [19:59:53 CEST] <Obliterous> ... [20:00:11 CEST] <Obliterous> I honestly have no clue. [20:00:12 CEST] <DHE> have you, say, made a named pipe (mkfifo) and tried outputting to that? [20:01:55 CEST] <Obliterous> here's the pertinent bits of the ffserver.conf : https://pastebin.com/5L3A6cdx [20:02:23 CEST] <Obliterous> trying to use video.js [20:03:18 CEST] <furq> ffserver sucks and nobody uses it [20:04:04 CEST] <furq> if you want to stream to a browser then use the hls muxer directly to serve from the same machine as ffmpeg, or nginx-rtmp otherwise [20:07:15 CEST] <Obliterous> as I'm aggregating multiple cam streames onto one page, I'll investigate ninx-rtmp. Care to point me in the propper direction? [20:07:50 CEST] <furq> https://github.com/arut/nginx-rtmp-module/ [20:11:17 CEST] <Obliterous> I'm going to explore ffserver more before I go down the road of changing webservers [20:11:34 CEST] <furq> you don't need to change web server [20:13:27 CEST] <Obliterous> Sure seems that way from the docs [20:14:34 CEST] <furq> you don't have to use this to serve http [20:14:49 CEST] <furq> you can presumably build nginx without the http module if you want [20:15:13 CEST] <furq> hls just dumps a playlist and a bunch of mpegts fragments in a directory [20:15:19 CEST] <furq> you can serve that directory with whatever you want [20:16:09 CEST] <furq> even if you do want to serve http with it, though, you can just run it on a different port and reverse proxy [20:16:21 CEST] <Obliterous> thats more pain that its worth [20:16:31 CEST] <furq> well yeah there's not much reason to do that [20:17:02 CEST] <Obliterous> as I said. I'll finish exploring the options with ffserver before I head down that road. [20:17:11 CEST] <MrZeus1> Parallelism in C++ using arrays and gcc: https://www.youtube.com/watch?v=Pc8DfEyAxzg [20:17:22 CEST] <furq> i'm pretty confident you're wasting your time [20:17:27 CEST] <furq> ffserver is basically useless [20:17:41 CEST] <furq> it shouldn't even exist any more really [20:47:02 CEST] <thebombzen> Soni: try -map_metadata 0 [20:47:04 CEST] <thebombzen> see what that does [20:47:15 CEST] <thebombzen> that intentionally maps metadata from input 0 [20:47:19 CEST] <thebombzen> if that doesn't work, that seems like a bug [21:09:16 CEST] <Tatsh> BtbN, seen tis option? [21:09:18 CEST] <Tatsh> this* [21:09:19 CEST] <Tatsh> -drop_second_field <boolean> .D.V.... Drop second field when deinterlacing (default false) [21:09:39 CEST] <BtbN> why would you want that? [21:09:40 CEST] <Tatsh> not sure yet what it really means, but it's false by default [21:09:52 CEST] <BtbN> it drops the second field. [21:10:15 CEST] <Tatsh> you were saying before that the framerate doubles but i'm not experiencing that [21:12:14 CEST] <BtbN> well, if you drop the second field, it obviously doesn't anymore. [21:30:32 CEST] <shamb> Hey, I am attempting to encode an audio file in chunks to aac. When I come to concat them together, the audio gradually goes out of sync. It seems like the concat format is not taking into account the audio priming at the start of each chunk. Is there anyway around this? [21:30:36 CEST] <shamb> thanks! [21:34:43 CEST] <furq> use the demuxer? [21:34:55 CEST] <kepstin> yeah, you won't be able to do anything about that without decoding and re-encoding the audio [21:36:12 CEST] <kepstin> it's not just the priming at the start, but also the padding to full frame sizes at the end, etc. [21:36:53 CEST] <shamb> That's what I was worried about - I am trying to distribute the audio encoding across servers by chunking - I guess this is not possible [21:37:55 CEST] <kazuma_> shamb same issue with copy /b ? [21:37:57 CEST] <kepstin> audio encoding's usually so much faster than video that it's not a problem - you could have 1 server do audio, then n-1 doing video chunks, for example [21:38:12 CEST] <kazuma_> ffmpeg concat always gives me sync issues, but mkvmerge and copy /b does not [21:38:14 CEST] <kazuma_> i don't know why [21:39:05 CEST] <shamb> I have tried mkvmerge but found the same problem - will give it another go though [21:40:00 CEST] <shamb> #kepstin thanks for the info, I see the argument that audio won't need this, but for our user case, encoding audio as a whole is still too long [21:40:13 CEST] <kepstin> mkvmerge's append mode might work, but I think it does it by timestamp manipulation in the container?, so I'm not sure how well it'll survive a transcode [21:40:23 CEST] <kazuma_> copy /b file1.aac + file2.aac +file3.aac output.aac [21:40:27 CEST] <kazuma_> worth a try [21:41:10 CEST] <kepstin> kazuma_: that should provide equivalent results to the ffmpeg concat format (except that ffmpeg can take aac in containers, rather than just raw aac) [21:42:32 CEST] <kepstin> of course, if you're only doing a small number of concats, the issue's probably not super noticable - i'd think a typical error of around 20-40ms? But if you have a lot, that adds up :/ [21:45:17 CEST] <shamb> yep, have taken that into consideration [21:45:23 CEST] <kepstin> and there'll be audible glitches on the joins no matter what you do :/ [21:46:27 CEST] <kepstin> since without re-encoding you can only cut on frame boundaries, which are ~23ms in aac(a)44.1KHz [21:47:43 CEST] <BtbN> if you make sure your segments are exactly on frame boundaries, it should be fine [21:47:47 CEST] <BtbN> But that'll be hard [21:48:12 CEST] <kepstin> well, you have to make sure you know exactly how many samples the encoder adds as priming to get that right :/ [21:48:33 CEST] <kepstin> and you'd probably want to do overlapping segments, then cut a few frames off the start and end [21:48:39 CEST] <kepstin> might be ok then? [21:49:48 CEST] <kepstin> it would be hard work just cutting up the audio to encode for that, depending on whether you can seek with sample accuracy in your input file :) [21:50:29 CEST] <shamb> but then I still have the issue with concatenating afterwards right? [21:52:30 CEST] <kepstin> if you know that the encoder uses e.g. 2112 priming samples, then you can do something like encode the audio starting 960 samples early, ending 1024 samples late, then drop the first three frames and last frame before concatenating [21:52:49 CEST] <kepstin> and it'll probably turn out okish [21:53:10 CEST] <Soni> thebombzen: well the idea is that I *don't* want metadata [21:53:31 CEST] <Soni> thebombzen: also ffmpeg fails to use exif rotation [21:54:19 CEST] <kepstin> shamb: and each segment obviously has to be a multiple of 1024 samples long [21:54:30 CEST] <kepstin> (plus the extra on the start+end) [21:54:38 CEST] <thebombzen> Soni: it doesn't "fail" it just doesn't do it unless you tell it to [21:54:46 CEST] <thebombzen> Soni: -map_metadata -1 refuses to map metadata [21:54:59 CEST] <thebombzen> that's how you intentionally decline to map metadata [21:55:43 CEST] <thebombzen> in fact, you could even do ffmpeg -i input.jpg -map_metadata -1 -c copy output.jpg to strip all metadata, but there's probably a better way of doing that than ffmpeg.c [21:56:15 CEST] <Soni> thebombzen: the flag is called -noautorotate IIRC, the default is to rotate [21:58:00 CEST] <thebombzen> how is that failing [21:58:04 CEST] <thebombzen> that sounds like succeeding [21:58:37 CEST] <thebombzen> or rather, what exactly are you trying to do? [22:00:05 CEST] Action: kepstin thinks about it a bit more, and would actually suggest starting 1984 samples early if the encoder uses 2112 priming samples, so there's at least 1024 real samples in the overlap window [22:15:01 CEST] <alexpigment> hey guys. does anyone know if the intel quick sync encoder is the same for all CPUs within a given generation of chips? [22:15:32 CEST] <alexpigment> in other words, could i get a $40 Celeron G3930 and expect the same H264 encoding performance as a 7700K? [22:22:47 CEST] <c3r1c3-Win> alexpigment: Depends and generally No. [22:24:20 CEST] <alexpigment> so is the graphics card number a more accurate factor in determining speed? [22:24:49 CEST] <alexpigment> for example, if the 7700K has an HD 630 chip, I could find the cheapest CPU with an HD 630 and get the same performance? [22:24:53 CEST] <jkqxz> The things which matter are, roughly in order of importance, (a) number of video codec blocks, (b) memory bandwidth, (c) graphics clock speed, and (d) number of graphics execution units. (c) and (d) are both less, so it will be noticably slower but not hugely. [22:25:18 CEST] <alexpigment> gotcha [22:25:20 CEST] <jkqxz> A 7300 would be the same as the 7700K in all of those, and would give the same performance. [22:25:35 CEST] <alexpigment> i was under the impression that the GPU speed wasn't applicable, so that's good to know [22:26:25 CEST] <jkqxz> It doesn't vary by very much, so while it's significant it doesn't end up making very much difference in reality. [22:28:10 CEST] <alexpigment> ok, thanks for the heads up [22:28:13 CEST] <jkqxz> In terms of chip cost / total codec throughput, the cheapest ones do totally win. (Though you have to be doing something pretty weird for that to be the only constraint.) [22:29:12 CEST] <alexpigment> yeah, i'm really just trying to get a few test systems to measure speed and quality on each generation [22:29:50 CEST] <alexpigment> but you're right in thinking that there could be other bottlenecks [22:29:59 CEST] <alexpigment> due to the number of CPU cores [23:51:33 CEST] <Tatsh> the -crop option for h264_cuvid is not quite the same as crop= filter [23:51:38 CEST] <Tatsh> of course the syntax is different [23:51:44 CEST] <Tatsh> the way it works is different too [23:52:07 CEST] <Tatsh> it appears that if i crop on the top by 2 pixels, this affects the right side by 2 pixels [23:52:34 CEST] <Tatsh> i wrote a function to convert between the normal syntax and nvenc and i'm not getting good results [23:52:51 CEST] <Tatsh> it seems like i have to add 2 pixels for each side for what i would use with the normal crop filter [23:53:10 CEST] <BtbN> how would the top affect the right? [23:53:44 CEST] <Tatsh> i'm using nnedi for deint and every time i use my crop value i got from mpv, and then i do the math to convert it, i get green sides [23:54:00 CEST] <Tatsh> in particular, green right and bottom [23:54:18 CEST] <BtbN> Can't reproduce that. [23:54:43 CEST] <Tatsh> for right now i'm sticking to nnedi with regular crop as there is no speed difference [23:55:05 CEST] <BtbN> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/cuvid.c;hb=HEAD#l… [23:55:10 CEST] <BtbN> that's all the cropping code there is [23:55:10 CEST] <Tatsh> the built-in deint leaves much to be desired [23:55:45 CEST] <BtbN> complain to nvidia about that. [23:57:09 CEST] <Tatsh> yea [23:57:16 CEST] <Tatsh> they need nnedi implemented in hardware so badly [23:57:20 CEST] <Tatsh> or qtgmc [23:57:22 CEST] <Tatsh> i'll take either one [23:58:33 CEST] <Tatsh> hmm [23:58:41 CEST] <Tatsh> with ffplay i can preview with nnedi and see the effect [23:58:49 CEST] <Tatsh> so i guess i should crop that way instead [00:00:00 CEST] --- Sat Jun 10 2017

1 0

ffmpeg-devel.log.20170608
by burek 09 Jun '17

09 Jun '17

[00:01:03 CEST] <blue_misfit> rcombs, yeah it works in VLC lol. Avid is writing the outputs fyi via "same as source" export when cutting mixed media sources [00:03:37 CEST] <wm4> blue_misfit: does that use edit lists? [00:03:48 CEST] <wm4> or is it a single track that changes codecs? [00:03:58 CEST] <wm4> (not too familiar with mp4 structure) [00:04:09 CEST] <blue_misfit> well, it's real mov, not mp4 ;) [00:04:14 CEST] <blue_misfit> but yeah there is an edit list [00:04:22 CEST] <blue_misfit> I can't really tell whether it describes the codec switch tho [00:04:25 CEST] <blue_misfit> don't have the tools to tell [00:04:37 CEST] <blue_misfit> premiere says "track contains 2 types of video" when you get properties for it [00:05:39 CEST] <wm4> l-smash boxdumper could help [00:07:54 CEST] <blue_misfit> wm4, good call [00:08:02 CEST] <blue_misfit> I'll do taht next time I have access to the file or one like it [00:39:09 CEST] <nevcairiel> wm4: i would be more worried about lavf surviving the switch of the codec in the same track, I agree that lavc should be left out of this [01:11:18 CEST] <atomnuker> jamrial_: seems like the decoder API restructuring commit broke opus decoding for someone using the old API [01:14:16 CEST] <atomnuker> (told him to update to the new API or submit a trac issue, I'm suspecting the issue might be on his side) [01:15:09 CEST] <nevcairiel> i still use the old api and opus decodes fine [01:19:43 CEST] <BBB> atomnuker: do you have any insights on how av1 standardization is progressing? [01:25:10 CEST] <tmatth> BBB: last I heard, bitstream freeze for this fall? [01:25:18 CEST] <kevmark> [17:42:47] <fiberbaby> https://www.ssllabs.com/ssltest/analyze.html?d=ffmpeg.org [01:25:18 CEST] <kevmark> [17:42:54] <fiberbaby> https://blog.qualys.com/ssllabs/2017/04/05/ssl-labs-distrusts-wosign-and-st… [01:25:18 CEST] <kevmark> [17:43:04] <fiberbaby> how about using Let's Encrypt [01:25:18 CEST] <kevmark> Not saying Let's Encrypt shouldn't be used but that "T" rating isn't worth very much when Chrome 60 and Firefox 53 doesn't complain at all about the current certificate [01:26:34 CEST] <cone-148> ffmpeg 03Nedeljko Babic 07master:c8e7fc8d9ad7: MAINTAINERS: Add Manojkumar Bhosale for MIPS, remove myself. [01:33:28 CEST] <kevmark> There's also this if Let's Encrypt is off the table. I've gotten a free open-source cert from them before. https://www.globalsign.com/en/ssl/ssl-open-source/ [01:43:04 CEST] <KGB> [13FFV1] 15dericed opened pull request #71: add iana considerations section (06master...06iana-considerations) 02https://git.io/vHi4n [03:00:16 CEST] <rcombs> kevmark: I would not be surprised if Google and Mozilla moved towards untrusting existing certs, and they're already not trusting new ones (so this will have to be changed before renewal) [03:01:07 CEST] <kevmark> rcombs, from StartCom? [03:01:16 CEST] <rcombs> yes [03:02:23 CEST] <kevmark> According to this https://security.googleblog.com/2016/10/distrusting-wosign-and-startcom.html the current cert on ffmpeg.org should be untrusted, but it's not. I'm guessing they walked back on the timeline? [03:03:04 CEST] <kevmark> Oh, sorry, I didn't see the issue date on the cert [03:03:53 CEST] <rcombs> if StartCom/WoSign are caught backdating any more certs they'll be purged immediately [03:03:54 CEST] <kevmark> So it sounds like we'd have until our cert goes up in April 2018, which is pretty generous. [03:04:12 CEST] <rcombs> and they've been caught doing just that in the past [03:04:14 CEST] <kevmark> unless, they yep [03:04:19 CEST] <rcombs> so I wouldn't call that timeframe "safe" [03:04:24 CEST] <kevmark> Fair [03:04:51 CEST] <kevmark> At least(?) ffmpeg.org isn't putting out HSTS [03:05:08 CEST] <rcombs> (they've been given an ultimatum with heavy consequences, so they _probably_ won't, but it's no guarantee) [03:05:56 CEST] <kevmark> Gotcha. Well as a FOSS project there'd by no issue getting a cert from a reputable provider (like GlobalSign) or just using LE [03:07:39 CEST] <rcombs> yeah [03:07:52 CEST] <rcombs> I'd lean towards LE with autorenew [03:08:32 CEST] <KGB> [13FFV1] 15dericed opened pull request #72: extend intro and add summary of version history (06master...06expand-intro) 02https://git.io/vHi2i [03:08:59 CEST] <kevmark> The problem I've personally experienced with LE's short expiry tickets is when using it with multiple services. [03:10:44 CEST] <kevmark> I believe their tool for automating Apache/nginx deployments works very well but I've had to spend a lot of time creating glue shell scripts that copy the certs over to where they're needed elsewhere. Like mail [03:11:21 CEST] <kevmark> I'm not sure how big of an issue that could be for ffmpeg.org but I'd just put my two cents in there [03:11:43 CEST] <kevmark> s/short expiry tickets/short expiry certs [03:12:14 CEST] <rcombs> well it's either write some shell scripts once, or do it manually every time expiry comes near [03:12:48 CEST] <kevmark> Or, if you're like me, write some shell scripts once, test them, have them break in three months (somehow), and write them again [03:12:54 CEST] <rcombs> hah [03:13:39 CEST] <kevmark> When the site goes down three months later, people complaining about the cert being expired, and you thinking to yourself "damn it the script I wrote explicitly so this would never happened didn't work" [03:14:48 CEST] <kevmark> Although I'm sure someone is working on getting it integrated into systemd as we speak. And that will solve all these problems. [03:17:47 CEST] <rcombs> clearly [05:01:31 CEST] <cone-141> ffmpeg 03Vittorio Giovara 07master:35c76f2e138b: vf_colorspace: Add support for smpte248 color primaries [10:30:27 CEST] <kierank> blue_misfit: that's insane [10:30:45 CEST] <JEEB> I wonder how it signals the switch within the same track [10:34:25 CEST] <nevcairiel> there is probably some mov magic to indicate which sample the codec changes [11:38:01 CEST] <durandal_1707> wm4: what you mean by directly loading coefficients? tge method i pick is most friendly to the user [11:42:20 CEST] <kierank> Interesting, hevc software encode on macs [11:42:28 CEST] <kierank> I wonder which hevc encoder they used [11:42:41 CEST] <wm4> durandal_1707: nothing about this is user friendly... [12:18:58 CEST] <durandal_1707> wm4: sorry but it cant be user friendly, thought once you set it up you can forget about it [12:24:33 CEST] <wm4> "it cant be user friendly" that's quite a statement [13:26:21 CEST] <durandal_1707> wm4: its very friendly, if you want dumb friendly then compile sofalizer [13:27:54 CEST] <durandal_1707> one just loads files for each loudspeaker he wants to use and than he can forget it [13:58:47 CEST] <durandal_1707> wm4: are you strictly against filter? [13:59:19 CEST] <wm4> well, not strictly, but I don't think it's overly useful/usable [14:05:40 CEST] <durandal_1707> just because it needs little complicated options [14:06:00 CEST] <RiCON> as long as the examples are clear enough [14:06:55 CEST] <durandal_1707> also im reclutant to apply libmysofa patch because library is buggy and only latest git master is working correctly [16:45:59 CEST] <cone-462> ffmpeg 03Vittorio Giovara 07master:f7f60749e0c9: vf_colorspace: Add support for jedec p22 primaries [17:37:57 CEST] <cone-462> ffmpeg 03Michael Niedermayer 07master:4e3ab1a5c12f: avcodec/ac3dec_fixed: Fix multiple runtime error: signed integer overflow: -39271008 * 59 cannot be represented in type 'int' [17:37:58 CEST] <cone-462> ffmpeg 03Michael Niedermayer 07master:a3b5b60bdf45: avcodec/indeo4: Check remaining data in Pic hdr extension parsing code [17:56:24 CEST] <hexianren> Hi, All,I would like to contribute to ffmpeg,but I don't know how.Could you allocate some easy job for me, and I will try my best to finish it. Thanks [18:12:03 CEST] <durandal_1707> ok to push gremlin stuff? [18:12:15 CEST] <tdjones> Is it intended for fate to fail if the encoder writes out to stdout? It seems like a mistake to disguise this as a functional error: "stddev: |6450.74 - 296| >= 30". I may just be misunderstanding the output from fate. [18:22:40 CEST] <jamrial> hexianren: maybe look at https://trac.ffmpeg.org/report for bug reports and feature requests, or search for TODO and FIXME lines in the codebase [18:22:49 CEST] <atomnuker> tdjones: yep, you can't write to stdout [18:41:38 CEST] <nevcairiel> Logging should go to stderr [18:42:33 CEST] <wm4> logging should use av_log [19:08:27 CEST] <tdjones> I was just quickly testing something on a different machine and didn't want to change compile settings for debugging symbols [19:09:29 CEST] <tdjones> Felt like there should've been a message about incorrectly outputting information rather than displaying it as if there was an error in the audio file output [19:11:08 CEST] <tdjones> But I was mistaken [20:05:07 CEST] <cslcm> Hi! Does anyone know if any progress has been made on a vp9_qsv encoder? [20:07:01 CEST] <kierank> i heard rumours about patches [20:15:17 CEST] <atomnuker> durandal_1707: how do you produce the wav coefficient files? [20:15:56 CEST] <durandal_1707> atomnuker: hrtf ones? [20:16:54 CEST] <durandal_1707> they are recorded same way as room impulse response filters [20:17:35 CEST] <durandal_1707> except subject is placed in special room wearing microphones in ears [20:18:10 CEST] <durandal_1707> subject can be human or artificial head [20:20:16 CEST] <atomnuker> no, I meant from existing formats (e.g. sofa files) [20:21:47 CEST] <durandal_1707> from sofa they can be only extracted, each wav takes exactly one loudspeaker position [20:22:27 CEST] <durandal_1707> there are other formats [20:23:16 CEST] <durandal_1707> do you want to try specific sofa file with this filter? [20:26:47 CEST] <atomnuker> no, I just think that you should add a small program to do that into e.g. ./tools/ [20:27:26 CEST] <atomnuker> also possibly for other filters which take coefficients as streams [20:30:13 CEST] <cone-462> ffmpeg 03Aman Gupta 07master:a32a6b4201dc: lavc: add mpeg2 mediacodec decoder [21:43:38 CEST] <J_Darnley> How many of you have built NASM releases from its Git repo? [21:46:05 CEST] <J_Darnley> Well clearly what's in the git repo isn't the same as the release tarball because the tarball builds and the repo doesn't. [21:46:12 CEST] <JEEB> hah :D [21:56:33 CEST] <J_Darnley> Yes. If anyone wants to build old nasm: grab the tarballs, don't use the repo [21:56:52 CEST] <J_Darnley> I've now got 10 years of nasm releases to test with [21:59:16 CEST] <BBB> are there examples of pull-interface filters? [21:59:18 CEST] <BBB> in libavfilter [22:03:01 CEST] <cone-462> ffmpeg 03Paul B Mahol 07master:2336c76b2246: avfilter/af_sofalizer: switch to libmysofa [22:05:16 CEST] <durandal_1707> BBB: wut? [22:06:17 CEST] <BBB> filters that request_frame their input pad instead of input being pushed into it using filter_Frame [22:36:13 CEST] <J_Darnley> Does anyone think requiring nasm-2.11 from 2013-12-31 is too new? [22:36:44 CEST] Action: J_Darnley goes to check yesterday's logs [22:41:08 CEST] <TMM> hi all [22:42:03 CEST] <TMM> I'm reverse engineering the two remaining unknown upcodes for the interplay MVE videos, and I got one of the opcodes completely working. (opcode 0x06 if anyone cares) I believe I have decoding of op 0x10 working too but I think I'm fucking something up with buffer flipping somehow [22:42:50 CEST] <TMM> I've not yet implemented the features in ffmpeg as I found the code a little too daunting to learn it as well as the codec at the same time btw. [22:43:44 CEST] <TMM> This is what the video playing now looks like: https://tmm.cx/nextcloud/s/vpKUOHsBwEvvOml [22:44:02 CEST] <TMM> can someone with more knowledge of video codecs in general please have a look at it and tell me if they see something that is obvious? [22:45:02 CEST] <TMM> if I copy the backbuffer to the front buffer with a flip then things look ALMOST correct [22:45:09 CEST] <TMM> but then there's some other garbage that stays behind [22:45:48 CEST] <TMM> it clearly has something to do with this if you look carefully at the beginning you see that the content of the background kind of blinks between the top 1/4 and the bottom 3/4 [23:13:20 CEST] <J_Darnley> Gramner: I'm applying the x86inc patches from x264. Should AESNI imply any other instructions other than SSE4.2? [23:14:10 CEST] <J_Darnley> Specifically on the "Add some aditional cpuflag relations" patch [23:14:44 CEST] <J_Darnley> jamrial BBB: do you want to give your thoughts on that ^^ [23:15:37 CEST] <BBB> I feel like I may not be knowledgeable enough to answer that (?) [23:15:50 CEST] <BBB> I would say probably not [23:15:54 CEST] <BBB> but Im not sure [23:16:06 CEST] <J_Darnley> understood [23:16:08 CEST] <jamrial> J_Darnley: no, afaik there are cpus with aesni that don't have avx [23:16:38 CEST] <J_Darnley> Yes there are: Celerons and Pentiums have no AVX [23:16:55 CEST] <J_Darnley> New ones I mean [23:19:14 CEST] <J_Darnley> Like this then: aesni implies sse42, avx implies aesni [23:20:57 CEST] <J_Darnley> Maybe I should upstream the aesni define [23:26:20 CEST] <rcombs> J_Darnley: I have some ancient patches for AESNI [23:36:35 CEST] <jkqxz> J_Darnley: Some platforms can disable AES-NI in the BIOS, so nothing can imply it. [23:36:52 CEST] <Gramner> J_Darnley_ aesni should probably imply sse42, but avx does not imply aesni. there are cpus with avx but without aesni. I guess due to market segmentation bullshit reasons [23:37:57 CEST] <TMM> if you want to be sure about that, libvirt has a very complete database of cpu flag combinations for various cpus [23:39:07 CEST] <nevcairiel> we generally check for actual availability reported by the cpu, but the implied relationships are helpful to know to simplify checks [23:40:32 CEST] <TMM> On Interplay MVE, I based my demuxer off of this document: https://multimedia.cx/interplay-mve.txt, it mentions "The rendering process keeps track of the most recent frame in a separate buffer, and uses this double-buffering technique in the common way for animation. The current frame's data is used in the construction of the next frame." I took that to mean that we just have 2 buffers and I just flip them [23:40:39 CEST] <TMM> is that a correct interpretation of that statement? [23:40:51 CEST] <TMM> (the diassembly bears this out as far as I can see btw) [23:41:35 CEST] <nevcairiel> that just describes typical simple inter coded images [23:41:54 CEST] <TMM> I'm not sure what that means [23:42:13 CEST] <nevcairiel> but in our world we wouldnt flip buffers, but just make new ones [23:43:00 CEST] <TMM> it seems that the original player really just swaps two ddraw surfaces [23:43:09 CEST] <TMM> and it draws direct to them [23:43:49 CEST] <TMM> nevcairiel, would you be willing to look at this? https://tmm.cx/nextcloud/s/vpKUOHsBwEvvOml <-- this is what my player does now for the unknown 0x10 frameformat for mve [23:44:26 CEST] <TMM> I'm like... 95% sure that the actual decoding of the frames is doing the exact same thing as the original player does, but something is obviously wrong :) [23:44:58 CEST] <nevcairiel> i dont know anything about that format [23:45:31 CEST] <TMM> I understand, but I was hoping that the pattern of artifacts may hint at a more generic misunderstanding of mine [23:45:48 CEST] <TMM> I'm vaguely hoping someone here will just go "Oh, you're not doing X" or something [23:49:09 CEST] <J_Darnley> Gramner: oh great! [23:49:25 CEST] <J_Darnley> jkqxz: that will change runtime detection [23:49:37 CEST] <J_Darnley> this is compile time stuff [23:51:59 CEST] <jamrial> rcombs: your aesni patches were all committed except the one that made the aes module use aesni, right? [23:52:14 CEST] <nevcairiel> so the most useful one missing? =P [23:52:20 CEST] <jamrial> yeah :p [23:52:57 CEST] <jamrial> rcombs: in any case, J_Darnley is trying to import the avx512 changes from x264 [23:53:12 CEST] <rcombs> ah, avx512, fun [23:53:17 CEST] <J_Darnley> I didn't get that far yet [23:53:47 CEST] <nevcairiel> how many changes did x86inc accumulate again, i feel like we synced a bunch of them not too long ago [23:54:19 CEST] <Gramner> quite a few recently actually [23:54:30 CEST] <Gramner> although most are trivial [23:54:48 CEST] <nevcairiel> only the 2017 changes missing [23:54:51 CEST] <nevcairiel> shouldnt be that much [23:54:52 CEST] <jamrial> i wonder if for avx512 we shouldn't try to go ahead with that aligment patchset in the ml to avoid making everything 64 byte aligned long before we even write the first avx512 function [23:59:01 CEST] <TMM> nevcairiel, at least, based off of that line is my interpretation of just having 2 buffers and swapping them a way to implement this? [23:59:59 CEST] <nevcairiel> if done incorrectly that might violate frame lifetime in avcodec [00:00:00 CEST] --- Fri Jun 9 2017

