December 2018 - Ffmpeg-devel-irc

ffmpeg-devel.log.20181210
by burek 11 Dec '18

11 Dec '18

[01:50:31 CET] <cone-901> ffmpeg 03Carl Eugen Hoyos 07master:9bcd40c29804: lavc/decode: Initialize return value for get_format() failure. [02:20:19 CET] <cone-901> ffmpeg 03Carl Eugen Hoyos 07master:de441ad52a4d: lavc/cbs_vp9: Make variable prob unsigned. [04:25:04 CET] <cone-901> ffmpeg 03Linjie Fu 07master:1c96d2e3998b: lavc/qsvenc: replace assert with error return [07:20:44 CET] <cone-901> ffmpeg 03Karthick J 07master:56503a692534: avformat/hlsenc: Handled error from ff_http_do_new_request() function [07:20:45 CET] <cone-901> ffmpeg 03Karthick J 07master:234926033097: avformat/hlsenc : Added an option to ignore IO errors [08:28:08 CET] <cone-901> ffmpeg 03Jun Zhao 07master:a27102521552: lavfi/vf_scale_vaapi: add scaling mode setting support. [11:37:53 CET] <cone-064> ffmpeg 03Paul B Mahol 07master:8f66d46ce5f4: avcodec: add gif parser [11:37:53 CET] <cone-064> ffmpeg 03Paul B Mahol 07master:eade85bbbd93: avformat: add gif pipe demuxer [12:02:43 CET] <thardin> looking at fate, are there no tests for encoders/muxers? [12:06:29 CET] <thardin> acodec.mak has some [12:29:24 CET] <thardin> there we go [17:35:09 CET] <cone-156> ffmpeg 03Paul B Mahol 07master:49d792fff62e: avcodec/proresdec2: allow changing resolution [17:35:09 CET] <cone-156> ffmpeg 03Paul B Mahol 07master:5e0d54a031e3: avformat/mxfenc: allow muxing prores [17:38:17 CET] <cone-156> ffmpeg 03Paul B Mahol 07master:14156e607d49: avformat/mxfenc: calculate and store DAR from user SAR [17:47:29 CET] <durandal_1707> why mxf muxer hardcodes aspect ratio for some codecs? [17:58:48 CET] <cone-156> ffmpeg 03Paul B Mahol 07master:8affd753c86d: avcodec/gifdec: truncate too big width/height for invalid gif files [18:45:33 CET] <durandal_1707> ubitux: are you working on gif transparency patch? [18:52:57 CET] <ubitux> durandal_1707: n [18:52:59 CET] <ubitux> no [19:01:10 CET] <gnafu> The saga continues. [19:04:49 CET] <durandal_1707> ubitux: i want to fix muxer & encoder so -c copy works, mainly allowing encoder to write extension and muxer to modify only delay in it, is this sane to you? [19:06:41 CET] <ubitux> durandal_1707: you mean moving part of the header from the muxer to the encoder? [19:08:41 CET] <durandal_1707> ubitux: yea, demuxer (and parser) only split gif stream, and extension comes with image block, while muxer extracts stuff breaking -c copy [19:09:24 CET] <ubitux> what happens to let's say, the loop option? [19:09:34 CET] <ubitux> does it stays on muxer side or does it end up in the encoder? [19:09:46 CET] <ubitux> what about the palette? [19:10:04 CET] <durandal_1707> loop option is overwritten by muxer [19:10:43 CET] <durandal_1707> local palette do not need to be exported any more ... [19:11:18 CET] <ubitux> ok well it sounds good on the paper [19:11:46 CET] <ubitux> i remember thinking about this but gave up due to packet format change between libs, requiring to bump everything [19:12:00 CET] <ubitux> or make extra random check for synchronicity [19:12:12 CET] <ubitux> the usual pita coming from multiple libs with different versions [19:12:32 CET] <durandal_1707> that was never supported [19:13:29 CET] <durandal_1707> main reason is to make -c copy working and to not need disposal exported from encoder as side data to muxer [19:14:15 CET] <ubitux> well if this politic wrt lib async is official, i can probably dig a lot of shit to submit [19:14:37 CET] <durandal_1707> what shit? [19:14:49 CET] <ubitux> iirc i had that issue with at least some subtitles [19:15:03 CET] <ubitux> iirc it was applied with random workaround [19:15:18 CET] <ubitux> like trying to detect which format it was reading the payload [19:15:28 CET] <ubitux> and take different codepath according to it [19:15:40 CET] <ubitux> i had to do that several times in the past [19:15:49 CET] <ubitux> not sure if i pushed everything [19:16:41 CET] <durandal_1707> nothing test async versions and claiming async should be supported and reason for not cleaning stuff is very bad and ugly [19:19:17 CET] <durandal_1707> what means "comply retroactively to terms of GPL" ? to send complete source code? [19:20:09 CET] <JEEB> yes. the whole binary and everything it is dependant on ("derivative work" of) [19:25:36 CET] <kierank> as if he's going to do that [19:27:08 CET] <JEEB> if that was part of his reply then in his mind it was mostly about just releasing the FFmpeg etc OSS software's code for his binary, and hope that it passes through [20:00:18 CET] <atomnuker> "We also use #Slack and host VAAPI Media Slack Team. You can signup by submitting your email address to our Slack Team invite page." [20:00:44 CET] <atomnuker> wow, I didn't expect to see that in any readme yet, even on an intel open source project [20:02:11 CET] <durandal_1707> atomnuker: answer me! [20:08:24 CET] <atomnuker> yeah, I'm not playing anything atm. [20:09:16 CET] <durandal_1707> you should (cos)play spiderman, kierank says its assom [20:09:34 CET] <JEEB> the new spider-man seemed alright [20:59:51 CET] <cone-156> ffmpeg 03Carl Eugen Hoyos 07master:fe300b2ab5d1: lavc/xpmdec: Allow more colours per character. [21:41:25 CET] <cone-156> ffmpeg 03Paul B Mahol 07master:6a20876b6203: avformat/gif: simplify signature writing [21:41:26 CET] <cone-156> ffmpeg 03Paul B Mahol 07master:00502370f6be: avcodec/xpmdec: fix small artifacts [21:41:27 CET] <cone-156> ffmpeg 03Paul B Mahol 07master:03beac5b97e8: avcodec/xpmdec: define constants [00:00:00 CET] --- Tue Dec 11 2018

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ffmpeg.log.20181210
by burek 11 Dec '18