1 0

ffmpeg.log.20170608
by burek 09 Jun '17

09 Jun '17

[00:01:07 CEST] <Papi> does someone know what to do??? [00:03:00 CEST] <furq> don't use -re [00:03:38 CEST] <furq> also pastebin the command you're using [00:05:46 CEST] <Papi> ffmpeg -i rtmp://127.0.0.1:1935/show/EOKvvX9IgGJYjwa38zLeVUMEVBwHGq925pwVA52aJaIpqdOGI2 -loglevel debug [00:05:56 CEST] <Papi> it's only testing command without output [00:06:12 CEST] <Papi> I expect error there's should be atleast one output [00:06:23 CEST] <Papi> But ffmpeg stuck [00:07:49 CEST] <ChocolateArmpits> Using "-probesize 32" newer versions of ffmpeg successfully open an mpegts input. An older version from 2014 simply closes. Both receive "not enough frames to estimate rate; consider increasing probesize". Is there some kind of difference how each are treated for both versions ? [00:08:28 CEST] <furq> Papi: should work [00:08:32 CEST] <furq> what ffmpeg version [00:09:39 CEST] <Papi> Version 3.2.5-1~bpo8+1 [00:10:06 CEST] <Papi> When I kill ffmpeg with CTRL+C I'll get this: rtmp://127.0.0.1:1935/show/EOKvvX9IgGJYjwa38zLeVUMEVBwHGq925pwVA52aJaIpqdOGI2: Input/output error Exiting normally, received signal 2. [00:16:45 CEST] <Papi> hm? [00:20:27 CEST] <Papi> Sometimes it works and sometimes not [00:20:30 CEST] <Papi> hmmmm [06:08:54 CEST] <furq> is there some difference between -f aac and -f adts [06:09:07 CEST] <furq> other than that you can only mux one and only demux the other [10:16:13 CEST] <Nacht> Anyone has any idea why when I use 'ffprobe -i file.mp4 -show_frames' on a HEVC MP4 files, I get 'coded_picture_number=0' on all frames ? [11:42:50 CEST] <Klojum> Hi guys, I'm having a problem with a Nvidia GT1030 doing encoding. Seems it's not recognized as a NVENC capable device. Using ffmpeg version 3.3.1-1~16.04.york0 which works fine for a Nvidia GTX950 card. Is it "too soon" for the GT1030 and ffmpeg, or is something else blocking my encoding attempts? Let me know if you need more info. Thanks in advance. [11:46:03 CEST] <Klojum> For those interested, here is my encoding attempt: http://paste.ubuntu.com/24806928/ [11:46:19 CEST] <furq> are you sure that card has nvenc [11:46:27 CEST] <furq> it normally isn't on the GT cards [11:47:56 CEST] <Klojum> Hmpf... I may have missed that. I'll dive deeper into that. :-\ [11:48:47 CEST] <DHE> as long as ffmpeg has nvenc and doesn't complain about a version mismatch, it's probably not ffmpeg's fault... [11:49:25 CEST] <furq> nvidia's website is being as loquacious about the matter [11:49:34 CEST] <furq> in that it's denying that there is any such thing as nvenc [11:49:41 CEST] <furq> +as you'd expect [11:49:58 CEST] <DHE> might try checking for the nvidia-branded feature like shadowplay [11:50:00 CEST] <furq> it does have nvdec though which is the only bit that's any good anyway [11:51:14 CEST] <Klojum> Yes, decoding works fine. No 10bit unfortunately under Linux, but in Windows it does work. [11:54:51 CEST] <Klojum> Well, thanks guys so far. I may gonna send the card back. :-| [13:46:08 CEST] <zerodefect> Using the C-API for decoding. Looking to seek back to the beginning of media once I reach the end. Do I need to flush the decoder completely before I start decoding again? [15:00:40 CEST] <zyme> Klojum appearently left, but I was going to ask if he has windows 10, because although I haven't tested it yet I have ffmpeg installed under bash in it's new linux subsystem [15:01:55 CEST] <zyme> It's kinda fun how it can still execute windows commands from it, if only it could go back and forth like one happy enviroment [15:03:01 CEST] <zyme> if a gui or wm would install and run that would probably seem pretty close [15:19:27 CEST] <zyme> whatever it is with windows, when DNS can resolve a LAN name as IPv4 (but not v6), windows can't resolve the v6 address, but when theres no resolution of the name at all (like what happens if I staticly assign the IPv4 address from the device instead of the DHCP server) then it resolves both using (I presume) some local name resolution protocol; [15:23:19 CEST] <zyme> the printers resolving as both right now but probably only because DNS can't resolve it at all atm. [15:29:17 CEST] <Nacht> Anyone know why FFprobe shows 'coded_picture_number=0' on all frames with HEVC files ? Is it due to the codec itself or is it a bug ? [16:42:54 CEST] <Fyr> guys, I'm trying to rip a DVD, ffprobe says: [16:42:54 CEST] <Fyr> Could not find codec parameters for stream 2 (Audio: dts, 0 channels): unspecified sample format Consider increasing the value for the 'analyzeduration' and 'probesize' options [16:43:14 CEST] <Fyr> how to make ffprobe understand DTS 5.1? [16:43:51 CEST] <Fyr> ffprobe version N-86265-gb946bd8 Copyright (c) 2007-2017 the FFmpeg developers [16:50:49 CEST] <alexpigment> do you guys know any simple ways to detect if QSV and NVENC are available? [16:51:20 CEST] <alexpigment> i'm trying to avoid doing a quick smptebars-ish encode and parsing for errors [16:51:40 CEST] <alexpigment> i was hoping for a way to simply check if the encoder is available and have ffmpeg return a yes/no [16:52:41 CEST] <DHE> there are problems with even that. nvenc on most GPUs will only allow 2 sessions and will fail on any further [16:56:20 CEST] <alexpigment> DHE: i will only need it to do one session and can safely assume that other sessions are not being used [17:07:20 CEST] <kms_> can i set paddings in box=1 videofilter? [17:09:51 CEST] <kms_> drawtext [17:16:45 CEST] <fred1807> is there a one liner to wipe all metadata from mp3 files? [17:24:03 CEST] <kepstin> fred1807: note that ffmpeg can't do inplace file tag editing, you're probably better off using a dedicated applicaton (or writing something using e.g. mutagen) [17:25:00 CEST] <kepstin> fred1807: but that said, 'ffmpeg -i input.mp3 -map_metadata -1 -map 0:a -c copy output.mp3' is probably pretty close; it might need an extra option somewhere to tell ffmpeg not to write encoder info metadata tho. [17:27:03 CEST] <Fyr> to delete metadata, I use pipes. [17:27:36 CEST] <celyr> I use pipes to flush my metadata [17:27:43 CEST] <celyr> ok I'll disappear after this one [17:27:56 CEST] <kepstin> adding the '-id3v2_version 0' output option to that will stop it from adding an id2v23 tag with the encoder metadata, fwiw [17:28:27 CEST] <kepstin> (well, i guess the default's actually id3v2.4, but whatever) [17:41:08 CEST] <Fyr> is it possible to use -hwaccel dxva2 and -hwaccel cuvid together? [17:41:56 CEST] <Fyr> something like: [17:41:56 CEST] <Fyr> ffmpeg -hwaccel dxva2 -threads 1 -hwaccel cuvid -c:v h264_cuvid -i file.mkv <...> [17:41:57 CEST] <Fyr> ? [19:08:17 CEST] <Papi> Hello [19:08:23 CEST] <Papi> Can someone help me with this issue? [19:08:24 CEST] <Papi> https://github.com/arut/nginx-rtmp-module/issues/1052 [19:11:30 CEST] <furq> Papi: https://www.johnvansickle.com/ffmpeg/ [19:11:33 CEST] <furq> does it work with those builds [19:13:31 CEST] <Papi> I'll try [19:15:23 CEST] <johnjay> Papi: at least you have nginx with rtmp working. :D [19:15:41 CEST] <johnjay> on windows that was a PITA to get working [19:16:13 CEST] <Filystyn> av_dict_set ( what does it do ? [19:16:29 CEST] <Papi> I don't need only nginx with RTMP :D [19:16:46 CEST] <Papi> I need complex setting for streaming service :D [19:17:54 CEST] <johnjay> i admire your achievements good sir [19:19:05 CEST] <johnjay> also damn furq I still can't get ffmpeg built. T_T [19:19:25 CEST] <johnjay> i'm halfway tempted to download those preinstalled deb images from Mr. Dot-Ninja domain [19:20:02 CEST] <Filystyn> av_dict_set ( what does it do ? [19:24:07 CEST] <Filystyn> Do i even need it for audio [19:25:00 CEST] <klaxa> av_dict_set() sets a dictionary value for a dictionary key [19:25:02 CEST] <DHE> it lets you pass options to codecs, containers, etc that don't fit the existing AVThingtypeContext structures [19:25:31 CEST] <Filystyn> for audio is there soemthign worth trying to pass? [19:25:38 CEST] <Filystyn> or just leave it? [19:25:53 CEST] <Filystyn> How do i know the options? [19:25:59 CEST] <DHE> usually if you have an ffmpeg parameter "-xxx yy" you can do av_dict_set(&dictptr, "xxx", "yy"); // use correct options though [19:26:02 CEST] <Papi> Static build of ffmpeg doesn't work too [19:26:27 CEST] <Filystyn> ok but can you give me exampel of parameter [19:26:29 CEST] <Filystyn> when it has sens? [19:30:21 CEST] <Filystyn> come on guys teach me [19:30:59 CEST] <klaxa> read some example files from doc/examples/ [19:32:21 CEST] <DHE> you want an example of a parameter? check ffmpeg-all. tons of them [19:32:37 CEST] <Filystyn> I am stuck on example but i just don't get where is the option list [19:32:41 CEST] <Filystyn> literaly look [19:32:59 CEST] <Filystyn> this is valid code ? av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0); [19:33:27 CEST] <Filystyn> passing strings seriously? [19:33:34 CEST] <Filystyn> well ptrs to strings [19:33:37 CEST] <Filystyn> but still [19:34:00 CEST] <Filystyn> let me ask you this way may i need this function for AUDIO [19:34:19 CEST] <Filystyn> can it be used with audio and give positive effect [19:34:46 CEST] <johnjay> Filystyn: look at it this way. how much $$$ would you be willing to pay for an answer [19:34:53 CEST] <Filystyn> 0 [19:34:56 CEST] <johnjay> if > 0 then remember it's free [19:35:05 CEST] <johnjay> ok well in that case there's no problems [19:35:08 CEST] <johnjay> you get what you pay for. :D [19:35:25 CEST] <Filystyn> ok ill just wait someone else answers eventualy [19:35:27 CEST] <Filystyn> thx for help [19:35:32 CEST] <klaxa> if you could explain what you want to do "with audio" we could probably be more helpful [19:35:54 CEST] <Filystyn> anything I just don't get the purpsoe what may or may not i change here [19:36:05 CEST] <klaxa> also yes, AVDictionary is using strings [19:36:18 CEST] <Filystyn> is tehre some list of those strings [19:36:30 CEST] <Filystyn> so i know what i cna use at least? [19:36:32 CEST] <ubone> hello http://ix.io/wKA how to divide $fps output to match 59.xy format? [19:37:16 CEST] <klaxa> depends on whatever you are using the dictionary on, it is (mostly?) used to pass parameters that are not present in the generic structs [19:37:28 CEST] <klaxa> also, i still don't understand what you want to do [19:37:29 CEST] <klaxa> like [19:37:48 CEST] <Filystyn> I am trying to understand the purpose of this function [19:37:50 CEST] <klaxa> do you want to just demux a file? do you want to decode a file? mux, encode, create...? [19:37:57 CEST] <klaxa> we explained it a few times [19:38:02 CEST] <Filystyn> My point is to decode audio file only [19:38:10 CEST] <Filystyn> and stream it out to hear music [19:38:16 CEST] <klaxa> then you probably don't even need it [19:38:25 CEST] <Filystyn> ok thank you ;-) [19:38:34 CEST] <klaxa> if you want to stream it, you probably don't want to stream it raw [19:38:48 CEST] <klaxa> so maybe you want remuxing and not decoding? [19:39:53 CEST] <Filystyn> i will stream raw to sdl [19:39:58 CEST] <Filystyn> that should work [19:40:02 CEST] <klaxa> ah [19:40:43 CEST] <klaxa> for decoding maybe have a look at: https://ffmpeg.org/doxygen/trunk/decode_audio_8c-example.html [19:42:32 CEST] <Filystyn> It is fine but I decided to find out what type i have so I was refering to https://ffmpeg.org/doxygen/trunk/demuxing_decoding_8c-example.html#a52 [19:42:39 CEST] <Filystyn> which was recommanded me on this channel [20:36:54 CEST] <Filystyn> how to extract metadata [20:37:01 CEST] <Filystyn> of file ? [20:37:17 CEST] <ChocolateArmpits> What metadata exactly? [20:37:30 CEST] <Filystyn> sorry from avformatcontext [20:37:45 CEST] <Filystyn> I see the structure but can't find way to read it like char strings;> [20:37:58 CEST] <Filystyn> AVDictionary * metadata [20:38:05 CEST] <Filystyn> title of album etc [20:39:27 CEST] <Filystyn> ok i found it [20:39:30 CEST] <Filystyn> while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX))) [20:39:32 CEST] <Filystyn> printf("%s=%s\n", tag->key, tag->value); [20:39:36 CEST] <Filystyn> this is the way probably [20:46:08 CEST] <Filystyn> and it refuses to work [20:46:20 CEST] <Filystyn> i get the metadata other way but this function behaves like nothign is in there [20:56:48 CEST] <zubzub> Is it possible to have ffmpeg output non udp wrapped rtp data? [20:57:11 CEST] <zubzub> the docs seems to say you need to set an ip & port when using rtp [20:57:29 CEST] <zubzub> but I need to get ot the rtp packet itself without the udp encapsulation [21:00:47 CEST] <Filystyn> the read metadata does not work [21:28:28 CEST] <Filystyn> dump format works [21:28:36 CEST] <Filystyn> but can't extract format by my self [21:28:43 CEST] <Filystyn> wth [21:30:18 CEST] <Filystyn> i get printf( "%d", av_dict_count( av_formcont->metadata ) ); 0 from this when [21:30:44 CEST] <Filystyn> av_dump_format( av_formcont, 0, PLAYFILE, 0); returns data [21:30:47 CEST] <Filystyn> how ? [21:30:51 CEST] <Filystyn> where is the mistake ?! [21:51:53 CEST] <Filystyn> well? [22:06:10 CEST] <Filystyn> wake up guys [22:07:18 CEST] <johnjay> well what [22:07:27 CEST] <Filystyn> hey [22:07:53 CEST] <Filystyn> when i use av_dump_format( av_formcont, 0, PLAYFILE, 0); i get on stderr output of metadata cool [22:08:11 CEST] <Filystyn> but when i use printf( "%d", av_dict_count( av_formcont->metadata ) ); 0 i get zero metadata [22:08:13 CEST] <Filystyn> wth [22:20:36 CEST] <Filystyn> well where is the support [22:20:38 CEST] <Filystyn> ;-) [22:20:46 CEST] <Filystyn> this seem to not work [22:20:50 CEST] <Filystyn> prablem [22:20:52 CEST] <Filystyn> ;-) [22:33:15 CEST] <Filystyn> i got the bug [22:33:21 CEST] <tmatth> is there a quick and dirty way to validate a -filter_complex file without actually doing a real transcode? [22:33:23 CEST] <Filystyn> on mp3 i catch meta data [22:33:39 CEST] <Filystyn> but on ogg i don't with one function but i get it on the dump [22:33:46 CEST] <Filystyn> if this is not bug than wtf is it [22:33:49 CEST] <Filystyn> ? [22:36:04 CEST] <Filystyn> well where is the support it's a bug or what? [22:36:14 CEST] <Filystyn> the api si not transparent [22:40:54 CEST] <TMM> hi all [22:58:53 CEST] <kepstin> tmatth: what, just for syntax? [22:59:22 CEST] <kepstin> tmatth: can't really do more than basic syntax without at the very least probing all the input media to try to set up all the filter links :/ [22:59:55 CEST] <Filystyn> what about supporting me [22:59:58 CEST] <Filystyn> where are devs [23:00:05 CEST] <Filystyn> Jezus this is so fucking no transparent [23:00:12 CEST] <Filystyn> It pisses the shit of mwe [23:09:04 CEST] <Soni> autorotate doesn't work with JPEG [23:19:00 CEST] <Maverick|MSG> is there a method to automatically stretch/shrink an audio file so that is always matches the output length of a video when they are combined together? [23:19:45 CEST] <Maverick|MSG> minus manually specifying the atempo value [23:35:06 CEST] <tmatth> kepstin: yeah just for syntax [00:00:00 CEST] --- Fri Jun 9 2017