11 Dec '18

[00:54:18 CET] <pi-> pink_mist: https://unix.stackexchange.com/questions/28803/how-can-i-reduce-a-videos-si… [00:55:59 CET] <pink_mist> pi-: you have two -vcodec arguments though [00:56:05 CET] <pink_mist> pi-: it doesn't make sense [00:57:05 CET] <pi-> my bad [00:57:11 CET] <pi-> It seems the second one overrides [00:57:17 CET] <pi-> tho [05:07:17 CET] <utack> what the hell is going on with webp...lossless animated webp comes out 250kb, "quality 50" webp comes out at 1mb and looks totally blurred [05:09:17 CET] <kepstin> iirc, the lossless mode in webp is basically a completely different format from the lossy mode. That sort of difference isn't really surprising if you're doing something like pixel art. [05:09:47 CET] Action: kepstin would probably recommend a vp9 webm, possibly in lossless mode, instead of webp for animations. [05:12:15 CET] <utack> kepstin that is exactly what i tried yeah, pixel art. i suspected it works well as webp animated [08:32:02 CET] <fling> Can I somehow add rtc data to a stream? [08:32:13 CET] <fling> I want to know the exact time of the stream capture start. [14:56:04 CET] <danieel> any tip how to merge all the dash segments i have downloaded? [14:56:13 CET] <danieel> got separate audio and video streams [14:59:54 CET] <pink_mist> I just let youtube-dl do it for me [15:01:12 CET] <danieel> will youtube-dl work for our ad-equpped local/nation TV website? [15:03:22 CET] <danieel> well.. its doing something [15:06:10 CET] <pink_mist> usually [15:06:45 CET] <pink_mist> https://rg3.github.io/youtube-dl/supportedsites.html it supports quite a number of sites [15:08:00 CET] <danieel> thanks! was not aware of that [16:25:59 CET] <th3_v0ice> How is ffmpeg handling the timestamp wrap around? I guess 32bit timestamps will run out pretty quickly, what happens then? [16:26:52 CET] <th3_v0ice> For example if a timebase is 1/90000 [16:27:22 CET] <Mavrik> IIRC internally timestamps are 64-bit [16:27:50 CET] <Mavrik> And there are wraparound handling modes for some demuxers which try to do the smart thing when expanding timestamps to 64-bit [16:28:05 CET] <Mavrik> But generally ffmpeg handles timestamp wraps and jumps rather poorly :) [16:38:04 CET] <JEEB> lavf has wrap-around handling but for MPEG-TS for some reason it onl works once [16:38:30 CET] <JEEB> after that if you're using ffmpeg.c the left-over wrapping logic there will instead attempt at doing things [16:38:56 CET] <JEEB> which is less than perfect since that thing only checks if the intra timestamps went more than 5 or 10 seconds forwards/backwards [16:39:47 CET] <JEEB> if I had time I'd check what the lavf wrap-around logic is herping a derp with [16:39:59 CET] <JEEB> because it's a weird feeling when the first case gets handled "for free" [16:40:08 CET] <JEEB> but then suddenly the API consumer has to care afterwards [16:44:33 CET] <Mavrik> Yeah, I had to patch a few things to get MPEG-TS working reliably [16:44:37 CET] <Mavrik> There's also things like vs [16:44:45 CET] <Mavrik> vf_fps not considering the wrap-around case [16:44:49 CET] <Mavrik> And some other filters as well [16:45:01 CET] <JEEB> at the point of the filter there really should not be wrap-arounds [16:45:11 CET] <JEEB> if any wrap-arounds get to that point something's failed [16:45:49 CET] <Mavrik> It [16:45:55 CET] <Mavrik> Eh, sorry, broken keyboard. [16:46:07 CET] <Mavrik> Anyway, non-linear PTS/DTS are a thing in most TV live streams I've seen [16:46:12 CET] <Mavrik> Which are a similar case :) [16:46:50 CET] <JEEB> you mean there being a reason for a timestamp jump? which is mostly some seconds forwards or so? [16:47:08 CET] <Mavrik> I've seen jumps all over the place since the providers spliced streams together [16:47:35 CET] <JEEB> yes, that I see mostly with subtitle tracks and you generally can fix that by adding "drop subtitle packets until PCR has been received" [16:47:55 CET] <Mavrik> I've seen it with all tracks - e.g. local video providers splicing in localized commercials [16:48:12 CET] <Mavrik> So you get a PTS/DTS jump (which is unrelated to previous stream) for a few minutes in your stream [16:48:20 CET] <Mavrik> On all streams not just the subtitles [16:48:39 CET] <Mavrik> Or jumps only on audio track for no apparent reason [16:48:58 CET] <JEEB> yea, the subtitle stuff I mostly see when some people decide it's a good idea to mux in dvb subs or teletext from another version of the channel [16:49:06 CET] <JEEB> and not fixing PTS [16:49:24 CET] <JEEB> so until you receive PCR lavf will blindly output completely bonkers values [16:49:29 CET] <Mavrik> Of course in hindsight I should probably handle all of that in demuxing layer and never output non-linear PTS on that project [16:49:31 CET] <JEEB> there's a separate logic for when PCR is around [16:49:36 CET] <Mavrik> But I was yound and stupid when starting out :D [16:50:16 CET] <JEEB> so I've been just thinking of looking up if the behavior can just be fixed by dropping "untrusted" streams' packets until PCR is received [16:50:24 CET] <JEEB> which is usually subtitle/teletext [16:50:44 CET] <JEEB> then there's of course random DTS/PTS jumps due to splicing [16:51:02 CET] <JEEB> most receivers just look at the PCR and attempt to fix it up [16:51:17 CET] <JEEB> or receipt time [16:51:26 CET] <JEEB> which is what upipe also saves in its packets [16:51:47 CET] <JEEB> a lot of stuff gets simpler if you know you're supposed to be running live :P [16:52:16 CET] <JEEB> "timestamps jumped by amount X but this packet I received was within the space for X Hz" [16:52:37 CET] <Mavrik> Yeah, I had a case where receivers weren't on the same machine but the input was received via UDP from somewhere else [16:52:43 CET] <Mavrik> So that data was out of whack to [16:52:55 CET] <Mavrik> But at least the FPS and audio samplerate was constant [16:54:15 CET] <JEEB> well with live inputs, even if not the receiver, you can get your app's receipt time [16:56:40 CET] <durandal_1707> is this