1 0

ffmpeg-devel.log.20170607
by burek 08 Jun '17

08 Jun '17

[00:06:19 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:eac6114e0187: avcodec/cavsdec: Fix runtime error: signed integer overflow: 59 + 2147483600 cannot be represented in type 'int' [00:06:19 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:fc74ac463c08: avcodec/pnm: Use ff_set_dimensions() [00:06:19 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:e3a1d133f71d: avcodec/ra144: Fixes runtime error: signed integer overflow: 7160 * 327138 cannot be represented in type 'int' [00:06:19 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:797621afab82: avcodec/hevc_ps: Fix runtime error: signed integer overflow: 2147483628 + 256 cannot be represented in type 'int' [00:06:19 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:706b427ff5de: avcodec/cinepak: Check input packet size before frame reallocation [00:06:19 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:30abd8e6f982: avcodec/wavpack: Fix runtime error: signed integer overflow: 2013265955 - -134217694 cannot be represented in type 'int' [00:06:19 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:4f02447d4568: avcodec/cfhd: Fix runtime error: signed integer overflow: 65280 * 65288 cannot be represented in type 'int' [00:06:20 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:3dc62e679ab2: avcodec/wavpack: Fix runtime error: shift exponent 32 is too large for 32-bit type 'int' [00:06:21 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:52a7ae844b05: avcodec/aacps: Fix runtime error: left shift of 1073741824 by 1 places cannot be represented in type 'INTFLOAT' (aka 'int') [00:06:22 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:4e6de49a5a4a: avformat/options: log filename on open [00:06:23 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:aae731b9d39f: avcodec/ac3dec_fixed: Fix runtime error: left shift of 419 by 23 places cannot be represented in type 'int' [00:06:24 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:b7904b58afdd: avcodec/pafvideo: Check packet size and frame code before ff_reget_buffer() [00:06:25 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:00312b5ea4a2: avcodec/dxv: Check remaining bytes in dxv_decompress_raw() [00:06:26 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:0d19167a657c: avcodec/hevc_ps: Fix runtime error: index 32 out of bounds for type 'uint8_t [32]' [00:06:27 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:cefbc513ea1c: avutil/softfloat: Fix sign error in and improve documentation of av_int2sf() [00:06:28 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:d34d06d1e233: avcodec/qdrw: Fix null pointer dereference [00:06:29 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:3dd1f38329e7: avformat/hls: Check local file extensions [00:06:30 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:0fb432a23b0a: avcodec/cavs: Fix runtime error: signed integer overflow: -12648062 * 256 cannot be represented in type 'int' [00:06:31 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:ec5e262e1d72: avcodec/tiff: Avoid loosing allocated geotag values [00:06:32 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:34887d091d53: avcodec/mjpegdec: Check that reference frame matches the current frame [00:06:33 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:07944df9a7d7: avcodec/takdec: Fix multiple runtime error: signed integer overflow: 637072 * 4096 cannot be represented in type 'int' [00:06:34 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:90b6425b1285: avcodec/pafvideo: Fix assertion failure [00:06:35 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:4c7477f132ea: avcodec/mpeg4videodec: Fix runtime error: signed integer overflow: 53098 * 40448 cannot be represented in type 'int' [00:06:36 CEST] <cone-114> ffmpeg 03Michael Niedermayer 07release/3.3:6d7192bcb7bb: Update for 3.3.2 [00:44:09 CEST] <atomnuker> TD-Linux: what does MIPS stand for, you think? [00:57:25 CEST] <TD-Linux> atomnuker, I can't think of any savory guesses [00:58:59 CEST] <iive> million instructions per second? [00:59:58 CEST] <atomnuker> TD-Linux: Much Insufficient Power Spare [01:00:21 CEST] <TD-Linux> I like it [04:12:05 CEST] <cone-760> ffmpeg 03Tyler Jones 07n3.3.2:HEAD: vorbisenc: Fix memory leak on errors [06:02:41 CEST] <KGB> [13FFV1] 15dericed opened pull request #70: expanding on median predictor section (06master...06median-predictor) 02https://git.io/vHKtx [08:21:47 CEST] <mateo`> looks like ios 11 defines a new type, named ... AVMediaType :D [08:26:53 CEST] <kevmark> mateo`, FFmpeg baked into iOS? [08:27:30 CEST] <mateo`> probably or apple wanting us to choose between ios and ffmpeg [08:27:40 CEST] <mateo`> it's in the AVFoundation framework [08:31:59 CEST] <kevmark> https://developer.apple.com/documentation/avfoundation/avmediatype [08:32:15 CEST] <kevmark> I'm guessing it's a coincidence if only because the AV prefix is in AVFoundation [08:35:16 CEST] <kevmark> And the AVMediaType struct from Apple is markedly different than the enum in FFmpeg [08:35:35 CEST] <kevmark> But maybe I'm interpreting what you're saying incorrectly [08:39:49 CEST] <JEEB> well it's a thing that has the same name under the same namespace ;) [08:50:49 CEST] <mateo`> kevmark: what i meant (beside the joke), is what JEEB said, it will conflict when you'll include both avformat and avfoundation in the same place [08:51:27 CEST] <kevmark> Ahhh hah [08:51:36 CEST] <kevmark> gg apple [09:42:03 CEST] <rcombs> merh, their is an NSString* and ours is an enum [09:42:50 CEST] <rcombs> that'll actually cause compile errors [09:42:58 CEST] <ubitux> we shouldn't need AVFoundation and FFmpeg at the same time [09:43:13 CEST] <ubitux> AVFoundation is way too high level [09:43:26 CEST] <ubitux> ah, unless we use it as an indev hhehe [09:44:05 CEST] <rcombs> or, wait [09:44:16 CEST] <rcombs> ours isn't typedeffed [09:44:34 CEST] <rcombs> so& it might just be fine [12:21:33 CEST] <J_Darnley_irssi> Is anyone else using nasm 2.13 yet? Do you have any problems running configure? [12:23:07 CEST] <atomnuker> I am, haven't had any issues [12:24:18 CEST] <RiCON> been using it since x264 required it, no problems so far [12:24:21 CEST] <nevcairiel> i installed nasm 2.13.1, but i still have yasm installed so configure uses that [12:24:41 CEST] <nevcairiel> 2.13.01* [12:33:06 CEST] <cone-196> ffmpeg 03wm4 07master:66cf78e93295: lavf: consider codec framerate for framerate detection [12:33:16 CEST] <J_Darnley_irssi> Hm... thanks [12:35:41 CEST] <BtbN> wm4, could that also be used for the cuvid frame doubling deinterlacing, or is that something entirely diffrent? [12:36:08 CEST] <wm4> BtbN: what I just pushed? it only fixes framerate detection for broadcast .ts files [12:39:22 CEST] <BtbN> hm, it only checks the probes framerate anyway [12:41:52 CEST] <wm4> what's the problem with qsv? can't you just require that the caller sets a doubled timebase [12:43:58 CEST] <BtbN> qsv? [12:44:08 CEST] <BtbN> Well, in case of cuvid, you have to do exactly that. [12:44:14 CEST] <wm4> wrong vendor... I meant cuvid [12:44:27 CEST] <BtbN> Which is kinda annoying if you have a bunch of files with diffrent framerates and you want to deinterlace them [12:45:07 CEST] <BtbN> I'd still like to turn the cuvid scaler/deinterlace into a filter, but I can't find any documentation on how to do that [12:45:19 CEST] <BtbN> The API is there, but no obvious way of using it works [12:45:22 CEST] <nevcairiel> just try all the things [12:45:43 CEST] <BtbN> Some nvidia guy claimed it to be possible [12:45:56 CEST] <BtbN> but never replied to me asking how [12:46:00 CEST] <wm4> lol [12:47:04 CEST] <BtbN> I have the generic filter done, but it just doesn't work [12:49:35 CEST] <wm4> so almost there? [12:51:35 CEST] <BtbN> https://github.com/BtbN/FFmpeg/commits/nvpp [12:51:36 CEST] <BtbN> well [12:51:40 CEST] <BtbN> I'm out of ideas [12:51:44 CEST] <BtbN> On how to use it [12:51:53 CEST] <BtbN> For starters, I just wanted to make it scale [12:52:05 CEST] <BtbN> but I have absolutely no idea how to make it process and output raw YUV [12:54:14 CEST] <wm4> so what about it doesn't work? [12:55:55 CEST] <BtbN> input and/or output of raw yuv [12:56:16 CEST] <BtbN> The API is not exactly clear on how to do that, and it's not in any written documentation [13:02:19 CEST] <wm4> aren't opaque surfaces more interesting? [13:12:16 CEST] <BtbN> hm? [13:12:36 CEST] <BtbN> CUDA frames are raw yuv data, so that's what needs to be processed, and it should support that, somehow [13:16:00 CEST] <wm4> ah, I understood "raw yuv" as data in CPU memory [13:16:21 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vHKy7 [13:16:21 CEST] <KGB> 13FFV1/06master 142113ed6 15Dave Rice: update readme to link ietf versions of ffv1... [13:54:40 CEST] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vHKHN [13:54:40 CEST] <KGB> 13FFV1/06master 14f566e93 15Dave Rice: equation typo... [14:18:09 CEST] <J_Darnley_irssi> weird. nasm is complaining that it doesn't know the -m option [14:18:40 CEST] <J_Darnley_irssi> I wonder what patches did Arch apply to it [14:20:06 CEST] <kierank> J_Darnley_irssi: huh [14:20:11 CEST] <kierank> you have left the other chnanel [14:20:57 CEST] <Gramner> there is no -m option in nasm so that's not really weird. [14:22:19 CEST] <J_Darnley_irssi> ah [14:22:51 CEST] <Gramner> there's also no point in using -m with yasm either. if you use the correct -f flag it will work with both [14:23:56 CEST] <Gramner> e.g. "-f win64" instead of "-f win32 -m amd64" etc. [14:29:38 CEST] <RiCON> J_Darnley_irssi: probably http://git.videolan.org/?p=ffmpeg.git;a=blob;f=configure;hb=HEAD#l5477 [14:30:55 CEST] <RiCON> 'nasm --version' works so configure adds "-m amd64" [14:32:42 CEST] <RiCON> libav fixed this by checking for '-v' for nasm first [14:34:02 CEST] <nevcairiel> those fixes are scheduled for merging e ither way [15:23:52 CEST] <J_Darnley_irssi> RiCON: oh yes [15:35:32 CEST] <ubitux> <not supported> cycles [15:35:33 CEST] <ubitux> <not supported> instructions [15:35:35 CEST] <ubitux> fuck this shit :( [15:36:51 CEST] <jamrial> what is that? [15:36:52 CEST] <iive> is that some performance monitor? [15:36:56 CEST] <DHE> it's perf [15:36:59 CEST] <DHE> what are you running it on? [15:37:13 CEST] <ubitux> it's perf, with a aarch64 mainline kernel on rpi3 [15:37:34 CEST] <ubitux> all the perf options seems on in the config [15:37:58 CEST] <nevcairiel> doing that on arm was really troublesome, you should ask wbs on how to get it working [15:37:58 CEST] <ubitux> (kernel 4.11.3) [15:39:17 CEST] <wbs> normally perf should work out of the box, since it uses the cycle counter instruction from kernel mode (which is ok); it's if using it from userspace (like for checkasm --bench) you need to manually enable it via executing a couple instructions in kernel mode [15:39:28 CEST] <wbs> (tl;dr can't help with that one) [15:40:44 CEST] <ubitux> yeah, it's actually supported out of the box on the odroid [15:41:14 CEST] <ubitux> it's just the rpi3 that's not supported [15:41:21 CEST] <ubitux> fucking garbage [15:41:50 CEST] <DHE> hardware, or is the kernel feature missing? [15:42:21 CEST] <wbs> probably something along the lines of hardware/kernel combo [15:43:26 CEST] <ubitux> yeah, so anyway, basically there is still no decent aarch64 dev board available [15:59:21 CEST] <jamrial> what's wrong with the odroid? [16:00:00 CEST] <wm4> bad video decoding/display drivers [16:00:00 CEST] <mateo`> it's stuck with a 3.14 kernel [16:46:39 CEST] <ubitux> jamrial: i updated my aacpsdsp branch to actually test the results from the functions in checkasm [16:46:51 CEST] <ubitux> and it appears to fail on x86 (but not aarch64) [16:47:04 CEST] <ubitux> but maybe i'm doing sth wrong [16:47:18 CEST] <ubitux> stereo_interpolate_ipdopd_sse3 is failing [16:47:36 CEST] <ubitux> like, it looks like the first entry is wrong (but not the others) [16:48:22 CEST] <ubitux> also, the test don't seem to like with a stride of 1 but i may have broken something [17:40:09 CEST] <jamrial> ubitux: huh, weird seeing that the standard fate test for ipdopd passes [17:48:57 CEST] <jamrial> also, remove all that INTFLOAT stuff, it's not needed here [18:18:46 CEST] <Compn> michaelni : i'm going to be away for a week. june 8-15. just fyi about mailing list moderations [18:29:12 CEST] <durandal_1707> Compn: give your admin flag to me [18:30:38 CEST] <jamrial> man, fuck you fate-aac-al_sbr_ps_04_ur [18:31:19 CEST] <jamrial> ubitux: found what was wrong. i'm surprised the above test didn't complain about it [18:54:49 CEST] <cone-148> ffmpeg 03James Almer 07master:497a4b554c8d: x86/aacpsdsp: fix output of ff_ps_stereo_interpolate_ipdopd_sse3 [18:55:21 CEST] <jamrial> ubitux: ^ thanks for pointing it out [19:25:09 CEST] <J_Darnley> I'll probably send an email on this topic too... [19:25:55 CEST] <J_Darnley> To support nasm 2.13 I wonder whether I should steal Libav's configure code for probing, [19:25:58 CEST] <J_Darnley> merge it, [19:26:30 CEST] <J_Darnley> put a band-aid on ours by swapping the checks. [19:26:58 CEST] <atomnuker> can't we just pick nasm by default [19:27:46 CEST] <J_Darnley> Sure. I would be willing to follow x264's lead and require 2.13 but I bet other are not. [19:29:23 CEST] <iive> my distro hasn't shipped 2.13 yet. [19:29:50 CEST] <J_Darnley> Get with the times! It is $CURRENT_DAY :) [19:29:52 CEST] <JEEB> debian put it into experimental so I could build packages :3 [19:30:18 CEST] <Gramner> debian stable might support 2.13 in 3 years or so [19:30:47 CEST] <J_Darnley> Perhaps a compromise of the "new" development of nasm would be acceptable. [19:30:55 CEST] <J_Darnley> 2.10-something I think [19:32:13 CEST] <iive> btw, does configure have an option that would make it pick nasm over yasm (or vice versa) if both are installed? [19:32:47 CEST] <J_Darnley> Yes: --yasmexe=anything_other_than_yasm [19:33:09 CEST] <iive> that's somehow hackish [19:33:17 CEST] <Gramner> 2.13 is required for avx-512 (because 2.12 was a minefield of evex bugs until I stumbled across them and fixed them myself or made someone else fix them) and self-relative addressing. ffmpeg uses neither at the moment so there's not really much point in requiring 2.13 [19:33:23 CEST] <J_Darnley> if the check for $yasmexe --version fails it will force yasmexe=nasm [19:34:31 CEST] <iive> Gramner: i guess yasm doesn't support avx-512 at all? [19:34:47 CEST] <Gramner> i should probably port the x86inc avx-512 changes to ffmpeg though if someone else wants to start cracking away on it [19:34:51 CEST] <J_Darnley> I think it doesn't even support all of AVX2 [19:34:53 CEST] <Gramner> correct. no avx-512 in yasm [19:35:25 CEST] <Gramner> avx2 is fully supported in yasm, at least I haven't noticed anything that isn't [19:35:31 CEST] <J_Darnley> I think there was some fused multiply-add or XOP instruction I wanted to use in flac but the assembler didn't recognise it. [19:35:45 CEST] <J_Darnley> (I guess XOP isn't AVX2) [19:35:55 CEST] <Gramner> xop is dead though, I wouldn't spend any time on writing such code [19:36:09 CEST] <J_Darnley> That was 2+ years ago. [19:36:10 CEST] <iive> is it fma3/4 like ? [19:36:19 CEST] <J_Darnley> Yes, but for integers [19:36:20 CEST] <ubitux> jamrial: nice! :) [19:36:39 CEST] <J_Darnley> Gramner: I will tackle importing the x264asm changes. [19:36:40 CEST] <ubitux> jamrial: about INTFLOAT, it makes supporting fixed later simpler [19:36:42 CEST] <Gramner> xop was nice for horizontal stuff. a shame intel doesn't seem to care about that [19:36:56 CEST] <Gramner> and that onle gets worse as vector widths increases [19:37:12 CEST] <Gramner> many use cases require summing all values in a register [19:37:50 CEST] <iive> and accessing higher parts of registers is uglier and uglier [19:37:57 CEST] <Gramner> which currently results in a ton of shuffle -> add -> shuffle -> add -> repeat [19:38:11 CEST] <Gramner> and two of those are cross-lane shuiffles when summing a zmm reg [19:38:23 CEST] <Gramner> so it takes forever [19:39:09 CEST] <iive> imho, they should have made the avx1/2 use joined registers [19:39:43 CEST] <Gramner> no, the lane stuff makes a lot of sense to be honest. I hated it at first but I've gotten used to it [19:39:50 CEST] <iive> aka, make xmm32-63 and let the new operations use the pair xmm0/xmm32 [19:39:54 CEST] <Gramner> and avx-512 helps a lot with the new shuffles [19:40:26 CEST] <iive> i'd love line stuff.. [19:41:39 CEST] <iive> my point is ... the new ymm operations does act like they are 2 joined xmm operations, the high and low halves of registers are separate. [19:43:28 CEST] <iive> and the extension to this idea is to have not just 2 but n-th joined registers, where n is written in the instruction prefix. [19:46:29 CEST] <Gramner> not sure if that'd be very useful compared to the significant complexity increase [19:46:48 CEST] <iive> there is no really a complexit increase [19:46:53 CEST] <iive> quite the contrary [19:47:34 CEST] <iive> e.g. with register size of 512, you do 4x128 operations. if you have 4 op units [19:48:05 CEST] <iive> you can pick if you want to do 4 different ops on 4 different xmm registers or 1x512 [19:49:37 CEST] <Gramner> that's basically how many cpus work internally already. amd does avx2 like that one every cpu, but's it's an implementation detail and you don't want to expose every implementation detail or you'll be forced to support that specific way of doing things forever [19:49:54 CEST] <iive> instead, we have a power-save feature that turns off the high register part if it is not used, and it takes 60000 clocks to power it up at first use. [19:50:49 CEST] <iive> Gramner: that's the whole point. that's how cpu work already. it doesn't need anything new or complicated [19:50:53 CEST] <Gramner> the power gating and thermal throttling of SIMD registers sucks balls, but that feels more like a poor architectural design than and instruction set issue [19:51:34 CEST] <iive> Gramner: it just means that the high part is not used at all. just wasted die. [19:51:49 CEST] <iive> also, who would not like 4 or 8 time more registers? [19:52:53 CEST] <J_Darnley> Reminds me of my old Athlon64. I'm not sure I ever ran 64-bit code on it. [19:54:35 CEST] <Gramner> I don't really see the need for going from 32 simd registers to 128 though. not sure what the use case for that is [19:55:51 CEST] <jamrial> 64 regs would be ideal for avx512, as BBB once mentioned. just like avx2 should have bumped it to 32 [19:56:03 CEST] <jamrial> perfect for transpose [19:56:43 CEST] <Gramner> also twice as much stuff to backup and restore on context switches [19:57:06 CEST] <iive> but you see, that the whole point [19:57:16 CEST] <iive> it is the same number of bits [19:57:25 CEST] <iive> aka 4 xmm regs or 1 ymm [19:57:44 CEST] <iive> you just break 1ymm on 4 addressable xmm registers [19:58:00 CEST] <iive> you can do permutations on them with mov. [19:58:23 CEST] <iive> instead of extracts that take 5 cycles each [19:58:59 CEST] <jamrial> iive: ymm is twice the size of xmm. you're thinking zmm [19:59:07 CEST] <iive> yep, sorry [20:00:53 CEST] <Gramner> why even have wider registers then? if you want 256-bit ops to run at half the speed of 128-bit (and 512-bit at 1/4 the speed) just stick with a 128-bit instruction set and skip the complexity [20:01:17 CEST] <iive> to have shorter opcodes [20:01:27 CEST] <iive> the same reason we have zmm ops now. [20:01:42 CEST] <J_Darnley> Bye again [20:01:45 CEST] <iive> actually i just had the idea, how about using the existing prefix "rep" [20:09:45 CEST] <iive> Gramner: 512bit ops does 4 x128 in parallel. they do it now, the 128 parts are separate. [20:10:05 CEST] <iive> So I do not see it as 512 been slower, but as 128 been 4 times faster. [21:40:51 CEST] <blue_misfit> hey guys! Should ffmpeg libraries theoretically support QuickTime MOV files with mixed types of video in a given track? e.g. prores + dnxhd [21:41:04 CEST] <blue_misfit> apparently this is possible to make in Avid, and native QuickTime apps like Premiere handle it just fine, crazily enough [21:46:15 CEST] <jamrial_> ubitux: https://github.com/jamrial/FFmpeg/commits/aarch64-aac couple changes you can squash [21:53:12 CEST] <durandal_170> blue_misfit: i doubt, why not try anyway, have a sample? [21:54:32 CEST] <blue_misfit> I do have a sample but I can't share (high security) but I'll try to make one that's low security. The file I looked at started with dnxhd which transcoded fine but when it switches to prores the output has black frames. When it switches back to dnxhd it recovers :) [22:58:34 CEST] <rcombs> blue_misfit: 0.o who does this [22:58:56 CEST] <rcombs> though actually, I need this for MPEGTS (broadcast TV) [22:59:13 CEST] <nevcairiel> in general we dont support this, no. [23:00:10 CEST] <nevcairiel> and it would probably be hard to do that as well, since there are many structures that depend on the codec and driving a re-init through everything would be rather complex [23:06:30 CEST] <wm4> obviously this should happen on a higher level [23:06:36 CEST] <wm4> mpv supports it [23:06:54 CEST] <wm4> and *shock* that didn't need lavc API changes [23:08:21 CEST] <rcombs> yeah, I'd imagine it'd be in the API consumer, perhaps with some assistance from lavf [23:42:47 CEST] <fiberbaby> https://www.ssllabs.com/ssltest/analyze.html?d=ffmpeg.org [23:42:54 CEST] <fiberbaby> https://blog.qualys.com/ssllabs/2017/04/05/ssl-labs-distrusts-wosign-and-st… [23:43:04 CEST] <fiberbaby> how about using Let's Encrypt [00:00:00 CEST] --- Thu Jun 8 2017

1 0

ffmpeg.log.20170607
by burek 08 Jun '17

08 Jun '17

[08:11:59 CEST] <ac_slater> sigh. Can -map let me dump to raw binary? [08:12:24 CEST] <ac_slater> I have an mpegts container with a data stream that ffmpeg identifies [08:12:29 CEST] <ac_slater> but there is no output type for it [08:15:27 CEST] <ac_slater> ahhh [08:16:00 CEST] <furq> -copy_unknown [08:17:45 CEST] <ac_slater> yu [08:17:46 CEST] <ac_slater> yup * [08:17:48 CEST] <ac_slater> thanks furq [08:18:02 CEST] <ac_slater> also, -f data worked [08:18:17 CEST] <ac_slater> just `ffmpeg -i input -map 0:1 -f data raw.bin` [13:31:18 CEST] <ArsenArsen> Anyone even tried to do MP4 generation from within C/C++ code without making a subprocess [13:33:09 CEST] <JEEB> yes, just use lavf API for muxing? [13:33:14 CEST] <JEEB> (and lavc if you need encoding) [13:36:50 CEST] <ArsenArsen> Aha. I see. Thanks :) [13:37:01 CEST] <JEEB> there are examples under docs [13:37:03 CEST] <JEEB> for API usage [13:37:14 CEST] <ArsenArsen> Found that one, reading through it now. [17:29:20 CEST] <Filystyn> how to probe with C api ?:> [17:29:27 CEST] <Filystyn> anyone did it before? [17:31:15 CEST] <DHE> clarify please [17:40:48 CEST] <c_14> Filystyn: avformat_find_stream_info [17:42:00 CEST] <c_14> check what https://ffmpeg.org/doxygen/trunk/demuxing_decoding_8c-example.html uses [17:49:12 CEST] <Filystyn> thank you [18:14:42 CEST] <Filystyn> how do i check AVERROR [18:14:43 CEST] <Filystyn> ? [20:32:56 CEST] <CaptainAmerica> HI everyone, I am completely stuck. I have 1 WAV + 1 GIF and I want to output 1 MP4 from this, so the video runs for the length of the audio with the image showing. That's all. But I keep getting loop errors. Does anyone know how to do this? [20:33:22 CEST] <CaptainAmerica> It is so simple but it driving me crazy trying to get it to work. Running a PC windows 10 [20:33:40 CEST] <CaptainAmerica> Thank you! [20:33:59 CEST] <furq> ffmpeg -ignore_loop 0 -i foo.gif -i bar.wav -shortest out.mp4 [20:36:11 CEST] <CaptainAmerica> Thank you @furq I will give it a go! [20:38:40 CEST] <CaptainAmerica> Nope @furq that doesn't work... It doesn't run for either the whole track OR show the image [20:38:42 CEST] <CaptainAmerica> Damn [20:41:58 CEST] <Filystyn> where do i find the audio stream.. ? [20:42:16 CEST] <Filystyn> I think i got it i guess this is raw audio stream but not sure where is it placed [20:43:05 CEST] <CaptainAmerica> Frustrating that ffmpeg is uber powerful and I can get it to do lots but the simplest thing I can't figure out. Doh. Any other ideas? [20:43:55 CEST] <ChocolateArmpits> CaptainAmerica, I think furq's command is missing -map 0:v -map 1:a [20:44:22 CEST] <CaptainAmerica> Thank you @chocolatearmpits (best handle ever!) I'll try that [20:45:47 CEST] <CaptainAmerica> I'm getting no such directory with the -map command which is a bit odd?! [20:46:20 CEST] <ChocolateArmpits> Can you paste your command line ? [20:46:22 CEST] <CaptainAmerica> Did I put it in the wrong place? >> ffmpeg -ignore_loop 0 -i -map 0:v -map 1:a foo.gif -i bar.wav -shortest out.mp4 [20:46:37 CEST] <ChocolateArmpits> oh yeah those two are output options [20:46:44 CEST] <ChocolateArmpits> you should place them after -shortest [20:46:51 CEST] <CaptainAmerica> Thank you I'll try that [20:48:33 CEST] <CaptainAmerica> nope same issue. Damn. [20:48:51 CEST] <CaptainAmerica> Maybe I do this is two steps. [20:49:36 CEST] <CaptainAmerica> WAV > MP4, then append image to display on the video on every frame. Any idea if that would work and how I'd do that? Thank you. [20:51:05 CEST] <CaptainAmerica> (or MP3 to MP4) [20:53:12 CEST] <ChocolateArmpits> maybe it would be smarter to create a video out of the gif first [20:53:41 CEST] <CaptainAmerica> yes sure... Just investigating how to do that now [20:55:57 CEST] <ChocolateArmpits> CaptainAmerica, does the image itself loop in image viewers or does it stop ? [20:56:23 CEST] <CaptainAmerica> it is just one title slide I want to join to an audio bed [20:56:33 CEST] <CaptainAmerica> I haven't got it to render as a video anywhere yet [20:56:52 CEST] <CaptainAmerica> ffmpeg foo.gif -t 00:55:00 bar.mp4 -- this gives me "output file does not contain any stream" error [20:57:12 CEST] <ChocolateArmpits> that's because you don't use -i input parameter [20:57:30 CEST] <ChocolateArmpits> that should go before foo.gif [21:02:07 CEST] <CaptainAmerica> Thank you ... Ok so it now appears in the video but the time/duration setting is completely ignored... Ie. It is just one frame [21:02:15 CEST] <CaptainAmerica> progress though :) [21:02:50 CEST] <ChocolateArmpits> try setting video encoding settings [21:02:56 CEST] <ChocolateArmpits> -vcodec libx264 for starters [21:03:17 CEST] <ChocolateArmpits> ffmpeg may be embedding the image as simple album art if no encoding information is given [21:07:32 CEST] <Filystyn> how do i error check on ffmpeg [21:07:43 CEST] <Filystyn> this manual is really confusing,... [21:09:28 CEST] <ChocolateArmpits> Filystyn, what error are you looking for ? [21:09:57 CEST] <ChocolateArmpits> You can get more verbose output with -loglevel trace [21:11:08 CEST] <Filystyn> i need the c api erorr [21:11:28 CEST] <Filystyn> having hard time to get the errno replacement for ffmpeg [21:11:29 CEST] <JEEB> the APIs generally define what is an error value [21:11:45 CEST] <Filystyn> what is the global errno [21:11:49 CEST] <JEEB> what? [21:11:50 CEST] <Filystyn> just the name of it [21:11:59 CEST] <JEEB> there is no global errno in the C APIs? [21:12:03 CEST] <JEEB> the APIs return values [21:12:15 CEST] <Filystyn> so it's different status ? [21:12:26 CEST] <JEEB> just look at the examples under docs? [21:12:30 CEST] <JEEB> like, there are examples in there [21:12:37 CEST] <JEEB> not the best ones, but there are [21:12:43 CEST] <JEEB> (29 [21:12:53 CEST] <Filystyn> i saw example sman but i tried to relay more on manual [21:13:11 CEST] <JEEB> the "manual" is for command line usage [21:13:14 CEST] <JEEB> not for the APIs [21:13:21 CEST] <JEEB> for API you look at the headers/doxygen [21:13:40 CEST] <Filystyn> yes im using thos https://www.ffmpeg.org/doxygen/trunk/error_8h.html [21:13:58 CEST] <Filystyn> anyway the way function are checked suggest that there was a global error code [21:14:06 CEST] <JEEB> uhh [21:14:07 CEST] <Filystyn> checke din the examples [21:14:16 CEST] <JEEB> I'm pretty sure there is no global error variable [21:14:21 CEST] <JEEB> it's almost always the return value [21:14:26 CEST] <Filystyn> well probably there is no since i could not find it [21:14:36 CEST] <Filystyn> i guess it's like with threads [21:14:39 CEST] <Filystyn> get the status [21:15:09 CEST] <JEEB> the error values are indeed defined and then can be gotten as string or something like that (human readable) [21:15:23 CEST] <Filystyn> ok the rest i can handle thank you for help ;') [21:18:47 CEST] <CaptainAmerica> Good news @chocolatearmpits... this is working re duration > ffmpeg -loop 1 -i foo.jpg -c:v libx264 -t 15 -pix_fmt yuv420p -vf scale=320:240 bar.mp4 [21:19:01 CEST] <ChocolateArmpits> cool [21:19:13 CEST] <CaptainAmerica> now I need to get duration exact and add audio on top for the next trick ;) [21:22:39 CEST] <kepstin> CaptainAmerica: easy way to do this, simply add another '-i input.mp3' or whatever for the audio file, add a "-c:a copy" (if the audio codec works in mp4) or some other codec (aac, libmp3lame), then replace the "-t 15" with "-shortest" and it'll automatically cut the video when the audio ends. [21:23:19 CEST] <CaptainAmerica> Thank you you @kepstin ... Will give that a go too! Thank you too [21:26:18 CEST] <CaptainAmerica> we're starting to win here guys... Thank you @kesptin @chocolatearmpits [21:27:19 CEST] <CaptainAmerica> last big challenge... I want to concat two videos (this one we just made and another) -- is it possible to cross fade / fade one into other? [ I know I am really pushing it now :) ] [21:29:39 CEST] <CaptainAmerica> maybe that is a bridge too far for now and I will just combine them :) [21:32:46 CEST] <Filystyn> where is open_codec_context defined what header [21:32:52 CEST] <Filystyn> and how to link this [21:32:56 CEST] <Filystyn> -lffmpeg doe snot work [21:33:36 CEST] <c_14> -lavformat probably [21:33:39 CEST] <c_14> or -lavcodec [21:33:51 CEST] <c_14> have you tried the search in the doxygen? [21:34:11 CEST] <Filystyn> yes im doxygening man [21:34:17 CEST] <blue_misfit> hey guys! Ever seen a quicktime file with 1 video track contain multiple types of video in that track (e.g. prores + dnxhd)?? [21:34:26 CEST] <Filystyn> the demuxing header is not there [21:34:59 CEST] <c_14> Filystyn: https://ffmpeg.org/doxygen/trunk/demuxing__decoding_8c_source.html#l00149 it's a private function in the examples [21:35:45 CEST] <Filystyn> ooo [21:35:53 CEST] <Filystyn> facepalmed k man but it;'s in manual [21:35:57 CEST] <Filystyn> anyway linking worked [21:35:59 CEST] <Filystyn> thx man [21:36:05 CEST] <Filystyn> should have just look on headers [21:37:00 CEST] <Filystyn> gona write my own for now i think we got this ;-) [21:37:10 CEST] <argwings> hi, did vf scale change recently? -vf scale=800:-1 doesn't seem to work anymore? sorry probably dumb question [21:37:27 CEST] <Filystyn> c_14 really big thx man you did help me much today [21:38:25 CEST] <c_14> argwings: probably not, should work [21:38:45 CEST] <c_14> what's the error? [21:42:01 CEST] <kepstin> argwings: keep in mind that with some source video files in some pixel formats (e.g. yuv420 with subsampled chroma), scale=800:-1 won't work because the size has to be a multiple of two. Use 'scale=800:-2' instead to fix that. [21:52:57 CEST] <argwings> kepstin: ah thanks yes it gave that error about multiple of two, and -2 worked [21:53:33 CEST] <argwings> cool, keep up the good work 8) [21:54:27 CEST] <CaptainAmerica> Hi again.. I'm trying to join 2 videos but it isn't working, it just outputs the first in the output file using > ffmpeg -i "concat:foo.mp4|C:\bar.wmv" -c copy output.mp4 [21:54:40 CEST] <CaptainAmerica> Can anyone see what I've done wrong? [21:54:42 CEST] <CaptainAmerica> Thank you [21:55:10 CEST] <kerio> well [21:55:10 CEST] <kerio> for instance [21:55:13 CEST] <blue_misfit> you probably won't be able to do a concat with -c copy mixing that mp4 and wmv, assuming they're different codecs [21:55:17 CEST] <kerio> you're here asking about how to concatenate two videos [21:55:24 CEST] <kerio> instead of learning an instrument to pick up chicks [21:55:30 CEST] <blue_misfit> ROFL [21:55:38 CEST] <kerio> it's a common mistake [21:55:55 CEST] <blue_misfit> switch -c copy to -c:v libx264 and see if it works. That will transcode, but you're likely gonna have to do that no matter what [21:56:25 CEST] <kerio> can you even -c:v copy with the concat demuxer [21:56:28 CEST] <kerio> like in general [21:56:37 CEST] <blue_misfit> yes, afaik [21:57:02 CEST] <blue_misfit> hmm not totally sure actually but I'd hope so, provided all inputs are the same format [21:57:23 CEST] <CaptainAmerica> Thank you @blue_misfit ... that didn't have an impact unfortunately :( [21:57:32 CEST] <furq> kerio: of course you can [21:57:45 CEST] <furq> obviously not in this case though [21:57:52 CEST] <furq> or in most cases, as it turns out, but it does occasionally work [21:58:21 CEST] <blue_misfit> CaptainAmerica, post your full cmd and output in a pastebin [21:58:38 CEST] <furq> that's the concat protocol though [21:59:05 CEST] <kerio> furq: mux both videos into mpegts, concat the files, pray? [21:59:23 CEST] <furq> not if one of them is wmv [21:59:35 CEST] <kerio> if one of them is wmv you must commit honorable subaru [22:00:07 CEST] <furq> why not mitsubishi [22:00:19 CEST] <kerio> anyway apple is adding flac support to ios [22:00:27 CEST] <kerio> but only in the Files app, not the Music app lul [22:00:49 CEST] <kerio> they're also adding support for opus because of webrtc :3 [22:01:05 CEST] <CaptainAmerica> this is my code https://pastebin.com/kRckeueu - thanks [22:01:05 CEST] <furq> astonishing news: apple are dickheads [22:01:11 CEST] <furq> "imagine that" [22:01:25 CEST] <blue_misfit> and of course hevc, which is neat [22:01:27 CEST] <kerio> the imac pro is so fucking slick tho [22:01:31 CEST] <furq> CaptainAmerica: https://trac.ffmpeg.org/wiki/Concatenate [22:01:44 CEST] <blue_misfit> it looks nice, but mehhhhhhhhhh it's for vanity pros [22:01:46 CEST] <furq> the protocol won't work with those input formats, use the demuxer or filter [22:01:58 CEST] <furq> also the hevc stuff looks like an absolute nightmare [22:02:01 CEST] <blue_misfit> there you go CaptainAmerica ! ^^ [22:02:11 CEST] <furq> aren't they using some bullshit hevc i-frame format for still images now [22:02:15 CEST] <blue_misfit> concat demuxer ftw [22:02:16 CEST] <kerio> :o [22:02:19 CEST] <blue_misfit> yeah, wouldn't that be awesome? [22:02:20 CEST] <CaptainAmerica> thanks @furq ... That is exactly the code I lifted and isn't working for me [22:02:36 CEST] <furq> which means you get to ride aboard the hevc licensing rollercoaster if you want to write something that decodes it without using the system APIs [22:03:11 CEST] <blue_misfit> well, I'm sure they still support jpg and stuff [22:03:17 CEST] <blue_misfit> and why wouldn't you use the system APIs? It's apple lol [22:03:18 CEST] <furq> and also avc, because the format allows you to use an avc i-frame as well [22:03:20 CEST] <CaptainAmerica> Under 'concat protocol' [22:03:38 CEST] <furq> 21:01:46 ( furq) the protocol won't work with those input formats, use the demuxer or filter [22:04:30 CEST] <CaptainAmerica> oh eurgh [22:04:40 CEST] <furq> you probably need to use the filter [22:05:05 CEST] <furq> -i foo.mp4 -i bar.wmv -filter_complex concat=n=2:v=1:a=1 [22:05:48 CEST] <furq> blue_misfit: what if i want to decode these images on a non-apple system [22:06:09 CEST] <furq> without buying five (at least) different licenses [22:06:21 CEST] <blue_misfit> oh well yeah sure [22:06:32 CEST] <blue_misfit> my guess is they will support jpeg export etc when sending images [22:06:46 CEST] <CaptainAmerica> Thank you @furq ... This is starting to give me a headache now :) [22:06:47 CEST] <blue_misfit> but maintain native exchange of data between apple devices for improved efficiency [22:07:16 CEST] <furq> and without doing this [22:07:16 CEST] <furq> https://www.youtube.com/watch?v=L397TWLwrUU&t=38s [22:07:57 CEST] <kerio> did mr "radiohead is such awful music" just link to a fucking judas priest song [22:08:06 CEST] <kerio> i dont even [22:08:25 CEST] <furq> is linking a video endorsement now [22:09:01 CEST] <furq> i like the idea that all the people in this video would be shocked into unconsciousness and broken glasses by a fairly sedate heavy metal song [22:11:17 CEST] <furq> anyway regardless of my feelings on judas priest they are definitely better than radiohead [22:11:41 CEST] <furq> here are some other bands that are better than radiohead: every band, including jedward, which is a band now [22:16:14 CEST] <CaptainAmerica> Thank you for your help guys on my ffmpeg issues... I'm going to pick this up again tomorrow after I've had some rest. Appreciate your help. [22:16:57 CEST] <blue_misfit> \m/ [22:20:51 CEST] <furq> my favourite take about the apple hevc stuff was "av1 is dead now...netflix and google will have to serve hevc, they won't risk pissing off iOS users" [22:21:17 CEST] <furq> as if netflix haven't already specifically done that and forced apple to have a secret mpeg-dash implementation in iOS that only netflix is allowed to use [22:21:45 CEST] <furq> and also as if google aren't going to relish giving iOS users a degraded experience compared to android (like they already do) [22:23:21 CEST] <blue_misfit> they already are serving hevc lol [22:23:28 CEST] <blue_misfit> to every 4k tv and connected device [22:23:29 CEST] <furq> netflix are for 4k, yeah [22:23:34 CEST] <blue_misfit> and it's fucking awesome [22:23:41 CEST] <furq> but i assume they'd much rather be serving av1 so they don't have to pay four different licensing authorities [22:23:43 CEST] <blue_misfit> my LG B6 looks amazing playing their dolby vision content [22:23:47 CEST] <furq> yes, it's four now [22:23:52 CEST] <blue_misfit> I mean they're part of AOM [22:23:55 CEST] <blue_misfit> so of course they will [22:24:09 CEST] <blue_misfit> once it becomes usable and there's silicon [22:24:18 CEST] <furq> yeah [22:24:32 CEST] <furq> it was just funny seeing people insisting that apple supporting hevc meant that hevc had won [22:24:43 CEST] <blue_misfit> lol I mean it means hevc is here to stay [22:25:05 CEST] <furq> i'm told that youtube supports uploading hevc now, which is a shame [22:25:07 CEST] <blue_misfit> it means providers will have to start encoding it if they're not already... eventually [22:25:22 CEST] <furq> i was hoping they were doing that for philosophical reasons in addition to them running an old-ass ffmpeg with no hevc decoder [22:25:38 CEST] <blue_misfit> but that's good news for them because having an hevc endpoint for a huge market share means we can save lots of cdn costs [22:25:45 CEST] <blue_misfit> maybe offset by hevc licensing, maybe not [22:25:58 CEST] <Filystyn> what does mean get best stream in file ? [22:26:13 CEST] <furq> i have nothing against the format itself, it's just the insane licensing [22:26:14 CEST] <Filystyn> vide av_find_best_stream [22:26:27 CEST] <furq> i'd sooner use vp9 in spite of its generally worse performance [22:26:42 CEST] <blue_misfit> sure, vp9 is fucking great! [22:28:02 CEST] <blue_misfit> it's just that if you're going to move away from avc you really want 10 bit and hdr [22:28:09 CEST] <blue_misfit> and you can do that with vp9 but it's esoteric [22:28:20 CEST] <blue_misfit> and widevine drm for vp9 is also esoteric [22:28:57 CEST] <blue_misfit> and there's also edge cases where devices that should support it (widevine vp9)... but don't quite [22:29:03 CEST] <furq> yeah i get the sense av1 is going to end up being about six months too late [22:29:23 CEST] <furq> 4k and bt.2020 isn't mainstream yet but it's close enough that big players are making decisions [22:29:58 CEST] Action: JEEB talked about the post-HEVC format at NHK's R&D labs [22:30:16 CEST] <JEEB> "are you also in the MPEG group?" "no, I'm just a random OSS guy" [22:30:56 CEST] <JEEB> blue_misfit: I thought widevine for VP9 pretty much meant ISOBMFF which on container level is specified as part of CENC? [22:31:32 CEST] <furq> JEEB: so what do you think about apple's new image format [22:31:33 CEST] <JEEB> of course the "HW support" for VP9 or 10bit VP9 is the real problem point [22:31:44 CEST] <JEEB> furq: "lol nokia" [22:31:54 CEST] <furq> which i believe is called "not webp" [22:31:57 CEST] <JEEB> HEVC already had the profile anyways [22:32:29 CEST] <furq> technicolor got to get paid son [22:34:10 CEST] <TD-Linux> JEEB, no there are webm encryption blocks for widevine [22:34:28 CEST] <JEEB> right [22:34:36 CEST] <JEEB> but they did recently define the ISOBMFF side of things, too [22:34:40 CEST] <Filystyn> what does mean get best stream in file ? vide av_find_best_stream ? [22:34:45 CEST] <Filystyn> what does it mean ? [22:35:12 CEST] <JEEB> it's an API for lazy people [22:35:34 CEST] <JEEB> libavformat has various random definitions by which it decides what is the "best" track of a type [22:35:53 CEST] <JEEB> if you are asking what that means and mean it, then do stream selection by yourself [22:36:09 CEST] <JEEB> since nobody forces y ou to use that lazy man's API [22:36:33 CEST] <JEEB> not to mention that there can be cases where the "best" according to that fuzzy logic is not what you want [22:38:33 CEST] <Filystyn> ok but what is best track of the type [22:38:40 CEST] <Filystyn> you mean there can be two tracks of same audio? [22:39:37 CEST] <JEEB> not best but "best", and if you really want to know what the flying fuck it means then go read the function's implementation [22:39:52 CEST] <JEEB> but yes, it's for cases where there are multiple tracks of the same type [22:40:22 CEST] <Filystyn> ok man now i understand well I have this for songs so i doubt this will even be used [22:40:51 CEST] <Filystyn> guess you can't tell which one is bets untill you hear it [22:40:55 CEST] <Filystyn> best* [22:41:09 CEST] <Filystyn> but when it comes to songs wel you get one stream always [22:47:38 CEST] <blue_misfit> JEEB, yeah vp9 in dash with widevine is done via webm - which again is esoteric [22:47:44 CEST] <blue_misfit> google does it for sure [22:47:56 CEST] <blue_misfit> netflix does it for their downloads as well afaik [22:48:10 CEST] <JEEB> yup, one of the ways at least [22:48:37 CEST] <blue_misfit> can you do vp9 in isobmff? [22:48:42 CEST] <JEEB> yes [22:48:45 CEST] <blue_misfit> oh wow! [22:48:55 CEST] <blue_misfit> had no clue :D [22:49:48 CEST] <JEEB> I think I saw it first getting implemented in content related systems somewhere last year? [22:50:40 CEST] <blue_misfit> neat! any particular advantage to that vs just using webm? [22:52:28 CEST] <JEEB> no idea, I haven't really looked into the support stuff on client side [23:39:32 CEST] <fiberbaby> https://www.ssllabs.com/ssltest/analyze.html?d=ffmpeg.org [23:41:28 CEST] <fiberbaby> https://blog.qualys.com/ssllabs/2017/04/05/ssl-labs-distrusts-wosign-and-st… [23:41:53 CEST] <fiberbaby> how about using Let's Encrypt [23:52:42 CEST] <furq> is the person in charge of the website even here [23:53:31 CEST] <Papi> hello [23:54:34 CEST] <Papi> I have nginx server with rtmp module. When I try to run ffmpeg... it will stuck at the end of this log... https://pastebin.com/FCPQ2jep [23:54:40 CEST] <Papi> Nothing more happens [23:55:12 CEST] <Papi> When I try to open RTMP stream in VLC it normally works [00:00:00 CEST] --- Thu Jun 8 2017