reported as bug on trac? [16:56:52 CET] <JEEB> which thing? [16:58:20 CET] <durandal_1707> pts issues [17:00:05 CET] <JEEB> there's two bugs that I know in lavf or the mpegts demuxer that I have just discussed: 1) the wrap-around handling only seems to work once (so you're going to have to let the client handle it suddenly as it happens the second time) - I would have reported this if I had a sample, I'm just really lazy to make a low low bit rate sample with like 26*N hours of content [17:00:19 CET] <JEEB> and 2) is that lavf trusts untrustworthy PTS [17:00:53 CET] <JEEB> (it processes possibly untrustworthy tracks before PCR is received - although in theory you *always* have to wait for PCR) [17:01:36 CET] <JEEB> I have a sample for 2) but I'm not sure if I made a ticket, I might have linked the sample on -devel at least [17:01:51 CET] <JEEB> because it's mostly a timing issue [17:02:04 CET] <durandal_1707> cant one for 1) just modify pts via lavfi and store it in file as is? [17:02:46 CET] <JEEB> well, to replicate you have to use a demuxer that's set to utilize the internal wrap-around logic [17:03:24 CET] <JEEB> see ff_read_packet [17:03:42 CET] <JEEB> it has update_wrap_reference and wrap_timestamp calls [17:03:57 CET] <JEEB> that seems to only work once [17:16:04 CET] <th3_v0ice> Mavrik, JEEB, thanks for the info! [17:46:35 CET] <skyroveRR> Hey guys, I'm trying to compile ffmpeg *on* the raspberry pi 3, using GCC 8.2.0. When I run "./configure --prefix=/usr", I get this error: http://termbin.com/zeui . Here's the ffbuild/config.log: http://termbin.com/pj5j4 ... to summarize, the configure script fails with "cc1: error: -mfloat-abi=hard: selected processor lacks an FPU. C compiler test failed." [17:46:44 CET] <skyroveRR> Any way to fix this? [17:55:15 CET] <skyroveRR> Oh, and this: gcc -march=armv5te -c -o /tmp/ffconf.VOX6I1FW/test.o /tmp/ffconf.VOX6I1FW/test.c [17:59:49 CET] <pink_mist> does https://trac.ffmpeg.org/wiki/CompilationGuide/RaspberryPi give you any new insight? [18:01:06 CET] <skyroveRR> Nope, they haven't been helpful. [18:01:13 CET] <skyroveRR> And I'm not cross-compiling. [18:01:48 CET] <pink_mist> maybe you should [18:01:57 CET] <skyroveRR> Nope. [18:03:17 CET] <pink_mist> either way, that page shows some configure flags you might want to try [18:03:23 CET] <skyroveRR> I did try. [18:03:29 CET] <skyroveRR> They aren't working. [18:03:30 CET] <pink_mist> you only showed --prefix=/usr [18:03:37 CET] <skyroveRR> One sec. [18:05:58 CET] <skyroveRR> http://termbin.com/0rmjb [18:09:11 CET] <pink_mist> the page I linked showed --arch=armel which you don't have [18:09:25 CET] <skyroveRR> My toolchain is armhf. [18:10:48 CET] <skyroveRR> Same output with --arch=armel. [18:11:03 CET] <TheRock> can i compile ffmpeg with openssl 1.1.1 [18:11:07 CET] <TheRock> or only 1.0.x [18:28:51 CET] <TheRock> ffmpeg config tests ask for error C2065: 'SSL_library_init': undeclared identifier [18:28:58 CET] <TheRock> but thats not exported by openssl 1.1.1 [18:29:13 CET] <skyroveRR> There's no mention of "armv7l" anywhere in the configure script. Most other ARM architectures are listed though. Here's my uname -a: Linux darkstar 4.9.59-v7+ #1047 SMP Sun Oct 29 12:19:23 GMT 2017 armv7l GNU/Linux [19:08:55 CET] <TheRock> is openssl 1.1. supported by ffmpeg? [19:09:08 CET] <TheRock> cl -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_USE_MATH_DEFINES -D_CRT_SECURE_NO_WARNINGS -D_CRT_NONSTDC_NO_WARNINGS -D_WIN32_WINNT=0x0600 -nologo -c -Fo./ffconf.vFbwzGpS/test.o ./ffconf.vFbwzGpS/test.c [19:09:08 CET] <TheRock> test.c [19:09:08 CET] <TheRock> ./ffconf.vFbwzGpS/test.c(3): error C2065: 'SSL_library_init': undeclared identifier [19:09:25 CET] <TheRock> that function does only exist in 1.0.x [19:11:11 CET] <JEEB> which version of FFmpeg are you building? [19:11:21 CET] <JEEB> if it's master, then make a ticket on trac about it [19:21:01 CET] <trashPanda_> Hello, using code (not command line) ffmpeg, and im trying to reduce the latency in my pipeline. I'm outputting as a mpeg2ts and it appears that the "av_interleaved_write_frame()" call buffers frames for about 8 seconds before outputting them. [19:21:08 CET] <trashPanda_> Is there a way to reduce this latency? [19:21:39 CET] <trashPanda_> I'm assuming there is some buffer that you can adjust, but I am unsure where to find the commands for it/how to give it. [19:28:43 CET] <JEEB> trashPanda_: first of all, I am extremely sorry for the function's naming even though I had no part in naming it back in the day (takes in AVPackets, is named "write_frame") [19:29:18 CET] <JEEB> second of all, I guess you've already made sure your latency in other parts of the flow is OK, and that it's just the muxing/protocol writing that has the latency [19:31:10 CET] <trashPanda_> Yeah, it seems to be in the write call exclusively. Everything runs through the pipeline as its read, and gets to the write call within a second. However, if I try to watch the output, it takes a solid 8 seconds or so to show up. [19:31:27 CET] <trashPanda_> Then holds that latency throughout its running. [19:31:50 CET] <JEEB> well that time-to-start can be just no random access points for a while, though? [19:32:19 CET] <JEEB> (also time-to-start and overall latency should be two separate things IMHO) [19:32:29 CET] <JEEB> you can have 10 seconds of time-to-start (of playback) [19:32:37 CET] <JEEB> but still sub-second latency of the whole stream [19:33:05 CET] <trashPanda_> I was unaware of that difference [19:34:05 CET] <JEEB> well f.ex. x264 by default does a random access point at ~240 frames, which is close to 10 seconds on many common frame rates [19:34:18 CET] <JEEB> or was it 250, I quite honestly don't remember 8) [19:34:35 CET] <JEEB> but the point is, your client can be om nom noming the stream, which by itself is relatively low latency [19:34:42 CET] <JEEB> but you will only get your first video frame decoded [19:34:48 CET] <JEEB> after the first random access point you receive [19:34:53 CET] <JEEB> so yes, those two are 100% separate :) [19:34:59 CET] <trashPanda_> Is that different than an I-frame? [19:35:11 CET] <JEEB> yes [19:35:18 CET] <JEEB> I frame is just a single frame that is intra-only [19:35:36 CET] <trashPanda_> I was under the assumption that with mpeg2ts it would start decoding once an Iframe was received [19:35:58 CET] <JEEB> in H.264 a random access point was either an IDR frame, or an I frame with a special open GOP random access SEI prepended [19:36:25 CET] <JEEB> in HEVC/H.265 they actually standardized the stuff into separate NAL units which specifically fell under the "IRAP" naming [19:36:33 CET] <JEEB> (intra random access point) [19:36:54 CET] <JEEB> additionally you then need the headers/initialization data [19:37:41 CET] <JEEB> so your first decode'able frame is <I got the headers/initialization data> + <I got my first intra-only frame> , *but* since it's not a random access point necessarily generally that's not the first frame where decoders start giving you output [19:37:59 CET] <JEEB> that is generally limited to <I got the headers/initialization data> + <I got my first random access point frame> [19:38:12 CET] <JEEB> in the FFmpeg API "keyframe" does tend to mean "random access point" [19:38:18 CET] <JEEB> and that is another thing I am terribly sorry for [19:38:26 CET] <JEEB> even though I had nothing to do with that naming choice in the early 2000s [19:38:27 CET] <JEEB> :) [19:38:39 CET] <JEEB> (I mean the AVPacket key_frame or keyframe field this is) [19:38:55 CET] <JEEB> that field is indeed meant to represent a random access point... usually? [19:39:26 CET] <trashPanda_> Wow thank you for all that knowledge. So you're saying that if I set the keyframe value to "true" it will generate a random access point? [19:39:45 CET] <JEEB> no, don't change what you got from the encoder [19:40:54 CET] <JEEB> what I'm saying is: "keep in mind that time-to-start is dependent on your random access point positioning" [19:41:04 CET] <JEEB> so you configure your encoder according to your requirements regarding that [19:41:30 CET] <JEEB> f.ex. if you never want x264 to make longer GOPs than 25 frames, you set the AVCodecContext's gop length thing to 25 [19:41:55 CET] <JEEB> also the mpeg-ts muxer has options regarding when it outputs some stuff [19:42:03 CET] <JEEB> check out the AVOptions in the mpeg-ts muxer :) [19:42:08 CET] <trashPanda_> ok, I think that is what I'm looking for [19:43:04 CET] <trashPanda_> what command would display that? [19:43:46 CET] <JEEB> for me generally I just open up the relevant libavformat/libavcodec file, but ffmpeg.c (the command line app) can also output the AVOptions of a component with the help option [19:43:52 CET] <JEEB> that said, I never remember the parameters for that [19:43:53 CET] <JEEB> soo [19:44:00 CET] <JEEB> > off to actual code digging for me [19:44:23 CET] <JEEB> ok, I remembered it correctly [19:44:26 CET] <JEEB> -h muxer=mpegts [19:44:44 CET] <JEEB> that gives you all the AVOptions you can set for the mpegts muxer [19:44:57 CET] <JEEB> also I'm pretty sure your protocol also has some buffering [19:45:15 CET] <JEEB> aww, too bad that doesn't work with protocols :P [19:45:21 CET] <JEEB> -h protocol=file is what I tested [19:46:31 CET] <trashPanda_> I've looked at the options but I don't see anything that would affect latency, unless I'm not reading them correctly. [19:46:49 CET] <Trel> Quick question -vcodec and -c:v are the same thing, right? [19:47:12 CET] <JEEB> vcodec is the old thing, and -codec or -c is the new thing, which you can then limit to specific streams [19:47:22 CET] <JEEB> for example -c:v:2 libx264 [19:48:31 CET] <Trel> Ok, that's even better, since I've been using -c:v over -vcodec, so if that's the newer one anyway, I'm happy! [19:48:33 CET] <Trel> Thanks! [19:48:36 CET] <JEEB> which means that the 3rd video stream encoder (or decoder if before -i) is set tolibx264 [19:48:53 CET] <Trel> 0 indexed, right? [19:49:03 CET] <JEEB> yup [19:49:11 CET] <Trel> Cool cool, thanks again [19:49:53 CET] <trashPanda_> JEEB, do you think the issue could be, not in the mpeg2ts muxer, but the "interleaved" buffer inside of the actual av_interlaved_write_frame call? [19:50:00 CET] <trashPanda_> Is there a way to reduce the buffer? [19:51:19 CET] <JEEB> sure, but so far it sounded like your largest thing was the time-to-start but it could of course be output buffering if your rates are low enough [19:51:43 CET] <trashPanda_> Well, I think I explained my "time-to-start" badly. [19:52:17 CET] <trashPanda_> What I meant was, when I initially start the pipeline there is a long time for something listening on the output to see the stream [19:52:31 CET] <trashPanda_> and that 8-10 second delay is present in the stream no matter when you tune into it. [19:52:43 CET] <trashPanda_> The "time-to-start" is next to nothing once the stream has already been running. [19:54:07 CET] <trashPanda_> So I think the issues lie more in the buffer than it holds while it correctly interleaves the output, and I was wondering if there was a way to reduce that [20:02:58 CET] <trashPanda_> JEEB, Do you know of any way to mess with that stuff? [20:03:16 CET] <JEEB> it can be either muxer side or AVIO side [20:03:51 CET] <trashPanda_> Is there a command to reduce the buffer inside of the av_interleaved_write_frame call for me to test? [20:05:01 CET] <JEEB> can't straight out say, I've mostly just poked around with custom IO callbacks [20:09:13 CET] <trashPanda_> any idea where to start looking? [20:36:57 CET] <TheRock> im back [20:37:08 CET] <TheRock> did anyone answer my question? Is openssl 1.1.1 possible with ffmpeg 4.1? [20:37:54 CET] <JEEB> 20:11 < JEEB> which version of FFmpeg are you building? [20:37:55 CET] <JEEB> 20:11 < JEEB> if it's master, then make a ticket on trac about it [20:38:10 CET] <TheRock> FFMPEG 4.1.0 [20:38:31 CET] <TheRock> Looks to me, that it isnt a bug [20:38:54 CET] <JEEB> test current