1 0

ffmpeg-devel.log.20170606
by burek 07 Jun '17

07 Jun '17

[00:11:13 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:88fabd801693: avcodec/aacsbr_fixed: Fix multiple runtime error: shift exponent 170 is too large for 32-bit type 'int' [00:11:14 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:7d61e21953f9: avcodec/sbrdsp_fixed: fix runtime error: left shift of 1 by 31 places cannot be represented in type 'int' [00:11:15 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:11e2bdd53e5a: avcodec/mlpdsp: Fix runtime error: signed integer overflow: -24419392 * 128 cannot be represented in type 'int' [00:11:16 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:45c9fe61d932: avcodec/takdec: Fix runtime error: left shift of negative value -63 [00:11:17 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:0155d1c1cb61: avcodec/aac_defines: Fix: runtime error: left shift of negative value -2 [00:11:18 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:ea4b807c0a13: avcodec/takdec: Fix runtime error: signed integer overflow: 8192 * 524308 cannot be represented in type 'int' [00:11:19 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:5ceb27b0eb7a: avcodec/vmnc: Check location before use [00:11:20 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:8bc5e90bb218: avcodec/mpeg4videodec: Check for multiple VOL headers [00:11:21 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:dd373b4027d6: avcodec/aacdec_fixed: Fix runtime error: shift exponent 34 is too large for 32-bit type 'int' [00:11:22 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:32ac3f1b1f91: avcodec/mjpegdec: Fix runtime error: signed integer overflow: -32767 * 130560 cannot be represented in type 'int' [00:11:23 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:5148395e86e6: avcodec/ivi_dsp: Fix multiple runtime error: left shift of negative value -71 [00:11:24 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:52e470097aee: avcodec/jpeglsdec: Check get_bits_left() before decoding a picture [00:11:25 CEST] <cone-248> ffmpeg 03Max Justicz 07release/2.8:87c13e4e88ec: avcodec/sanm: Fix uninitialized reference frames [00:11:26 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:75d32ff24d30: avcodec/jpeg2000dec: Check tile offsets [00:11:27 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:345f29613583: avcodec/jpeg2000dec: Fix copy and paste error [00:11:28 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:5529a6781709: avcodec/smc: Check remaining input [00:11:29 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:134171355020: avcodec/aacdec_fixed: Fix runtime error: signed integer overflow: -2147483648 * -1 cannot be represented in type 'int' [00:11:30 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:6025edc0c7c6: avutil/internal: Do not enable CHECKED with DEBUG [00:11:31 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:fc449d70cf4d: avformat/mux: Fix copy an paste typo [00:11:32 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:50fb6addc9ce: avcodec/ra144dec: Fix runtime error: left shift of negative value -17 [00:11:33 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:59436dc8f852: avcodec/mlpdec: Do not leave invalid values in matrix_out_ch[] on error [00:11:34 CEST] <cone-248> ffmpeg 03Kevin Mark 07release/2.8:fde04ca71800: doc/filters: Clarify scale2ref example [00:11:35 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:259582feaa7e: avcodec/ivi_dsp: Fix runtime error: left shift of negative value -2 [00:11:36 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:492099f9c49c: avcodec/sbrdsp_template: Fix: runtime error: signed integer overflow: 849815297 + 1315389781 cannot be represented in type 'int' [00:11:37 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:8fbe1f634fdf: avcodec/libfdk-aacdec: Correct buffer_size parameter [00:11:38 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:19556586d508: avcodec/wnv1: More strict buffer size check [00:11:39 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:7b1d93ce9ca2: avcodec/aacdec_fixed: Fix multiple runtime error: shift exponent 127 is too large for 32-bit type 'int' [00:11:40 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:0103b3ea924c: avcodec/jpeg2000dec: Check tile offsets more completely [00:11:41 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:65f38d1285f3: avcodec/jpeg2000: Fix runtime error: signed integer overflow: 4185 + 2147483394 cannot be represented in type 'int' [00:11:42 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:1d88ec2fb0e6: avcodec/snow: Fix runtime error: signed integer overflow: 1086573993 + 1086573994 cannot be represented in type 'int' [00:11:43 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:5ee31596fa90: avcodec/aacdec_fixed: Fix runtime error: left shift of 1 by 31 places cannot be represented in type 'int' [00:11:44 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:3fe7d4c403fd: avcodec/webp: Fixes null pointer dereference [00:11:45 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:ba3bc229407a: avcodec/aac_defines: Add missing () to AAC_HALF_SUM() macro [00:11:46 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:a7878fe2478e: avcodec/ra144: Fix runtime error: signed integer overflow: 11184810 * 404 cannot be represented in type 'int' [00:11:47 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:33aed7bb767a: avcodec/ra144: Fix runtime error: signed integer overflow: -2449 * 1398101 cannot be represented in type 'int' [00:11:48 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:b1777d92f9e2: avcodec/truemotion2: Fix runtime error: left shift of 1 by 31 places cannot be represented in type 'int' [00:11:49 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:bbc6bfae376d: avcodec/truemotion2: Fix passing null pointer to memset() [00:11:50 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:20b9b1fd77d4: avcodec/jpeg2000dec: Use ff_set_dimensions() [00:11:51 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:8d1cd5fa08dd: avcodec/ansi: Fix frame memleak [00:11:52 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:9a9247855633: avcodec/wavpack: Fix runtime error: signed integer overflow: 24 * -2147483648 cannot be represented in type 'int' [00:11:53 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:3a56bbe001fc: avcodec/wavpack: Check float_shift [00:11:54 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:464288cc5ea8: avcodec/acelp_pitch_delay: Fix runtime error: value 4.83233e+39 is outside the range of representable values of type 'float' [00:11:55 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:bb7df2232873: avformat/avidec: Limit formats in gab2 to srt and ass/ssa [00:11:56 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:b62a5c83fcc5: avcodec/cavsdec: Fix runtime error: signed integer overflow: 59 + 2147483600 cannot be represented in type 'int' [00:11:57 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:9c0c749c334d: avcodec/pnm: Use ff_set_dimensions() [00:11:58 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:368aa6aac74f: avcodec/ra144: Fixes runtime error: signed integer overflow: 7160 * 327138 cannot be represented in type 'int' [00:11:59 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:041ad6d2bb44: avcodec/hevc_ps: Fix runtime error: signed integer overflow: 2147483628 + 256 cannot be represented in type 'int' [00:12:00 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:653de8249dab: avcodec/cinepak: Check input packet size before frame reallocation [00:12:01 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:d1390a8670db: avcodec/wavpack: Fix runtime error: signed integer overflow: 2013265955 - -134217694 cannot be represented in type 'int' [00:12:02 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:6ebdffe73c42: avcodec/wavpack: Fix runtime error: shift exponent 32 is too large for 32-bit type 'int' [00:12:03 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:ad600ec88633: avcodec/aacps: Fix runtime error: left shift of 1073741824 by 1 places cannot be represented in type 'INTFLOAT' (aka 'int') [00:12:04 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:195b5b6cea07: avcodec/ac3dec_fixed: Fix runtime error: left shift of 419 by 23 places cannot be represented in type 'int' [00:12:05 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:6636b35ef0d0: avcodec/pafvideo: Check packet size and frame code before ff_reget_buffer() [00:12:06 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:268d84a74231: avcodec/hevc_ps: Fix runtime error: index 32 out of bounds for type 'uint8_t [32]' [00:12:07 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:591a3d263247: avutil/softfloat: Fix sign error in and improve documentation of av_int2sf() [00:12:08 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:a20e70835290: avcodec/qdrw: Fix null pointer dereference [00:12:09 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:e447d3143faa: avformat/hls: Check local file extensions [00:12:10 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:dbe771e50de5: avcodec/cavs: Fix runtime error: signed integer overflow: -12648062 * 256 cannot be represented in type 'int' [00:12:11 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:05d7c10fd30f: avcodec/tiff: Avoid loosing allocated geotag values [00:12:12 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:fb95f1d9d6d7: avcodec/mjpegdec: Check that reference frame matches the current frame [00:12:13 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07release/2.8:ffab459e4e49: Update for 2.8.12 [01:26:11 CEST] <cone-248> ffmpeg 03James Almer 07master:c1503cd95e20: fate: add fate-adts-id3v1-demux [02:48:38 CEST] <Zeranoe> Should this https://ffmpeg.zeranoe.com/forum/viewtopic.php?f=13&p=12244#p12242 be addressed and removed? These builds appear to be including fdk-aac through delayed loading. [03:40:39 CEST] <cone-248> ffmpeg 03James Almer 07n2.8.12:HEAD: fate: add fate-adts-id3v1-demux [06:37:51 CEST] <nevcairiel> that doesnt exactly fix the license situation, so they would be producing non-free builds [08:06:32 CEST] <kierank> Classic unfinished drive by commit:https://blog.savoirfairelinux.com/en-ca/2017/ffmpegs-technological-c… [08:54:39 CEST] <kevmark> 549acc9995 and 01718dc0df? [08:57:38 CEST] <kevmark> Are you referring to them adding unfinished code they have no apparent intention of maintaining? [09:08:10 CEST] <mateo`> tmm1: hello, i'll start testing your patch tomorrow (i'm currently on holiday) [10:08:36 CEST] <kierank> kevmark: exactly [10:09:23 CEST] <kevmark> kierank, I'm sure it'll get a CVE later [10:09:33 CEST] <kierank> probably [12:23:18 CEST] <cone-969> ffmpeg 03Stefano Sabatini 07master:dc70ea8c193a: lavc/mpegvideo_enc: allow low_delay for non MPEG2 codecs depending on strict_std_compliance [14:12:53 CEST] <cone-969> ffmpeg 03Ilia Valiakhmetov 07master:73d9a9a6af5d: libavcodec/vp9: ipred_dl_32x32_16 avx2 implementation [14:13:19 CEST] <BBB> atomnuker: ^^ done [14:14:10 CEST] <atomnuker> nice [15:18:59 CEST] <J_Darnley> BBB: Thank you so very much. [15:19:09 CEST] <BBB> yw dude [15:27:23 CEST] <jamrial> tdjones, atomnuker: nice seeing vorbisenc getting some love :) [15:42:49 CEST] <tdjones> Yeah, it's my GSoC project, so hopefully it gets closer to a usable state over the next several weeks [15:48:14 CEST] <atomnuker> jamrial: just found out how to pipe anything from decoders to filters in under 20 lines - https://pars.ee/temp/comp_l_ff.webm [15:49:12 CEST] <atomnuker> this eliminates any guesswork when optimizing transient detection or bit distribution [15:50:04 CEST] <atomnuker> its not hacky either, its just avframes with side data [15:50:48 CEST] <atomnuker> if I only had the idea to do this earlier [15:56:20 CEST] <atomnuker> I bet no one has ever thought of doing that, this is like discovering steam power [15:57:14 CEST] <atomnuker> and with just a single command I can find that my transient detector is too sensitive to high frequencies so I need to window my per-band transient contribution [15:58:26 CEST] <wm4> doing what? [16:00:06 CEST] <atomnuker> rolling fft graph with coloring depending on random codec data [16:00:21 CEST] <atomnuker> (red == transient frame in this case) [16:45:06 CEST] <Compn> atomnuker : make a howto on trac ? [16:55:39 CEST] <cone-969> ffmpeg 03Michael Niedermayer 07master:e4efd41b83e7: avcodec/takdec: Fix multiple runtime error: signed integer overflow: 637072 * 4096 cannot be represented in type 'int' [16:55:40 CEST] <cone-969> ffmpeg 03Michael Niedermayer 07master:c4360559ee2a: avcodec/pafvideo: Fix assertion failure [16:55:41 CEST] <cone-969> ffmpeg 03Michael Niedermayer 07master:18bca25adbae: avcodec/mpeg4videodec: Fix runtime error: signed integer overflow: 53098 * 40448 cannot be represented in type 'int' [17:29:36 CEST] <cone-969> ffmpeg 03Vittorio Giovara 07master:d9909b11d9b0: vf_colorspace: Add support for gbr color space [17:36:52 CEST] <kierank> BBB: !!! [17:37:03 CEST] <BBB> wut? [17:37:12 CEST] <kierank> "This makes the output of dct-test exact:" [17:37:28 CEST] <kierank> nice work [17:37:44 CEST] <BBB> \o/ [18:19:24 CEST] <masikh> Hello dear all, I found (and fixed) 2 devision by zero errors in ffserver. I'd like to give back the code to the community, but hell I don't know how to do a git mail patch and alike. [18:19:36 CEST] <masikh> maybe someone might like to help me on this? [18:21:25 CEST] <masikh> ffserver.c: Line: 469 -> change to -> if (cur_time == drd->time1 || cur_time == 0 || drd->time1 == 0) [18:22:06 CEST] <tmm1> there is some documentation on https://ffmpeg.org/developer.html#Submitting-patches [18:23:28 CEST] <masikh> Yes, I read, I didn't understoot [18:23:31 CEST] <masikh> hence I am here! [18:23:34 CEST] <masikh> hello! [18:24:37 CEST] <tmm1> do you know how to use git? [18:24:37 CEST] <masikh> ffserver.c: Line: 1910 (around) change to -> [18:24:42 CEST] <masikh> if (st->time_base.num != 0) { [18:24:42 CEST] <masikh> snprintf(parameters, sizeof(parameters), [18:24:42 CEST] <masikh> "%dx%d, q=%d-%d, fps=%d", st->codecpar->width, [18:24:42 CEST] <masikh> st->codecpar->height, st->codec->qmin, st->codec->qmax, [18:24:42 CEST] <masikh> st->time_base.den / st->time_base.num); [18:24:43 CEST] <masikh> } else { [18:24:43 CEST] <masikh> snprintf(parameters, sizeof(parameters), [18:24:44 CEST] <masikh> "%dx%d, q=%d-%d, fps=0", st->codecpar->width, [18:24:44 CEST] <masikh> st->codecpar->height, st->codec->qmin, st->codec->qmax); [18:24:45 CEST] <masikh> } [18:24:57 CEST] <masikh> that's it [18:25:23 CEST] <masikh> So maybe tmm1, you DO understand how to make these patches? [18:25:55 CEST] <masikh> Cause I don't, but I'd like these fixes being given to the community [18:26:21 CEST] <masikh> So if you could help me that would be splendid! [18:26:47 CEST] <tmm1> i am trying to help you [18:26:52 CEST] <masikh> Cool! [18:27:10 CEST] <tmm1> you need to make a git commit first. visit https://github.com/FFmpeg/FFmpeg/blob/master/ffserver.c and click the pencil icon [18:27:14 CEST] <tmm1> and make your changes there [18:27:28 CEST] <tmm1> pasting coding into IRC is considered spam [18:29:26 CEST] <masikh> ok, working on it 1 moment please [18:32:49 CEST] <BBB> kierank: dont forget Im equally happy someone finally wrote a real SSE2 version of the idct, or ported it from inlien assembly to real world stuff [18:32:58 CEST] <masikh> Ok, done and now? [18:32:59 CEST] <cone-969> ffmpeg 03Michael Niedermayer 07master:850cbd496f8d: avformat/utils: Slightly un-clutter code in determinable_frame_size() by using a switch [18:33:00 CEST] <cone-969> ffmpeg 03Kevin Mark 07master:482566ccc3fd: libavutil/eval: Add round function to expression parser [18:33:53 CEST] <BBB> $ grep volatile ~/Projects/ffmpeg/libavcodec/x86/*.c|cut -d: -f1|uniq|wc -l [18:33:53 CEST] <BBB> 19 [18:33:56 CEST] <tmm1> masikh: now you can email the link to your commit to ffmpeg-devel(a)ffmpeg.org [18:33:57 CEST] <BBB> kierank: ^^ [18:34:36 CEST] <kierank> If there's anything we care about then I guess we can but I doubt it [18:34:53 CEST] <kierank> Need to move on to other stuff [18:35:18 CEST] <masikh> Do you mean this link? -> https://github.com/masikh/FFmpeg/commit/50326ef5683a85eef73b3db628c6dbacc51… [18:36:33 CEST] <tmm1> masikh: yes, or you can add .patch at the end like https://github.com/masikh/FFmpeg/commit/50326ef568.patch [18:36:49 CEST] <masikh> Ok, will do [18:36:57 CEST] <masikh> Thank you very much for the help! [18:42:17 CEST] <masikh> @tmm1: Mail away! And now we wait, I guess! [18:58:13 CEST] <cone-969> ffmpeg 03Tyler Jones 07master:34c52005605d: vorbisenc: Fix memory leak on errors [21:11:25 CEST] <BBB> kierank: understood :-p [21:11:39 CEST] Action: kierank looks [21:12:26 CEST] <kierank> mpeg-2 dequant I guess [21:12:36 CEST] <kierank> but that might be one of those riced dequants [21:12:39 CEST] <kierank> which really need removing [21:13:30 CEST] <kierank> if (!(s->avctx->flags & AV_CODEC_FLAG_BITEXACT)) [21:13:30 CEST] <kierank> s->dct_unquantize_mpeg2_intra = dct_unquantize_mpeg2_intra_mmx; [21:13:37 CEST] <kierank> yeah non bitexact mpeg2 is really pointless in this day and age [21:16:54 CEST] <iive> i'm more puzzled, why it is not bitexact [21:34:37 CEST] <jamrial> iive: probably doing some tricks that makes calculations slightly off for the sake of speed [21:43:59 CEST] <iive> there are 2 C functions, the second one is called bitexact. the difference is that it does the odd/even thing from the standard. [22:01:52 CEST] <tmatth> out of curiousity, any reason "round(x)" isn't implemented as an expression? https://www.ffmpeg.org/ffmpeg-utils.html#Expression-Evaluation [22:04:10 CEST] <tmatth> nevermind, apparently that got added today (!) [22:18:55 CEST] <kevmark> tmatth, ;) [00:00:00 CEST] --- Wed Jun 7 2017