master, if it still fails then it still has to be fixed (probably, if we want to keep openssl support) and then back-ported to anything that makes sense [20:39:01 CET] <JEEB> otherwise just back-ported if master works :P [20:39:10 CET] <TheRock> https://pastebin.com/NcB133D6 [20:39:27 CET] <TheRock> the config tests clearly ask for SSL_library_init [20:39:31 CET] <TheRock> which is not in 1.1 available [20:40:16 CET] <TheRock> ./ffconf.vFbwzGpS/test.c(3) [20:40:20 CET] <TheRock> i don't know where to find this file [20:40:47 CET] <JEEB> that's a temporary file [20:40:55 CET] <JEEB> you can find the contents of the file in ffbuild/config.log [20:41:04 CET] <TheRock> i would like to verify the source-code [20:41:08 CET] <TheRock> if there is an if else statement [20:41:20 CET] <TheRock> because my enviroment could be wrong [20:41:23 CET] <JEEB> you can see it in hte configure file as well as ffbuild/configure.log [20:41:37 CET] <TheRock> ah, ok [20:41:38 CET] <JEEB> <build_dir>/ffbuild/configure.log is where all that is [20:41:52 CET] <TheRock> ye, just saw it [20:42:01 CET] <TheRock> thus, openssl 1.1. is not supported currently [20:43:23 CET] <JEEB> most people making redistributed binaries use GnuTLS or on embedded stuff the other TLS library by ARM [20:43:38 CET] <JEEB> because openssl's older license at least was not GPL-compatible [20:44:53 CET] <TheRock> well, i use ffmpeg as backend. if ffmpeg uses 1.0.2u for example, and my other code 1.1., i have to ship two different openssl versions [20:45:26 CET] <JEEB> yes, then open up a ticket about it or fix it yourself and post a patch fixing it on the mailing list [20:45:40 CET] <TheRock> i'll just swtich back to 1.0.2 [20:45:43 CET] <TheRock> it will be easier [20:45:47 CET] <JEEB> also... [20:46:23 CET] <JEEB> openssl is EXTERNAL_LIBRARY_NONFREE_LIST [20:46:32 CET] <JEEB> that's not supposed to be redistributable :P [20:47:09 CET] <JEEB> gnutls or mbedtls would be [20:47:18 CET] <JEEB> so you specifically are having to enable --enable-nonfree [20:47:19 CET] <JEEB> for openssl [20:47:29 CET] <JEEB> which also specifically in the binaries says "not redistributable" [20:47:51 CET] <TheRock> i have compiled ffmpeg successful without that switch [20:47:56 CET] <TheRock> for 1.0.x [20:48:20 CET] <TheRock> and the module was listed in the overview [20:48:34 CET] <JEEB> that's weird because it *is* under the nonfree list [20:49:07 CET] <TheRock> yes. ot [20:49:09 CET] <TheRock> it's weird [20:49:26 CET] <TheRock> actually i have compiled many versions including 3.4+, before with openssl [20:49:45 CET] Action: JEEB waits for git blame to work out when each line was changed [20:49:56 CET] <TheRock> but for 4.1, i wanted to swtich to 1.1 [20:50:16 CET] <JEEB> yae, that's 2016 [20:50:24 CET] <JEEB> so it definitely has been under enable-nonfree :P [20:50:27 CET] <JEEB> for quite a while now [20:51:30 CET] <JEEB> and before that in 2011 "die_license_disabled nonfree openssl" [20:52:58 CET] <TheRock> most SO posts also just say --enable-openssl [20:53:01 CET] <TheRock> from >2015 [20:53:41 CET] <TheRock> your define might be there, but it's not included :P [20:54:50 CET] <JEEB> there's nothing to include, really :P [20:54:56 CET] <JEEB> it's within the configure script [20:55:34 CET] <JEEB> but yes, there were some that only got fired when you enabled GPL [20:56:41 CET] <JEEB> yea it smells like the nonfree things was originally added for GPL-incompatibilities [20:56:45 CET] <JEEB> because of "enabled gpl && map "die_license_disabled_gpl nonfree" $EXTERNAL_LIBRARY_NONFREE_LIST" [20:56:54 CET] <JEEB> so if you haven't enabled GPL that won't get hit [20:57:04 CET] <JEEB> which is correct in some cases of nonfree, but not all [20:57:11 CET] <JEEB> (jesus christ this is a mess) [20:57:13 CET] <TheRock> check_lib openssl openssl/ssl.h SSL_library_init -lssl -lcrypto -lws2_32 -lgdi32 || [20:57:25 CET] <JEEB> yea, that's the primary configure check which comes out before [20:57:55 CET] <TheRock> libeay32 and ssleay32 [20:57:59 CET] <TheRock> is correct for 1.0.x [20:58:10 CET] <TheRock> at least the git clone from openssl build the libraries with that names [20:58:19 CET] <TheRock> and from > 1.0, they are called ssl, crypto [20:58:23 CET] <JEEB> yes, anyways - make a ticket for that if you don't want to fix 1.1 compatibility yourself :) [20:59:13 CET] <TheRock> i'll go with the old version [21:00:09 CET] <JEEB> also openssl got re-licensed under Apache version 2 for newer versions so technically if you limit yourself to version3 in the future those newer versions would be buildable without enable-nonfree with GPL [21:01:01 CET] <JEEB> but yea, I /think/ current license and LGPL specifically is kosher, which is what the nonfree check covers [21:02:47 CET] <TheRock> i just need it for https [21:03:26 CET] <TheRock> which sucks, because i need to include all the libraries [21:08:34 CET] <trashPanda_> JEEB, I switched the av_interleaved_write_frame with just av_write_frame and the 10 sec lag went away completely. Its sub .5seconds now. So it definitely is an issue with the interleaved buffer size. Would you suggest just writing my own mixer? [21:09:17 CET] <JEEB> no [21:09:23 CET] <JEEB> unless you specifically need it for something [21:09:41 CET] <JEEB> there's some way of controlling how much whatever part is buffering there [21:09:49 CET] <JEEB> but I can't you tell it out of thin air for you [21:10:12 CET] <trashPanda_> I would imagine there was, I just don't know how to find it. Google is not helping atm. [21:10:51 CET] <TheRock> Compiling now with OpenSSL1.0.2q [22:38:43 CET] <trashPanda_> JEEB, thanks for helping point me in the right direction. I found the option, its an actual field in the avformatcontext, it can also be set with avoptionset, "max_interleave_delta" [22:39:44 CET] <trashPanda_> it was apparently defaulting pretty high for me, I hard set it to a quarter second and now things have next to no latency. [00:00:00 CET] --- Tue Dec 11 2018

1 0

ffmpeg-devel.log.20181209
by burek 10 Dec '18

10 Dec '18

[10:54:59 CET] <cone-066> ffmpeg 03Paul B Mahol 07master:c782e7aa9e98: avformat/dhav: export average frame rate too [12:00:24 CET] <durandal_1707> atomnuker: when you gonna stop playing games and fix opus decoder? [14:13:56 CET] <kierank> durandal_1707: when you stop trolling [14:18:03 CET] <durandal_1707> kierank: i'm not trolling, you do [14:26:27 CET] <ubitux> durandal_1707: you should really stop though [14:26:40 CET] <durandal_1707> or what? [14:27:02 CET] <ubitux> what are you, 5yr old? :/ [14:28:30 CET] <durandal_1707> 5yr old who can tell others to do their work [14:29:06 CET] <kierank> he don't have to do any work [14:29:08 CET] <kierank> it's christmas [14:29:14 CET] <durandal_1707> what? [14:29:23 CET] <durandal_1707> its 9.12 [14:29:33 CET] <durandal_1707> or 12.9 [14:32:21 CET] <durandal_1707> banana man will tell bad words about ffmpeg in general and native opus in particular if bug is not fixed ASAP! [16:02:50 CET] <thardin> hmm: https://codecs.multimedia.cx/2018/12/why-i-am-sceptical-about-av1/ [16:12:56 CET] <BBB> it's just typical kostya [16:14:13 CET] <kierank> everything is crap but bink [16:17:07 CET] <cone-901> ffmpeg 03Jun Zhao 07master:5ecd4f24f00b: lavfi/buffersink: fix abuffersink flag setting issue [16:34:14 CET] <BtbN> When do h264 patents expire btw.? [16:35:18 CET] <atomnuker> 2024 [17:41:23 CET] <durandal_1707> atomnuker: still playing game(s)? [18:49:51 CET] <cone-901> ffmpeg 03Paul B Mahol 07master:8be56ee211f9: avfilter/vf_showinfo: allow checksums calculation to be disabled [19:29:49 CET] <cone-901> ffmpeg 03James Almer 07master:d12d4d45153e: avcodec/libaomenc: add row-mt option [19:56:24 CET] <cone-901> ffmpeg 03Mark Thompson 07master:34cab708189b: qsvdec: Fix running with assert_level > 0 [20:23:38 CET] <cone-901> ffmpeg 03Paul B Mahol 07master:3981f262ac72: avformat/mxfdec: fix indentation [20:55:58 CET] <pross> bink! [20:56:34 CET] <JEEB> there was a bink2 WIP patch on kostya's site, IIRC [20:57:52 CET] <pross> well there goes my summer (its summer here) [20:59:48 CET] <durandal_1707> pross: you are working on bink2? [21:23:22 CET] <durandal_1707> i wanted also to work on bink2 [21:24:58 CET] <JEEB> I uploaded a bunch of samples earlier this year if you still might have the 0x0.st links :P [21:30:00 CET] <pross> not currently. i am knee deep in another codec [21:30:45 CET] <JEEB> :) [21:33:52 CET] <durandal_1707> does anybody know any software able to mux uncompressed video into mxf (and not just 422 packed variants but more)? [23:29:28 CET] <cone-901> ffmpeg 03James Almer 07master:04ee1b8da564: avcodec: bump micro version after d12d4d4515 [00:00:00 CET] --- Mon Dec 10 2018

1 0

ffmpeg.log.20181209
by burek 10 Dec '18

10 Dec '18

[18:58:49 CET] <FishPencil> I have a video that's been save with the smaller colorspace. How do I tell FFmpeg the input is the restricted color space and expand it to the full? [19:04:13 CET] <durandal_1707> FishPencil: you mean color range or? [19:04:44 CET] <FishPencil> durandal_1707: Yeah [19:06:04 CET] <durandal_1707> FishPencil: setrange/setparams and scale filter [19:12:05 CET] <FishPencil> durandal_1707: -vf scale=in_range=limited,setrange=full ? [19:14:05 CET] <durandal_1707> FishPencil: -vf scale=in_range=limited:out_range=full , setrange=full just changes metadata but video remains same [20:42:33 CET] <giaco> hello! [20:44:44 CET] <giaco> I've an audio file that starts at t > 0, how to reset timestamps? (I think the correct word is PTS but not sure) [21:26:28 CET] <Kanibal> Hi! I'm trying to use ffmpeg to extract a subtilte file from a mkv container like this: [21:26:30 CET] <Kanibal> $ ffmpeg -i input.mkv -map 0:s:6 output.srt [21:26:42 CET] <Kanibal> I'm getting the output: [21:26:45 CET] <Kanibal> Stream mapping: [21:26:47 CET] <Kanibal> Stream #0:9 -> #0:0 (dvd_subtitle (dvdsub) -> subrip (srt)) [21:26:50 CET] <Kanibal> But then it fails with the error: [21:26:55 CET] <Kanibal> Subtitle encoding currently only possible from text to text or bitmap to bitmap [21:27:13 CET] <JEEB> yea, DVD/DVB/PGS (last is blu-ray) subtitles are images [21:27:16 CET] <JEEB> and SRT is text [21:27:17 CET] <Kanibal> As far as I am understanding, I *do* want to extract it as text (as a subrip srt file) [21:27:22 CET] <Kanibal> oh. [21:27:30 CET] <Kanibal> damn ... [21:27:45 CET] <TheAMM> There are tools to OCR bitmap subtitles [21:27:55 CET] <pink_mist> none are very good though [21:28:01 CET] <TheAMM> They're not magic, no [21:28:10 CET] <JEEB> there's even an OCR filter in libavfilter, but IIRC you do not get that text in an easily catchable format (and ffmpeg.c wouldn't support that anyways) [21:28:11 CET] <TheAMM> but a whole lot better than manually typing them all out [21:28:21 CET] <pink_mist> I've yet to see one that gets l/I right even half the time [21:28:35 CET] <JEEB> Kanibal: you could try subtitle edit or so? [21:28:41 CET] <TheAMM> subrip at least has some mild postprocessing [21:28:42 CET] <JEEB> I think that had the least bad interface for OCR'ing [21:28:48 CET] <TheAMM> But there was a more modern tool [21:29:01 CET] <TheAMM> I'd think google can offer some [21:29:04 CET] <JEEB> subrip -> suprip -> subtitle edit [21:29:08 CET] <JEEB> if I recall correctly [21:34:14 CET] <Kanibal> JEEB: It's just that I digitalized my DVD collection and now I want to play them on my TV [21:34:40 CET] <Kanibal> but that dumb TV only takes srt-files *next* to the video file [21:35:04 CET] <JEEB> well then you will have to either convert - or grab someone else's creations :P [21:35:12 CET] <JEEB> not much else there to do [21:35:29 CET] <JEEB> since you need to either do the OCR yourself, or hope that something on the internet is as good [21:35:29 CET] <Kanibal> Should be possible. It's Game of Thrones :p [21:35:41 CET] <JEEB> yeh [21:36:08 CET] <Kanibal> Thanks for clearing up the problem though! [21:48:11 CET] <runfalk> Hi, I'm trying to generate thumbnails from videos using ffmpeg -ss 00:01 -i video.mp4 -frames:v 1 -qscale:v <quality> out.jpg [21:48:24 CET] <runfalk> how does the qscale parameter work? [21:48:31 CET] <JEEB> 100% depends on the encoder [21:48:49 CET] <runfalk> encoder being the output image format? [21:49:28 CET] <JEEB> in this case it's the (M)JPEG encoder in FFmpeg, which is then multiplexed (lol) into the JPEG format :P [21:49:39 CET] <JEEB> which probably is close to raw JPEG bit stream? [21:51:44 CET] <runfalk> I'm writing a cms-plugin that can take an arbitrary video file and output a specific frame (thumbnailed) [21:52:05 CET] <runfalk> I want to provide some way of specifying quality on something like a 0-100 scale [21:52:19 CET] <JEEB> then you have to map that according to the output encoder :P [21:52:35 CET] <JEEB> including which parameter it matches against [21:52:39 CET] <runfalk> but the output encoder will always be the same for jpeg? [21:52:43 CET] <JEEB> yes [21:52:55 CET] <JEEB> so you will have to look up what the (M)JPEG encoder does in FFmpeg [21:53:22 CET] <runfalk> all right, thank you! [21:54:29 CET] <runfalk> do you happen to know what its name in the c-code is? [22:24:41 CET] <runfalk> seems like it is integers between 2-31 [22:24:48 CET] <runfalk> 2 is high quality [23:38:37 CET] <pi-> `ffmpeg -i in_audio.wav -i in_vid.mov -acodec aac -cutoff 22000 -vcodec copy -map 0:a:0 -map 1:v:0 -ab 384k -vcodec libx264 -crf 24 out.avi` [23:38:58 CET] <pi-> Could anyone suggest why this output file plays with VLC but not QuickTime or Keynote? [23:42:25 CET] <pink_mist> what's up with -vcodec copy -vcodec libx264? [00:00:00 CET] --- Mon Dec 10 2018