1 0

ffmpeg.log.20170606
by burek 07 Jun '17

07 Jun '17

[03:35:03 CEST] <Obliterous> anyone know how to record audio from a webcam with ffmpeg 3.3? -f alsa no longer works... [03:37:55 CEST] <Obliterous> I just get: Unknown input format: 'alsa' [03:38:37 CEST] <Obliterous> arecord -l [03:38:41 CEST] <Obliterous> D'oh [03:46:31 CEST] Action: Obliterous gets stabby with alsa [05:11:47 CEST] <zippers> I'm making video clips of all the dialogue in a movie, to help me learn Chinese. I'm using a shell script with commnads as below. The runtime is really slow - about 26 hours for a 1h46 movie. How can I make it faster? [05:11:50 CEST] <zippers> ffmpeg -i "movie.mp4" -ss 01:36:29.542 -to 01:36:33.542 -async 1 -strict -2 "Clips/01-36-29.542_01-36-33.542.mp4" [05:40:40 CEST] Action: Obliterous bangs head on wall [05:40:53 CEST] <Obliterous> Why isn't alsa a recognized input format?!?!?!?!? [05:41:25 CEST] <debianuser> Obliterous: You probably built ffmpeg without alsa support. [05:41:41 CEST] <Obliterous> how can I check that? [05:42:24 CEST] <debianuser> Check the output of configure? [05:42:42 CEST] <debianuser> how do you build ffmpeg? [05:43:02 CEST] <Obliterous> alsa is/was listed there [05:43:29 CEST] <Obliterous> Enabled indevs: alsa dv1394 fbdev lavfi oss v4l2 [05:45:00 CEST] <debianuser> Can you go to the ffmpeg build dir, run `./ffmpeg -formats` there and check if you have "alsa" among them? [05:45:58 CEST] <Obliterous> Yup, its there. [05:46:07 CEST] <Obliterous> ... [05:46:17 CEST] <Obliterous> let me check for something stupid... [05:46:25 CEST] <debianuser> which ffmpeg? ;) [05:46:29 CEST] <debianuser> `which ffmpeg`? ;) [05:46:49 CEST] <Obliterous> headwall [05:46:52 CEST] <Obliterous> headbrick [05:46:57 CEST] <Obliterous> headcar [05:47:02 CEST] <Obliterous> headrock [05:47:35 CEST] <Obliterous> 'updatedb&;locate ffmpeg' returns 3 answers... [05:47:38 CEST] Action: Obliterous sighs [05:47:57 CEST] <Obliterous> *nukepave [05:48:56 CEST] <Obliterous> fscking deviated install locations, and all in the path. [05:50:09 CEST] <Obliterous> fixt. now to see if 'make install' gives me alsa support [05:51:51 CEST] Action: debianuser doesn't like `make install`... It's usually better to build a package. But in the case of ffmpeg I never install it at all, I just do `cp ./ffmpeg ~/bin/`. :) [05:52:07 CEST] <Obliterous> yippityboo! [05:52:24 CEST] <Obliterous> now I get a completely different error~! [05:53:05 CEST] <Obliterous> THIS one I know how to deal with. must turn on streaming server. [05:56:02 CEST] <Obliterous> yay. now to check the rest of them. [05:56:02 CEST] <zippers> Can I make many clips at one time, instead of seeking for each one? Seeking is slow. [05:59:38 CEST] <Obliterous> zippers: this appears to have notes on faster seeking: https://trac.ffmpeg.org/wiki/Seeking#Cuttingsmallsections [06:03:50 CEST] <Obliterous> debianuser: thanks for helping me cure my case of the stupids [06:04:21 CEST] <debianuser> Obliterous: you're welcome! I wish all problems were that easy to solve :) [06:05:29 CEST] <Obliterous> Hah. mine have just started. I've got to get multiple webcam streams from multiple machines all visible on a public-facing web page... [06:05:54 CEST] <Obliterous> the wife wants to watch the squirrels on our back porch. from work. :-s [06:07:57 CEST] <zippers> Obliterous: My previous command is ffmpeg -i "movie.mp4" -ss 01:36:29.542 -to 01:36:33.542 -async 1 -strict -2 "Clips/01-36-29.542_01-36-33.542.mp4" [06:08:46 CEST] <zippers> Obliterous: If I change the command to the first "faster seek" example, the whole video (not just the clip) is exported. ffmpeg -ss 01:36:29.542 -i "movie.mp4" -to 01:36:33.542 -async 1 -strict -2 "Clips/01-36-29.542_01-36-33.542.mp4" [06:09:44 CEST] <Obliterous> thats because the first example uses -to 0:0:0 as a DURATION [06:10:28 CEST] <Obliterous> so if you only want 5 seconts, it would be $ ffmpeg -ss 01:36:29.542 -i "movie.mp4" -to 00:00:05 -async 1 -strict -2 "Clips/01-36-29.542_01-36-33.542.mp4" [06:11:35 CEST] <zippers> I don't know how long I want. I only know the start and end times (it's coming from an srt subtitle file) [06:12:29 CEST] <Obliterous> well, in the case of the example you posted, basic math says you only want 4 seconds.... [06:12:31 CEST] <zippers> It says "If you want to keep the original timestamps, add the -copyts option", and that makes a clip very fast, but the video and audio are blank. [06:12:56 CEST] <zippers> Yes, basic math is fine, but I'm trying to do this for 1321 items. [06:14:15 CEST] <Obliterous> Sounds like a script-worthy challenge. [06:16:05 CEST] <zippers> I think that's what I need to do - calculate all the clip lengths in advance. Times are complicated though, especially with millisecond intervals. Now I know what I'll be busy doing this afternoon. [06:16:43 CEST] <Obliterous> hopefully the fast seek method works better for you [06:21:39 CEST] <zippers> Thanks, a quick test seems to be running much faster (40 clips/minute = probably about 33 minutes runtime, instead of 26 hours) [08:06:18 CEST] <Pandela> Aye [08:06:35 CEST] <Pandela> Does ffmpeg support 360 video or spatial media playback yet? [08:24:39 CEST] <johnjay> furq does ubuntu do the same thing as debian w.r.t. stable/testing/sid? [08:25:07 CEST] <johnjay> I'm on this askubuntu question and it sounds like they call it stable and "development" [09:35:45 CEST] <madprops> Hi. I captured a video of my screen using Sharex, using an ffmpeg version provided by Collision. The problem is it seems the silent part of the audio channel was chopped off. When the video is reproduced the audio starts immidiately, ignoring the silence. Weird thing is, if I open it in a video editor it does show the two channels correctly. I tried re-encoding it and did some other offset and adding silence tricks but they didn't work. [09:39:27 CEST] <k_sze[work]> Is there a guide to modernize ffmpeg API usage? I wrote an app a while back by following a tutorial (I don't remember which one, might have been dranger's). I now get deprecation warnings if I compile the app against FFmpeg 3.x. [09:39:52 CEST] <k_sze[work]> e.g. warning: 'context_model' is deprecated [09:40:55 CEST] <k_sze[work]> warning: 'avcodec_encode_video2' is deprecated [09:40:56 CEST] <k_sze[work]> etc [11:23:22 CEST] <guarani> Hi, I have a question for legally using ffmpeg. I want to make a website for video processing, like timelapse. Is there anything else I have to do except the License Compliance Checklist in the Legal tab on the ffmpeg website? [11:25:05 CEST] <furq> if it's a website and you're using ffmpeg on the backend then you're not distributing ffmpeg [11:25:17 CEST] <furq> so you don't even really have to pay attention to that [11:25:57 CEST] <furq> you only really need to be concerned about potential codec licensing fees [11:27:15 CEST] <guarani> like H.264 ? [11:28:45 CEST] <furq> yeah [11:29:02 CEST] <guarani> you ffmpeg guys are awesome ! 1st of all - great product! 2nd - well documented and 3rd - the support and everything is just awesome! [11:33:05 CEST] <DHE> thankfully ffmpeg isn't AGPL and I'm not sure of any components that are, so that's usually fine. you're not redistributing ffmpeg itself or any of its components [15:05:42 CEST] <phillipp> hey [15:06:02 CEST] <phillipp> got a special question about extracting frames from a video. does someone have a minute? [15:06:26 CEST] <c_14> just ask your question, if someone can help you they will [15:07:54 CEST] <phillipp> ok, basically, i want to extract every 10th frame from a video, where the files are saved as frame_<frame>.png where frame is 1, 11, 21, 31 etc and not 1, 2, 3, 4 [15:09:16 CEST] <c_14> probably easiest to just 2pass the process [15:09:26 CEST] <c_14> generate every image then delete 9 out of every 10 [15:09:42 CEST] <c_14> don't think the image2 muxer supports setting increments [15:10:31 CEST] <phillipp> thats what i do right now [15:10:44 CEST] <phillipp> but it takes quite long and is also quite an overheat [15:12:15 CEST] <c_14> write a program that accepts a stream of images on stdin and writes every 10th frame out using ffmpeg [] -f image2pipe - | program [15:12:18 CEST] <c_14> ? [15:12:23 CEST] <c_14> can be a python script or something [15:14:27 CEST] <phillipp> hmm, not sure about it. i would like to have ffmpeg handle the image generation from the video [15:14:39 CEST] <furq> -i foo.mp4 -vf select='not(mod(n\,10))' out%04d.png [15:14:43 CEST] <furq> then rename the images afterwards [15:14:50 CEST] <c_14> that works too I guess [15:15:08 CEST] <phillipp> my other solution, which i tried first, was to spawn an ffmpeg command for each frame [15:15:16 CEST] <BtbN> Or just add a 0 after the %04d [15:15:20 CEST] <furq> yeah [15:15:20 CEST] <phillipp> similar to your one there [15:15:28 CEST] <furq> except that'd be out by 10 [15:15:42 CEST] <phillipp> it works with a small video but takes even longer for normal videos (tested with 5 min vid) [15:17:29 CEST] <furq> that's not really similar to what i suggested [15:18:09 CEST] <phillipp> yeah just realized, sorry [15:19:01 CEST] <furq> -i foo.mp4 -vf select='not(mod(n\,10))' -start_number 0 out%04d0.png [15:19:03 CEST] <furq> should work [15:19:23 CEST] <phillipp> let me try and see if its faster [15:22:03 CEST] <phillipp> hmm, no, it doesnt process faster [15:22:32 CEST] <furq> you're not going to get much faster than that [15:24:15 CEST] <phillipp> hmm, thats bad [15:24:49 CEST] <phillipp> guess i have to handle each frame on its own and put my worker on a server with many cores to handle it concurrently [15:25:48 CEST] <furq> that makes even less sense [15:26:03 CEST] <furq> you have to decode multiple frames to get the frame you want [15:26:07 CEST] <phillipp> why? that way i only process 920 frames instead of 9200 [15:26:15 CEST] <furq> you still process every frame [15:26:27 CEST] <phillipp> yeah but i can do it concurrent [15:26:49 CEST] <furq> yeah but you'll end up decoding the same frame multiple times [15:26:58 CEST] <furq> up to 25 times if this is x264 with normal settings [15:28:11 CEST] <furq> if it's h264 or some other codec with a multithreaded decoder you can try adding -threads n before -i [15:28:12 CEST] <phillipp> i think i have to test it on 8 or 16 cores [15:28:55 CEST] <furq> it would be faster to do that if you split by gop [15:28:59 CEST] <furq> which i guess is the only way to do it anyway [15:29:28 CEST] <phillipp> split by gop? [15:29:38 CEST] <furq> https://en.wikipedia.org/wiki/Group_of_pictures [15:32:03 CEST] <phillipp> how can i make use of it with ffmpeg? [15:32:11 CEST] <furq> !muxer segment [15:32:11 CEST] <nfobot> furq: http://ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c-… [15:33:25 CEST] <furq> the problem there is if you don't have fixed-length gops [15:33:56 CEST] <furq> which a lot of modern encoders don't by default [15:34:37 CEST] <phillipp> the first problem is that i am completely confused now, haha xD [16:28:55 CEST] <Nacht> Anyone know why, when I use -t 00:00:02.01, and I see a nice output duration of "time=00:00:02.00", I get a totally different time when I use ffprobe to check the duration of that same file? "Duration: 00:00:02.20, start: 1.480000" [16:37:53 CEST] <zerodefect> Attempting to understand how to programmatically configure multithreading for an individual filter (C-API). It looks like I have to set the correct 'thread_type' flag in both the AVFilterGraph and AVFilterContext. The confusion comes that there are two similar flags I've come across: AVFILTER_THREAD_SLICE and AVFILTER_FLAG_SLICE_THREADS. Which one do I set where? [16:51:45 CEST] <c_14> afaik FLAG_SLICE_THREADS is an internal flag stating that the filter supports slice threads [16:52:00 CEST] <c_14> >Add support for slice multithreading to lavfi. Filters supporting threading are marked with AVFILTER_FLAG_SLICE_THREADS. [16:52:02 CEST] <c_14> yeah [16:52:41 CEST] <c_14> so you'll want to set THREAD_SLICE and maybe check SLICE_THREADS (to see if the filter supports threading) [16:53:43 CEST] <zerodefect> Ok cool. Thanks c_14. I saw that in 'avfilter.c' : line 933. I wasn't checking if filter supported threading though. Let me try that. [17:08:24 CEST] <zerodefect> c_14: I'm using the dev packages of ffmpeg that are available through Ubuntu Zesty's pkg manager. Would there need to be special compiler flags for enabling threading in avfilter? [18:13:10 CEST] <c_14> zerodefect: packages probably too old [18:13:44 CEST] <c_14> hmm, although [18:15:08 CEST] <zerodefect> https://packages.ubuntu.com/zesty/libavfilter6 [18:16:07 CEST] <zerodefect> ...which looks to be 3.2.4 if I read it correctly :) [18:16:13 CEST] <c_14> should probably be fine [18:16:17 CEST] <c_14> are you getting an error or? [18:17:12 CEST] <zerodefect> No errors, I get frames pushed in, and then get them on the otherside without an issue. If I put a breakpoint in the callback 'execute' function pointer, it's never hit. [18:18:01 CEST] <zerodefect> Just been trying with encoding too, and I'm not having much success there either. Maybe I ought to right a sample app? [18:19:09 CEST] <c_14> probably a good idea [19:33:27 CEST] <ChocolateArmpits> It seems x264 attempts to use more than 1 physical cpu only when the load is high enough. [19:37:18 CEST] <DHE> x264 doesn't measure CPU load. it does as it's told. most it does is count how many CPUs are available for automatic detection [19:41:22 CEST] <kepstin> if you're encoding fairly low res video using a faster preset, you're not gonna get much advantage from the multithreading code, of course [20:06:07 CEST] <raduser> good evening #ffmpeg, is this also a good place to ask about audio encodings? I'm at my wit's end trying to figure out the difference between two .wav files. the only measurable difference is ffmpeg reports the encoder being for Lavf56.40.101 for one, and Lavf56.40.101 (libsndfile-1.0.25) for the other. [20:15:13 CEST] <ChocolateArmpits> raduser, did you try running ffprobe -show_streams input.wav and comparing stream information for both? [20:15:16 CEST] <ChocolateArmpits> DHE, so how does it initiliaze the second processor? [20:15:25 CEST] <DHE> ChocolateArmpits: huh? [20:16:41 CEST] <ChocolateArmpits> I have two physical CPUs on the target system, if the load isn't high ffmpeg running x264 encode only works on one of the cpus. If the load is high the second cpu is engaged parallely [20:16:57 CEST] <ChocolateArmpits> I'm using -threads 40 too [20:17:45 CEST] <DHE> could be related to other things, multi-socket motherboards have NUMA. I don't know how linux does NUMA in great detail. [20:18:00 CEST] <ChocolateArmpits> System is running Windows 10, not Linux [20:24:33 CEST] <raduser> ChocolateArmpits, first, great username. second, before i read your message i thought about opening up the wav files as bytes in python and trying to find the descrepency that way. I have a better idea what the problem is, but not the fix. [20:25:21 CEST] <raduser> the error i get when i send these wav files to the program that is supposed to accept them and turn them into wav forms, tells me that when the bad formatted wav file DOESNT work, it's because it (and i'm paraphrasing) found "LIST" instead of "data" [20:26:07 CEST] <raduser> looking at the first 200 bytes of each wav file shows that the one that doesn't work has the bytes spelling out "LIST" before its bytes spell out "data" which suggests that when the good wavfile is made, it is excluding this metadata [20:26:25 CEST] <raduser> but i'll run that ffprobe command to see if it's more helpful than looking at bytes in python XD [20:28:36 CEST] <raduser> ffprobe says every single listed STREAM is identical for both files [20:29:28 CEST] <raduser> sadly i don't have the slightest clue as to what LIST or data has to do with wav files [20:30:11 CEST] <raduser> i'm half tempted to use python to crop out those 4 bytes spelling LIST and seeing if it'll work XD [20:30:20 CEST] <kepstin> ChocolateArmpits: yeah, I suspect the os cpu scheduler is keeping the x264 threads all on one socket due to numa - iirc the threading code is very memory intensive. [20:30:42 CEST] <kepstin> ChocolateArmpits: but ymmv, and you might get better results in linux [20:31:53 CEST] <kepstin> raduser: what's the actual problem you're having? ffmpeg's report is just saying that there's some metadata differences between the files, but i'd expect both would work fine... [20:33:06 CEST] <kepstin> raduser: it could be that whatever app your using can't handle some of the fancier riff stuff in wav files, so consider using ffmpeg to "convert" from wav to wav to fix it [20:33:22 CEST] <raduser> kepstin: i'm using Tensorflow to turn wav files into spectrograms. when i open up an m4a file with audacity and save it as a wav file (meeting the tensorflow program's parameters as far as bit length and such) [20:33:30 CEST] <kepstin> raduser: also make sure you're not using wav files >=4gb, that just causes pain. [20:34:04 CEST] <raduser> when i give the audacity-made wav file to tensorflow everything works hunky-dory. but if i do the same conversion with ffmpeg then the same wav_to_spectrogram program fails with the hint of something like "found LIST, expected data" [20:34:17 CEST] <raduser> lol wav files are WELL under 4 gb XD [20:34:30 CEST] <raduser> i never plan on them being more than 10-20 seconds [20:34:54 CEST] <kepstin> raduser: right, that's just a bug in the program, it should just skip chunks it doesn't recognize :/ [20:35:16 CEST] <kepstin> that said, there might be an option in the ffmpeg wav muxer to make it not write them... [20:35:57 CEST] <raduser> would it be easier to change that, or since i can just open it up with the python it'll be passing through anyways just cut out the section? [20:36:24 CEST] <raduser> like if i can find that section's specifications for whatever this LIST thing is i could just found out x bytes before and/or after that string [20:36:58 CEST] <kepstin> it's a standard riff chunk, it's a 4-byte name followed by a 4-byte length field, iirc; trivial to skip. [20:37:18 CEST] <kepstin> ideally you'd just fix the wav parser in the tensorflow stuff to skip unknown chunks until it finds a data chunk :/ [20:37:19 CEST] <raduser> sounds MUCH easier then trying to understand Tensorflow [20:37:41 CEST] <raduser> absolutey that's ideal. but it's.. uh... complicated [20:37:53 CEST] <raduser> i mean, i know c++, but that thing is on a whole different level [20:46:01 CEST] <kepstin> anyways, looks like there's no way to tell ffmpeg not to write the "LIST" chunk with tags (or even just move it to a different spot, e.g. after the "data" chunk) [20:46:47 CEST] <kepstin> given the issues with the tensorflow wav parser, i wouldn't be surprised if it does something like ignore the data chunk length and treat anything after it as part of the audio, tho ... [21:00:37 CEST] <raduser> you said that chunks are 4-byte names and then 4 bytes describing length, right? coz another that might get in the way after LIST is INFOISFT [21:01:17 CEST] <raduser> nvm, duh, just google it -_- [21:04:41 CEST] <kepstin> raduser: the INFO stuff inside the LIST chunk, so if you skip the whole LIST all at once you never even see it [21:06:00 CEST] <raduser> ya that's what i'm reading, i'm going to try cutting from index of LIST to the index of data and see if it's fine with it [21:06:35 CEST] <johnjay> raduser: are you submitting a patch to ffmpeg to fix something? [21:07:01 CEST] <raduser> no [21:07:34 CEST] <raduser> this is discovering a problem with the thing using the converted file, not with ffmpeg's conversion [21:08:56 CEST] <kepstin> a patch to tell ffmpeg not to write the LIST chunk might be acceptable, and probably wouldn't be that hard to write. [21:09:58 CEST] <raduser> is it really a problem, though? [21:10:33 CEST] <kepstin> it's not usually, but some really simple software that doesn't implement a correct riff parser sometimes has trouble with it, as you've seen. [21:11:12 CEST] <raduser> i'd have thought that would be the simple software's problem [21:11:46 CEST] <raduser> if i made such a software that's what i'd think upon finding this issue [21:12:36 CEST] <raduser> just in case in the future it would be using a file converted from something other than ffmpeg that might also be including the LIST chunk for the sake of complete data [21:16:48 CEST] <kepstin> but yeah, fixing this on the tensorflow side shouldn't be that hard, probably just a matter of adding a loop in the wav reading function to skip chunks until it finds one named "data" :/ you should probably open a bug about it at least. [21:17:51 CEST] <raduser> IT WORKS! cutting it out produced a nice wavform. finally. [21:18:03 CEST] <raduser> and much rejoicing was had! \o/ [21:19:12 CEST] <kepstin> to make a "correct" wav file with the list chunk removed, you should also update the file length field on the top level "RIFF" chunk, but I think tensorflow just ignores that. [21:25:09 CEST] <raduser> as soon as it's turned into a wavform the file is getting deleted anyway, so it's not that important to me [22:13:51 CEST] <zerodefect> Trying to get mp2v mt encoding going using the C-API. I've modified an existing FFmpeg code sample to generates a .ts file containing mp2v video, but I cannot figure out why multi threaded encoding will not work (well, the ffmpeg library will not call back into my custom execute callback funcs). https://pastebin.com/y1kae2fu [22:16:06 CEST] <kepstin> zerodefect: not sure what you're trying to do? the mpeg2video encoder doesn't support multithreading... [22:18:36 CEST] <kepstin> or, maybe i'm wrong about that :/ [22:18:39 CEST] <zerodefect> I should be clearer. When I say multithreading, I'm actually trying to get parallel encoding going so that it uses my own threadpool. I tried it within a larger testing rig, but I couldn't get it to work, so I've fallen back to the next best thing which is a sample app. [22:19:08 CEST] Action: kepstin takes a closer look [22:19:12 CEST] <zerodefect> You know more than me ;) [22:20:12 CEST] <kepstin> mpeg1/2video doesn't set AV_CODEC_CAP_FRAME_THREADS but does have AV_CODEC_CAP_SLICE_THREADS, huh [22:20:18 CEST] <zerodefect> So to remove the complexity of a external threadpool, I've instead just substituted some simpler code so that the 'execute' functions just perform the operations serially (single threadeed). [22:20:48 CEST] <zerodefect> Yeah, was that a question or observation (the huh threw me off :) ) [22:20:53 CEST] <zerodefect> ? [22:21:14 CEST] <kepstin> observation. [22:21:41 CEST] <zerodefect> Ah. Yes, that is what I've observed running it in a debugger too. [22:22:39 CEST] <zerodefect> So I should mention that I'm using the FFmpeg dev packages on an Ubuntu Zesty (17.04) system which looks to be using ffmpeg 3.2.4 - not the latest and greatest, but it seems it should still work. [22:23:52 CEST] <kepstin> yeah, the slice threading code i don't expect has changed in a long time [22:29:02 CEST] <rafal_rr> Hi. I want to convert subtitles from MicroDVD format to SRT format: "ffmpeg -i 1.txt 1.srt". Unfortunately output subtitles are scaled. E.g. Text at frame 800 is placed in 5min 33 sec (24 fps) instead of 5min 20 sec (25 fps, what is correct). How to tell ffmpeg to set fps=25 ? [22:29:46 CEST] <rafal_rr> I meant "at frame 8000", of course [22:30:10 CEST] <MoPac> Hello - I have a (rookie?) resolution question. I'm making 1080p VP9 .webm clips out of software-generated .png frames (not photos) that I can create at any arbitrary resolution. Is there a significant benefit to creating the .pngs in higher res (4k? 8k?) and then telling ffmpeg to downscale? [22:31:19 CEST] <MoPac> I've read about the benefits of bigger-than-native frames, but most of that discussion is about photos because of cameras' limitations [22:34:35 CEST] <kepstin> zerodefect: yeah, not sure what's up. it looks like you've put in enough to trigger multithreaded encoding, fwiw, so there should be at least one call to the execute() function - it doesn't use execute2 - on most frames, I think [22:36:16 CEST] <zerodefect> Thanks for taking a look. Do you think I ought to try the mailing list (Libav-user)? [22:36:22 CEST] <kepstin> MoPac: probably the only reason to generate at higher res and downscale is if your generating tool doesn't anti-alias the images [22:36:48 CEST] <kepstin> MoPac: in which case, rendering higher res and downscaling is, well, basic supersampling anti-aliasing. [22:37:22 CEST] <MoPac> kepstin: ah, thanks, I'll have to see whether it does or not [22:40:52 CEST] <kepstin> rafal_rr: use the '-subfps' input option (see "ffmpeg -h demuxer=microdvd") [22:44:55 CEST] <rafal_rr> kepstin: it works, thanks! I couldn't find it before. many thanks [00:00:00 CEST] --- Wed Jun 7 2017