1 0

ffmpeg-devel.log.20181208
by burek 09 Dec '18

09 Dec '18

[01:20:01 CET] <cone-752> ffmpeg 03Artyom Lebedev 07master:6b1c4ce8cf60: Fix bug in mpegts muxer which affects KLV async stream generation. [01:54:26 CET] <pross> cehoyos bet me to it [01:54:27 CET] <pross> cheers [11:05:58 CET] <cone-612> ffmpeg 03Gyan Doshi 07master:d91197c610e1: doc: vapoursynth [11:31:33 CET] <cone-612> ffmpeg 03Carl Eugen Hoyos 07master:936b0aa28057: doc/faq: Fix flags for possible video quality improvements. [11:31:34 CET] <cone-612> ffmpeg 03Carl Eugen Hoyos 07master:0cb1ccd1d189: doc/codecs: Do not mention a removed flags. [12:20:09 CET] <durandal_1707> what .split in parser do? [12:34:00 CET] <nevcairiel> it used to be for splitting extradata out [12:34:08 CET] <nevcairiel> but its been superceeded by the extract_extradata bsf [12:34:25 CET] <nevcairiel> not sure why it wasnt removed yet, possibly waiting out deprecation cycle [17:02:36 CET] <cone-771> ffmpeg 03Michael Niedermayer 07master:2bc3811c0d6b: avcodec/dxv: Check that there is enough data to decompress [17:02:37 CET] <cone-771> ffmpeg 03Michael Niedermayer 07master:fe0416f8a1be: avcodec/jpeg2000dec: Skip DWT if nothing is coded [00:00:00 CET] --- Sun Dec 9 2018