1 0

ffmpeg-devel.log.20170605
by burek 06 Jun '17

06 Jun '17

[03:06:19 CEST] <cone-544> ffmpeg 03Kevin Mark 07master:4af496473a06: FATE: Add test for libavfilter/scale2ref [03:06:19 CEST] <cone-544> ffmpeg 03Sysiphus 07master:caf7d6178a4d: avformat/hls: Check local file extensions [03:21:40 CEST] <jamrial> lol [03:32:16 CEST] <kevmark> I have no idea what's going on in that thread but that's a quality commit right there [03:40:02 CEST] <Compn> its a CYA commit [03:40:12 CEST] <Compn> hey... we tried :P [03:40:35 CEST] <Compn> not me but the royal "we" [03:41:26 CEST] <kevmark> CYA as in "hey, we patched it" yes? haha [03:54:52 CEST] <kierank> jb_pissed is going to love that [04:35:01 CEST] <cone-544> ffmpeg 03James Almer 07master:933dd62288ba: x86/aacpsdsp: optimize ff_ps_mul_pair_single_sse [07:11:12 CEST] <cone-544> ffmpeg 03Rostislav Pehlivanov 07master:10b7adf79d70: fate: add test for the Dirac low delay profile [07:15:18 CEST] <atomnuker> hls sucks and no one should ever use it [07:17:46 CEST] <kevmark> As someone without push access to the ffmpeg git should I be signing off on my patches? The developer's guide seems to imply that I should [07:18:20 CEST] <kevmark> Or is signing off only for devs with commit access? [07:41:43 CEST] <atomnuker> nah, its for yourself [07:42:10 CEST] <atomnuker> some don't sign patches and that's okay too [07:48:09 CEST] <kevmark> Okay, thanks [08:15:35 CEST] <rcombs> holy fuck [08:15:41 CEST] <rcombs> the DASH people have gone off the deep end [08:16:01 CEST] <rcombs> they wrapped WebVTT in MP4 [08:16:20 CEST] <rcombs> but couldn't figure out how to have lines overlap, so you have to split them [08:16:49 CEST] <rcombs> and include a "continuation" if a previous line is still supposed to run while the new one does, which duplicates its content [08:17:06 CEST] <rcombs> (this, in a format that requires a full index!) [08:17:28 CEST] <rcombs> and you have to code empty regions [08:18:04 CEST] <rcombs> and to distinguish between "continuation" cues and regular cues and empty cues (a category unto themselves!), they PUT ATOMS IN THE MDAT [08:18:29 CEST] <ritsuka> so, same way as tx3g [08:21:19 CEST] <rcombs> looks like it's similar but not quite identical [08:21:27 CEST] <rcombs> but who thought tx3g was a good model for anything [08:21:37 CEST] <JEEB> I looked mostly at the ttml side which seemed more sane [08:21:41 CEST] <rcombs> and how the hell am I supposed to parse/generate this [08:21:59 CEST] <JEEB> since there was an id for each line [08:22:17 CEST] <rcombs> my current reaction is to tell the people asking me for it to go jump in a lake [08:22:38 CEST] <JEEB> :D [08:22:48 CEST] <rcombs> why the hell would you want to put WebVTT in MP4 anyway [08:23:02 CEST] <rcombs> I could maybe see it if was a track in a mux [08:23:12 CEST] <rcombs> but this is for single-track files for DASH [08:23:28 CEST] <rcombs> and they apparently end up parsing this shit in JS [08:23:40 CEST] <ritsuka> :| [08:23:40 CEST] <rcombs> do they not know you can transfer text files over HTTP [08:23:55 CEST] <rcombs> seriously, that's the only use-case I can find [08:24:07 CEST] <rcombs> also the format manages to have more overhead than the WebVTT text format [08:24:25 CEST] <rcombs> yup, they muxed a text-based format into binary and _it got bigger_ [08:24:33 CEST] <rcombs> only MP4 [13:56:54 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07master:c0702ab83018: Revert "avformat/hls: Check local file extensions" [13:56:55 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07master:189ff4219644: avformat/hls: Check local file extensions [14:18:08 CEST] <kierank> LOL [14:31:48 CEST] <cbsrobot> trac seems down [14:56:58 CEST] <BtbN> works for me [15:00:59 CEST] <cbsrobot> yeah works again, not sure what happend [15:06:07 CEST] <BBB> J_Darnley: do you have a tree I can pull so I can work on your patches? [15:51:28 CEST] <J_Darnley> BBB: I will push to my gitlab in a moment. [15:51:42 CEST] <BBB> ty [15:56:29 CEST] <J_Darnley> BBB: https://gitlab.com/J_Darnley/ffmpeg/commits/mpeg2_asm [16:19:11 CEST] <BBB> J_Darnley: ty [16:19:14 CEST] <BBB> will check it out [17:08:27 CEST] <jamrial> michaelni: can you replace id3v1.aac with http://0x0.st/6k0.aac ? [17:38:02 CEST] <atomnuker> tdjones: very nice, it works perfectly [17:49:09 CEST] <tdjones> atomnuker: Thank you [17:49:57 CEST] <tdjones> I am trying to move all samples into venc->samples before processing so that overlapping actually works correctly. However, it makes it incompatible with the aac psych. [17:50:25 CEST] <tdjones> There seems to be two options, (1) write a temp (real simple) psych system until I incorporate with the model you are using for opus or (2) rewrite the internal samples array so that each channel is a pointer to samples rather than storing it linearly and using arithmetic. I'm not sure what the better route probably is [18:09:23 CEST] <cone-248> ffmpeg 03Tyler Jones 07master:610864dc36e2: avcodec/vorbisenc: Include fdsp [18:09:24 CEST] <cone-248> ffmpeg 03Tyler Jones 07master:79941602a317: avcodec/vorbisenc: Use fdsp for applying windows [18:09:25 CEST] <cone-248> ffmpeg 03Tyler Jones 07master:25260b5161af: avcodec/vorbisenc: Include bufqueue and afqueue [18:09:26 CEST] <cone-248> ffmpeg 03Tyler Jones 07master:29c13fed68ac: avcodec/vorbisenc: Use a bufqueue in encoding with smaller lengths [18:09:27 CEST] <cone-248> ffmpeg 03Rostislav Pehlivanov 07master:7fc1be9a01c5: vorbisenc: signal samples to skip [18:09:57 CEST] <atomnuker> rcombs, JEEB: one of you *did* test whether the native aac encoder correctly signals the samples to skip at the end, right? [18:13:18 CEST] <atomnuker> (I ask because when I try to measure the number of samples to skip I'm always a frame more which doesn't seem right) [18:13:49 CEST] <atomnuker> also CRAP WHY IS OUR OPUS/VORBIS MUXER NOT INSERING END_TRIMMING, this is a huge undersight [18:14:15 CEST] <JEEB> atomnuker: not the end stuff [18:14:34 CEST] <JEEB> only encoder delay is what I check [18:19:29 CEST] <atomnuker> tdjones: I'd suggest writing to a temporary buffer so you can still use the AAC psy system until mine is ready [18:19:51 CEST] <atomnuker> what you'll likely need to do however is extend the psy system I'm writing for vorbis [18:19:55 CEST] <cone-248> ffmpeg 03Vittorio Giovara 07master:2ef9fc997dba: ffprobe: Use pixdesc API to provide color space names [18:20:37 CEST] <atomnuker> more specifically, use the per-frame statistics to translate to bit distribution in the encoder [18:21:12 CEST] <atomnuker> since the one I'm writing for opus is fundamentally different (and much more complicated because the black box bit alloc algorithm) than vorbis [18:22:04 CEST] <atomnuker> where you can just say "eh, this band's bits are going to be a log of its tonality" and it would be alright [18:23:03 CEST] <atomnuker> since the psychoacoustic system already does low passing you'll not hugely overshoot the bitrate on a per-frame basis and the rate control system will do the right thing [18:25:42 CEST] <atomnuker> but in opus you have to know how many bits you'd like to use beforehand so you either have to use heuristics (like libopus) to make a guess or deliberately touch the number of bits away from equalibrium and feed it through an rc system [18:27:27 CEST] <atomnuker> (actually in vorbis you'd use the log of the 1/tonality since vorbis does well with tonal values but bad with noise while opus does well with noise but bad with tones) [18:28:41 CEST] <RiCON> atomnuker: i thought this was known "<TD-Linux> jmspeex, now we just need preskip and end trimming in ffmpeg and libav :)" [18:30:05 CEST] <tdjones> atomnuker: Using a temporary buffer felt wasteful to me, but I wasn't sure if there were actually any better alternatives. I'll do that until/unless I figure something better. [18:31:00 CEST] <tdjones> I'll take a look and see how easily I can extend the opus psy for vorbis [18:54:36 CEST] <TD-Linux> RiCON, it's a bit more subtle than that, there are bits which work and bits which don't. I think remuxing preserves the values, and correct playback happens with mpv (but not ffmpeg.c) [18:57:24 CEST] <JEEB> yea, if you encode wav to AAC in mp4 and then output wav out of ffmpeg.c with that mp4 file it will note that wav doesn't support negative timestamps and will output the padding as well [18:57:30 CEST] <michaelni> jamrial, replaced [18:58:07 CEST] <TD-Linux> for the case of opus there's a nice (decoding) test case here https://people.xiph.org/~greg/opus_testvectors/correctness_trimming_nobeeps… [18:59:41 CEST] <atomnuker> yep, tested that sample and it works [19:00:58 CEST] <TD-Linux> I assume in mpv (ffmpeg -i f.opus f.wav doesn't work for me) [19:02:15 CEST] <atomnuker> ye [19:02:18 CEST] <atomnuker> *yes [19:05:29 CEST] <jamrial> michaelni: thanks [19:33:27 CEST] <atomnuker> BBB: will you push the new vp9 avx2 soon? [19:55:13 CEST] <BBB> yes [19:55:29 CEST] <BBB> I mean its just one intra pred function right? [19:55:33 CEST] <BBB> the impact isnt that massive yet [21:14:49 CEST] <tmm1> mateo`: looks like i maybe need to pass seq header/extension into mediacodec csd-0 for mpeg2 decoding [22:26:32 CEST] <BBB> https://nokiatech.github.io/heif/ <- hevc to the rescue! [22:27:27 CEST] <atomnuker> its hell [22:27:48 CEST] <atomnuker> defines a container format based on but incompatible apparently with isobmff [22:28:02 CEST] <atomnuker> can contain jpeg h264 or hevc [22:28:06 CEST] <BBB> but it has new patents so it must be better [22:28:23 CEST] <BBB> we all know that it can only be better if its more expensive [22:28:23 CEST] <atomnuker> absolutely [22:28:38 CEST] <BBB> only worthless trash is cheaper and competes on economic terms [22:28:59 CEST] <BBB> plus we all know how succesful hevc is already :D [22:29:07 CEST] <atomnuker> there's the old saying: "there's nothing more dangerous than something free" [22:43:31 CEST] <jamrial> so now there are three hevc based image formats. the still picture profile, bpg, and heif [22:44:12 CEST] <atomnuker> there's a still picture profile? [22:44:42 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07master:1e6ee86d9254: avcodec/cavs: Fix runtime error: signed integer overflow: -12648062 * 256 cannot be represented in type 'int' [22:44:43 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07master:d7cbeab4c138: avcodec/tiff: Avoid loosing allocated geotag values [22:44:44 CEST] <cone-248> ffmpeg 03Michael Niedermayer 07master:4705edbbb96e: avcodec/mjpegdec: Check that reference frame matches the current frame [22:45:36 CEST] <jamrial> yeah, part of the hevc spec. seems to be one of the profiles heif uses as well [22:47:53 CEST] <BBB> I think this is supposed to complement the still image profile [22:48:00 CEST] <BBB> as a container layer around it [22:48:08 CEST] <BBB> (after all, you can never have too many layers) [22:48:19 CEST] <BBB> no manager was ever fired for having more layers [22:57:14 CEST] <TD-Linux> can I stream it over HLS [23:12:28 CEST] <kevmark> atomnuker, the "based on (but incompatible with)" really got me [23:15:48 CEST] <kevmark> Even in their technical comparison with other image formats some things don't seem right to me. They say WebP doesn't have "Multiple-of-90-degree rotations" but isn't that a part of Exif? Which WebP supports? [00:00:00 CEST] --- Tue Jun 6 2017

1 0

ffmpeg.log.20170605
by burek 06 Jun '17

06 Jun '17

[00:19:40 CEST] <c_14> windows? [01:17:06 CEST] <kms_> can i grab video from my canon digital camera? [01:18:27 CEST] <kms_> by using it as webcam [02:34:28 CEST] <hxla> Hello guys, I have multiple gopro video files, that has multiple data streams, and I want to merge the video, audio and one specific data stream, but I can't, I get an error that the codec is not supported (https://pastebin.com/ZxSinT0P) is there a way to do this? [02:40:23 CEST] <hendry> RE "hardware decoding generally doesn't work for anything other than 4:2:0" ... I assume you referring to 'pix_fmt yuv420p'. That iiuc https://en.wikipedia.org/wiki/YUV is like a file format. I wonder what yuvj420p means [02:40:59 CEST] <hendry> initially when i saw 420p i had it confused between profiles and resolution. kinda awful naming or ? [02:41:12 CEST] <furq> yuvj420p is full range (j for jpeg) [02:41:18 CEST] <furq> as opposed to yuv420p which is limited range [02:41:26 CEST] <furq> generally just don't use j ever [02:41:57 CEST] <furq> https://en.wikipedia.org/wiki/Chroma_subsampling [02:42:02 CEST] <furq> also that's probably the wikipedia article you want [03:17:38 CEST] <Obliterous> I just compiled ffmpeg 3.3, and its now giving me an error: Unknown input format: 'alsa' [03:18:14 CEST] <Obliterous> trying to capture video from an attached USB webcam [03:19:49 CEST] <Obliterous> Did I miss something in my build, or is this new behavior? [03:23:33 CEST] <Obliterous> https://pastebin.com/0LiAqqVx [04:39:00 CEST] <hendry> i still don't quite understand variable bit rates. is there a tool i can view a video file where i can how bitrates were adapted/changed from scene to scene ? [04:39:13 CEST] <hendry> furq: thanks for filling me in there :-) [04:40:51 CEST] <hendry> furq: http://i.imgur.com/JebYP15.png left is yuv444p and right is yuv420p, correct? [04:42:07 CEST] <furq> left is rgb [04:42:15 CEST] <furq> 444 should look the same though [04:42:20 CEST] <hendry> furq: wonder if you can tell how i captured https://youtu.be/MmNwwQngsv8?t=34 [04:42:30 CEST] <furq> youtube is always 420 [04:45:09 CEST] <furq> 1px wide #aa0000 text is a pathological worst case for subsampling [04:45:35 CEST] <furq> if it's like #ffff00 on a black background or something then there's enough contrast that the luma resolution makes up for it [04:46:24 CEST] <furq> you can see the same artifacts around the edges of much larger coloured text though [04:46:38 CEST] <furq> that's why anyone who's any good at youtube outlines their text or takes care that it's high contast [04:46:42 CEST] <furq> +r [05:06:05 CEST] <zumba_addict> just heard about middle-out compression from my brother. Not sure what he's talking about. LOL. Is x265 compression better or they can't be compared to each other? [05:06:35 CEST] <furq> compared to what [05:07:17 CEST] <zumba_addict> middle out algorithm vs the algorithm use in x265 [05:08:12 CEST] <furq> well one of them exists and the other one doesn't [05:09:02 CEST] <zumba_addict> got it [05:09:29 CEST] <furq> if you mean that dropbox thing that a bunch of clickbait sites claimed was "middle-out compression" a while back, that's specifically a jpeg compressor [05:09:35 CEST] <furq> it's not a generally applicable thing afaik [05:09:46 CEST] <zumba_addict> yeah, i saw it - https://github.com/dropbox/lepton [05:10:00 CEST] <zumba_addict> it's a github link which is good [05:10:18 CEST] <zumba_addict> what are your thoughts on x265? Is it widely acceptable now? [05:10:22 CEST] <furq> not really [05:10:39 CEST] <furq> i personally plan to skip it and wait for av1, but then i'm not a CIO at a major video company so it doesn't matter [05:10:52 CEST] <furq> but i suspect google are planning to do the same thing [05:11:23 CEST] <zumba_addict> k. I've never heard about av1. You have links? [05:11:34 CEST] <furq> https://en.wikipedia.org/wiki/AOMedia_Video_1 [05:11:42 CEST] <zumba_addict> must be successor to avc? [05:11:48 CEST] <furq> it's more or less vp10 [05:11:49 CEST] <zumba_addict> i mean, avhdc something [05:11:56 CEST] <zumba_addict> ok [05:11:58 CEST] <furq> nothing to do with mpeg-la [05:12:10 CEST] <furq> the big issue with h265 is the ridiculous licensing [05:12:18 CEST] <furq> you need like three separate licenses to use it commercially [05:12:27 CEST] <furq> which is why very few people are bothering [05:12:29 CEST] <zumba_addict> ah [05:12:42 CEST] <furq> that and there's nothing worth broadcasting in 4k yet [05:12:47 CEST] <zumba_addict> i bought a small device called odroid c2 and it can play 4k h265 [05:12:56 CEST] <zumba_addict> yeah because of the huge data [05:13:02 CEST] <zumba_addict> thanks for the link [05:13:06 CEST] <furq> afaik only netflix are doing 4k h265 [05:13:24 CEST] <zumba_addict> most likely very highly compress [05:13:49 CEST] <zumba_addict> can av1 be used in 4k? [05:13:56 CEST] <furq> i should hope so [05:14:10 CEST] <zumba_addict> ok [05:15:52 CEST] <furq> h264 is still good enough for 1080p, and av1 should be A Thing within a year or two [05:17:11 CEST] <zumba_addict> gotcha [05:32:25 CEST] <johnjay> >tfw that feeling when a zumba addict has a better ARM device than you do [09:01:25 CEST] <theodrim> Hello, is it possible to convert ass to srt without formating with ffmpeg (i.e. drop all colors/size)? Or I better off with some sed/awk? [09:17:58 CEST] <Baumfaust> hi [09:18:14 CEST] <dongs> sed/awk is never an answer regardless of waht teh question is [09:18:30 CEST] <Baumfaust> how can i set the end and of a video? [09:18:43 CEST] <Baumfaust> fmpeg -ss 00:19:30 -i dragon.mp4 -to 00:28:00 -vf fps=1/60 img%03d.png [09:18:47 CEST] <johnjay> awk is the answer. Now what's the question? [09:18:47 CEST] <Baumfaust> to does not work [09:20:31 CEST] <theodrim> johnjay, strip formating from ass to srt (i.e. drop all <font> for example). [09:20:37 CEST] <furq> theodrim: does ffmpeg -i foo.ass bar.srt not work [09:20:50 CEST] <theodrim> furq, sadly no, it left formating in tact [09:20:58 CEST] <furq> huh [09:21:01 CEST] <theodrim> I assume it's doing it by desing [09:21:06 CEST] <theodrim> design* [09:21:35 CEST] <furq> oh i forgot srt actually has colours [09:21:48 CEST] <johnjay> theodrim: i'll let you know once i master sed and awk [09:21:57 CEST] <furq> and by "actually has" i mean it doesn't but some bastards have hacked it in and people have accepted that this is good [09:22:19 CEST] <furq> maybe go via a format that doesn't have formatting like .vtt [09:22:33 CEST] <theodrim> Will try. [09:23:36 CEST] <theodrim> Yeah, it'll do, in simple way, thanks furq :) [09:24:26 CEST] <furq> you could probably do it with sed as long as none of your tags contain a > [09:25:41 CEST] <theodrim> It can be escaped by \> [09:26:01 CEST] <theodrim> But not the point, most easy way is just use vtt as format. [09:26:24 CEST] <furq> mov_text might be better because that actually doesn't support formatting [09:26:33 CEST] <furq> whereas vtt does, i just don't think ffmpeg knows about it yet [09:38:24 CEST] <theodrim> scodec text will also produce simple result. [10:16:35 CEST] <Diego__> Hi there. I'm recording webRTC video with a builtin server app that creates 2 files: audio.mjr and video.mjr for each user in the videoroom. That plugin allows me to pre-process those files into .opus and .webm files, but I want to know if ffmpeg can manipulate those .mjr files too. Also, I need to know if I can create a process to create a single output video in realtime. I mean, while the rtc server is recording and adding data to . [10:17:04 CEST] <Diego__> the information from those files and add them into a single one, even if the server is generating more files because a new user has joined the room? [10:35:43 CEST] <Diego__> To clarify my question a bit: (e.g. 2 users in a room: a user and an administrator) Room administrator calls for a recording request via API to the server, the server sends to every client connected to that room to send a recording request to the webRTC server app. The webRTC server starts recording 2 .mjr files for each user (4 in this case. 2 for each user). While the files are being recorded, what I want is that a ffmpeg process st [10:36:34 CEST] <robswain[m]> the mjr files are made by janus, right? [10:36:36 CEST] <Diego__> file. If a user joins the room and the room is in recording mode, that user will send a request recording to webRTC server app and I want that ffmpeg adds that new user [10:36:40 CEST] <Diego__> Yes [10:36:51 CEST] <robswain[m]> it's a janus-specific format [10:37:12 CEST] <Diego__> So there is nothing I can do with ffmpeg? :( [10:37:43 CEST] <robswain[m]> unless you know how to code, you have to process the mjr files with the janus post processing tool and then you can do stuff with ffmpeg on those files [10:38:33 CEST] <robswain[m]> so if you want to do something in real-time, you would need to edit the janus code, or make a feature request and wait [10:39:18 CEST] <robswain[m]> also, i know for a fact that the janus post-processing code is not meant for real-time use [10:39:41 CEST] <robswain[m]> it scans the entire file first and then re-orders the data in it [10:40:05 CEST] <Diego__> Yep, thats the problem [10:40:08 CEST] <Diego__> The question is because I want to be able to start recording, then people joins the room or start recording if they are already in the room. The problem here is that if I allow that and I want to do a mosaic output video, I need to know when the people joined the room and start adding videos in different timeline. Otherwise, I should only allow recording by administrator room own risks [10:40:36 CEST] <Diego__> I can't see another way to do it :/ [10:41:22 CEST] <robswain[m]> these questions should go to janus really, but i think the time at which recording started is written either into the mjr file or the filename [10:41:42 CEST] <robswain[m]> that can be used for some loose syncronisation [10:41:52 CEST] <robswain[m]> janus' recording is pretty basic [10:42:33 CEST] <Diego__> I tried also a JS recording for RTC, the problem is that they don't understand janus streams received in JS [10:42:35 CEST] <robswain[m]> it basically just writes the rtp packets to disk with one file per stream and some custom format which JSON in it to easily add some metadata to the file [10:43:22 CEST] <robswain[m]> you mean having one client use the MediaRecorder javascript API? [10:43:30 CEST] <robswain[m]> in the browser, and then send the file somewhere else? [10:43:31 CEST] <Diego__> Yep [10:43:46 CEST] <Diego__> I dont like it because of the user computer consumption [10:43:59 CEST] <Diego__> CPU usage* [10:44:05 CEST] <robswain[m]> taking a step back, why do you need to ship the recording somewhere else in real-time? [10:45:33 CEST] <Diego__> the output files are stored in the server and I want them in real-time to do a mosaic and "prevent" having to handle with timelines, because users joined lately to the room [10:47:41 CEST] <Diego__> I saw a mediarecorder api usage to do it. Add streams in real time and build your mosaic, but it has problems with janus stream and the room administrator cpu usage while head to the top [10:47:55 CEST] <Diego__> mediarecorder api plugin* [10:47:59 CEST] <robswain[m]> without modification, you can't do that with janus [10:48:18 CEST] <Diego__> Then I have to lead with timelines u.u [10:48:32 CEST] <robswain[m]> kurento can do some video mixing [10:49:37 CEST] <robswain[m]> or use some other MCU that mixes video and have janus as a receive-only client that records the mixed video [10:49:45 CEST] Action: robswain[m] needs to work now [10:49:46 CEST] <robswain[m]> good luck [10:49:56 CEST] <Diego__> thanks for your help [10:50:09 CEST] <Diego__> have a nice day :) [11:11:45 CEST] <guther> Hi, i have an a/v-sync problem - here's my output: https://nopaste.me/view/c501910d [11:13:25 CEST] <guther> First 17 lines is what ffplay says when i play the infile [11:13:52 CEST] <guther> At the bottom you can see my ffmpeg cnd line [11:14:09 CEST] <guther> ^cnd^cmd [11:44:46 CEST] <dorvan> hello [11:45:33 CEST] <dorvan> hi all.. [12:01:40 CEST] <hendry> furq: so the text on my tutorial video looked good to you? RE https://youtu.be/MmNwwQngsv8?t=34 [12:46:40 CEST] <Ethereco> Hello, with my self compiled minimal audio ffmpeg i got an error: Decoder (codec mp3) not found for input stream #0:0 - Full output of ffmpeg: https://pastebin.com/Zpfd4xF2 [12:52:38 CEST] <BtbN> seems like you did not build ffmpeg with an mp3 decoder [12:54:08 CEST] <Ethereco> hmm, --enable-decoder='libfdk_aac,libopus,libvorbis,mp3,flac,pcm_s16le' mp3 is included [12:54:34 CEST] <Ethereco> mp3 encoding was no problem, that works fine [13:22:13 CEST] <Ethereco> ffprobe from my build: https://pastebin.com/7bAtMGsm - ffprobe from older fullbuild: https://pastebin.com/PqaTQhtc [16:32:51 CEST] <CptnOblivious> I have another gif related question for you guys. I was using a line I found in an online tutorial for gifs. I found another tutorial with slightly different lines which produces a gif in about half the size as the first. I don't know enough about ffmpeg, can anyone tell me the difference between the two lines? Different compression? https://pastebin.com/RQfF1V0W [17:01:09 CEST] <iive> CptnOblivious: this is an interesting question, because the reason is not obvious [17:02:14 CEST] <iive> try removing the :flags=lanczos suboption. it is a method for aproximation of pixels when scaling. [17:02:27 CEST] <iive> it is supposed to be higher quality. [17:14:57 CEST] <CptnOblivious> thanks iive I'll try removing that part [17:15:50 CEST] <iive> if this is not it, then looking at the text output might give a hint what is wrong. [17:30:30 CEST] <CptnOblivious> The output came out around the same size removing that option [17:53:08 CEST] <CptnOblivious> Yeah iive there was only a 300kb difference taking out that option. The difference between the original two lines is 1mb. I usually wouldn't care but the forum I'm posting these to has a size limit, so the extra savings helps [18:14:35 CEST] <iive> CptnOblivious: try with scale=400:225:sws_dither=none [18:31:25 CEST] <Filystyn> hello [18:31:39 CEST] <Filystyn> Can you give me advice how to start with audio file ? [18:31:43 CEST] <Filystyn> open it etc [18:31:58 CEST] <Filystyn> most tutorials are about video files [18:34:16 CEST] <c_14> I assume you mean the API? [18:34:26 CEST] <Filystyn> yes [18:34:28 CEST] <Filystyn> C api [18:34:47 CEST] <c_14> https://ffmpeg.org/doxygen/trunk/decode_audio_8c-example.html [18:35:12 CEST] <c_14> there's a bunch more here https://ffmpeg.org/doxygen/trunk/examples.html [18:35:58 CEST] <Filystyn> thank you [18:36:09 CEST] <Filystyn> Ill analyse and ask if problem shall be encountered [18:53:12 CEST] <CptnOblivious> Thanks iive that seems to cut the size in half from 7.7mb to 3.5mb. I'm logged in remotely atm so can't tell the difference between quality. [18:55:13 CEST] <iive> if you have optimized colors, the quality should be the same. [18:55:33 CEST] <iive> there is a ffmpeg guide how to optimize the colors for your video. [18:58:45 CEST] <CptnOblivious> I see. That's where using the palette line works, right? [18:59:16 CEST] <iive> i guess [19:00:53 CEST] <CptnOblivious> ok thanks, I appreciate it. I'll keep messing around with it [19:01:39 CEST] <Filystyn> does this codec = avcodec_find_decoder(AV_CODEC_ID_MP2); sets codec to specified format? [19:02:13 CEST] <iive> http://blog.pkh.me/p/21-high-quality-gif-with-ffmpeg.html [19:02:59 CEST] <Filystyn> if yes how can i check what codedc uses file? [19:05:17 CEST] <iive> Filystyn: ffprobe should be the tool for you :D [19:06:29 CEST] <Filystyn> thank you [19:11:13 CEST] <Filystyn> is there only pure source as referance? [19:17:08 CEST] <Filystyn> well ?:) [19:17:19 CEST] <Filystyn> come on guys give me few hints [19:27:52 CEST] Action: Obliterous yawns and wakes up. [20:54:42 CEST] <the_k> can ffmpeg keep an eye on free disk space? [20:59:10 CEST] <ChocolateArmpits> I would guess not [21:14:55 CEST] <Filystyn> use system functions [21:15:15 CEST] <Filystyn> quite easy to writwe one that keeps eye on unix in few lines [22:50:15 CEST] <pgorley> are there filters that can be hardware accelerated? [23:03:59 CEST] <BtbN> define hardware accelerated [23:04:04 CEST] <BtbN> a very few filters can use opencl [23:09:01 CEST] <pgorley> hardware accelerated as in on the gpu [23:11:45 CEST] <pgorley> BtbN: basically, i'm decoding video using vdpau, and i'd like to apply some filters to it before bringing it back to main memory [23:11:51 CEST] <pgorley> is this possible at the moment? [23:11:58 CEST] <BtbN> no [23:12:15 CEST] <pgorley> oh, ok, thanks [23:13:06 CEST] <BtbN> And also not possible to add, there are no processing functions that operate on vdpau frames [23:13:11 CEST] <ChocolateArmpits> maybe via Avisynth ? [23:13:16 CEST] <BtbN> no [23:13:24 CEST] <ChocolateArmpits> Vapoursynth ... ? [23:13:41 CEST] <BtbN> With what kinda magic should those suddenly operate on frames on the GPU? [23:13:41 CEST] <JEEB> both of those generally work with frames in RAM [23:13:43 CEST] <JEEB> instead of VRAM [23:14:04 CEST] <JEEB> although you could in theory make filters run on the GPU. but that usually means copying the frames back to VRAM from RAM [23:14:20 CEST] <BtbN> if you use cuvid for decoding, you get CUDA frames, on which you can, in theory, operate with CUDA on the GPU [23:14:25 CEST] <JEEB> in theory you could do in-VRAM stuff in vs I guess? [23:14:28 CEST] <BtbN> But sombody has to write all the filters first [23:14:32 CEST] <JEEB> but yes :) [23:14:44 CEST] <JEEB> no filters effectively means not there (yet) [23:14:50 CEST] <BtbN> There is one [23:14:54 CEST] <BtbN> vf_scale_cuda [23:15:45 CEST] <pgorley> good to know, thanks [00:00:00 CEST] --- Tue Jun 6 2017

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