1 0

ffmpeg.log.20181208
by burek 09 Dec '18

09 Dec '18

[00:12:32 CET] <jeaye> I've 50 frames of an animation with which I'm building a webm file; I've done it this way many times before, with no issue, but this particular set of files seems to be resulting in a corrupted output. I don't get any errors from ffmpeg and the video plays entirely and then has visual corruption at the end. Any ideas what might be going on? [00:12:37 CET] <jeaye> CLI usage and output: https://gist.github.com/jeaye/cc4db72103a5a121df01e012e78780cb [00:20:37 CET] <jeaye> Ok, this is due to the size of the frames. The 30x30 and 80x80 versions I have show the issue, but not the 125x125 version. [01:50:21 CET] <aicra> hi. this is kind of a "user" question. I have been admiring your DX - your website - and I am curious about what you used... sphinx? jekyll? something else? [01:52:01 CET] <furq> aicra: https://github.com/FFmpeg/web [02:16:02 CET] <aicra> furq: thanks! [11:44:38 CET] <dv_> does anybody here know anything about vp8? I have a hw codec here that I am wrapping. the problem is that I dont get information from it about the frame type. specifically, I don't know if the frame is a golden or alternate reference frame. but these I have to skip, so I need to know if it is an altref/golden frame. [11:44:57 CET] <dv_> so, I need to check myself if it is a golden or altref frame. but I didn't find out how. the specs I found only speak of the golden/altref frames themselves, not how to identify them. [13:12:50 CET] <pink_mist> loads of people here know lots about vp8 ... I'm not one of them though. suggest you hand around until one of them shows up [13:13:39 CET] <pink_mist> *hang [13:13:45 CET] <pink_mist> though I guess you've already been here for an hour and a half, so I'm presuming hanging around won't be a problem [13:17:40 CET] <atomnuker> dv_: just parse the field yourself then, its in the packet header [14:38:07 CET] <dv_> atomnuker: when you say packet, what packet do you mean? webm packet? or vp8 bitstream packet? sorry I'm not familiar at all with how vp8 is encoded [14:38:37 CET] <atomnuker> vp8 bitstream packets [14:39:22 CET] <atomnuker> you could use webm packets too I suppose if you trust the muxer has done its job at marking keyframes correctly but I'm not familiar with those [15:07:57 CET] <dv_> atomnuker: so the part that is relevant for me would be from https://tools.ietf.org/html/rfc6386#section-9.1 the bit that defines whether or not the frame is to be displayed, right? [15:16:52 CET] <atomnuker> lol no [15:36:57 CET] <dv_> atomnuker: then where? [15:37:32 CET] <dv_> because it does sound a lot like it. as said, I need to know whether or not the decoded frame is to be displayed.. [15:37:47 CET] <dv_> altref frames I have to skip instead [15:43:40 CET] <atomnuker> the key frame flag is what you were asking about [16:05:19 CET] <p0a> Hello I want to video capture part of my screen [16:05:26 CET] <p0a> the commands I've seen online capture the whole screen [16:05:27 CET] <p0a> is it possible? [16:10:30 CET] <p0a> I think I found some information online [16:21:42 CET] <jkqxz> dv_: For reference frame use in the decoder, there are stored frames - "last", "golden" and "altref". These need not be distinct - they can all contain the same frame. When a new frame is decoded, it always gets stored in the "last" slot. [16:22:26 CET] <jkqxz> There are also bits in the frame header to indicate tat it should be stored in the "golden" and "altref" slots (independently, so it can be in both). [16:22:54 CET] <dv_> that's what I saw when digging through the libvpx code. slots for golden and altref. [16:23:30 CET] <jkqxz> Further, the "golden" and "altref" slots can be filled by a frame already in one of the other slots, so for example you can copy the previous "last" frame to "golden" , or any other combination. [16:25:33 CET] <dv_> well, okay. but what I need is to know when to skip / not-display a frame. [16:25:34 CET] <jkqxz> Since you can always change the type later and rearrange slots, there isn't really any sense in which a single decoded frame is "altref" or "golden" because at any given time it can be none or one or both of those. [16:26:14 CET] <dv_> do I really have to skip all keyframes ? [16:26:49 CET] <jkqxz> Sounds like you actually want the show_frame flag, which is completely unrelated to the reference frame storage. [16:27:41 CET] <dv_> yeah, I guess so. I did conflate "is it an altref frame" with "should I skip it earlier", so I caused some confusion I guess [16:42:57 CET] <dv_> jkqxz: yeah, this seems to be it, thanks! one last question: are these "invisible" frames unique to vp8, or do they also occur in vp9 and vp6? [16:46:08 CET] <jkqxz> VP9 and AV1 both have them too, and in those there is a special "display old invisible frame" command. [20:47:01 CET] <kab0m> Hello everyone [20:49:17 CET] <kab0m> Im having troubles using ffmpeg on the cli. "ffmpeg -i old.flac -acodec alac new.m4a" does not copy every metatag present in the FLAC file to the converted ALAC file. Please help me, i have quite a lot of files to convert. tia [20:50:23 CET] <alone-x> ct [20:50:28 CET] <alone-x> kab0m, hello [20:50:38 CET] <alone-x> may be u know how to encode with qsv? [20:51:06 CET] <kab0m> alone-x: no, what is qsv? [20:51:29 CET] <alone-x> gpu acceleration [20:51:34 CET] <alone-x> with intel CPU [20:52:23 CET] <klaxa> kab0m: i think you want -map_metadata https://www.ffmpeg.org/ffmpeg.html#Advanced-options [20:52:45 CET] <kab0m> alone-x: i dont really understand what gpu acceleration has to do with the tags? [20:53:15 CET] <kab0m> klaxa: thanks for this info, i allready tried "-map_metadata 0 -id3v2_version 3" but without success [20:53:22 CET] <alone-x> well, with that will be faster. [20:53:59 CET] <kab0m> alone-x: thanks for this tipp ;) but the speed is not the problem..the problem is about missing metatags after conversion [20:54:37 CET] <furq> which tags are missing [20:55:03 CET] <alone-x> well, all i need reducing CPU usage with screen capturing [20:57:08 CET] <kab0m> furq: quite a lot...like BARCODE, CD-NUMBER, ITUNESADVISORY, etc... [20:58:33 CET] <kab0m> I just want ffmpeg to copy over every metatag, not just a few...but i dont know how to do this. "-map_metadata 0" wasnt successful [20:59:31 CET] <furq> i don't think the m4a muxer will write any itunes tags [20:59:55 CET] <furq> maybe try the latest ffmpeg if you're not already on it [21:00:11 CET] <alone-x> furq hello [21:00:23 CET] <alone-x> may be you know how to usd QSV? [21:00:26 CET] <furq> i do not [21:00:30 CET] <alone-x> i have no success [21:00:40 CET] <alone-x> but with bandicam it's working [21:01:41 CET] <kab0m> furq: hmmm, i dont understand where the problem is. i mean, i can successfully copy every missing tags by hand via kid3, but this is no option for a few thousand files... [21:02:31 CET] <furq> all the tags you mentioned are itunes-specific extensions, they're not part of the spec [21:02:35 CET] <furq> so ffmpeg just doesn't bother with them [21:02:42 CET] <pink_mist> kab0m: I always used foobar2000 on windows to manage tags [21:02:43 CET] <furq> you could write a mutagen script or something that runs after ffmpeg [21:03:25 CET] <furq> http://vpaste.net/peWR6 [21:03:27 CET] <furq> something like that [21:03:34 CET] <furq> i'm pretty sure mutagen respects itunes-specific metadata [21:03:48 CET] <kab0m> pink_mist: thanks for the tip, but im on linux/mac and need automated script for this [21:05:05 CET] <kab0m> furq: thank you, i never worked with phyton before...i tried to get this working with kid3-cli but had no success so far [21:07:51 CET] <furq> ffmpeg -i foo.flac -c:a alac bar.m4a && copytags foo.flac bar.m4a [21:07:53 CET] <furq> should work [21:09:09 CET] <kab0m> is copytags a package i have to install or the phyton-script you posted before? [21:09:14 CET] <furq> copytags is the script i posted [21:09:22 CET] <furq> you'll need to install mutagen for python 3 but that should be in your repos [21:09:42 CET] <kab0m> furq: thanks for the help, i will test and report :) [21:15:29 CET] <ariyasu> hello [21:16:10 CET] <ariyasu> "ffmpeg -t 60 -i input.mkv -c copy output.mkv" [21:16:26 CET] <ariyasu> this is only copying tracks set to default and skipping all other tracks [21:16:34 CET] <ariyasu> is there someway to keep them all? [21:16:42 CET] <klaxa> add -map 0 [21:16:49 CET] <ariyasu> thanks [21:16:55 CET] <klaxa> ffmpeg -t 60 -i input.mkv -map 0 output.mkv should work [21:17:29 CET] <ariyasu> yes that worked :) [21:17:31 CET] <ariyasu> thank you klaxa [21:17:40 CET] <klaxa> :) [21:22:11 CET] <kab0m> furq: it did not work. when i call teh script via "python copytags.py in.flac out.m4a" nothing happens [21:24:03 CET] <pink_mist> did the tags not get copied? [21:24:20 CET] <kab0m> pink_mist: nope [21:45:58 CET] <kab0m> furq: where do you have this script from? Do you if can get any support for this script or have you written this by yourself? Does anyone else has an idea how i can achieve this task? tia [21:46:51 CET] <furq> i wrote it myself [21:47:01 CET] <furq> i checked mutagen and it doesn't look like their tag mapping is as good as i thought [21:49:20 CET] <kab0m> furq: ahh i see. So i guess this mean that the script wont work? Im a total noob in this, so i don't know any script/tool which does exactly what i need except kid3-cli ... but i dont understand the syntax from kid3-cli exactly :( [21:50:26 CET] <kab0m> Does anyone in here know about how to work with kid3-cli? [23:09:11 CET] <GuiToris> hello, how can I check how many frames a specific video clip has? [23:09:46 CET] <durandal_1707> by demuxing and/or decoding all of them [23:10:17 CET] <GuiToris> oh :( [23:12:13 CET] <GuiToris> and what's wrong with yuvj420p? ffmpeg says: deprecated pixel format used, make sure you did set range correctly [23:12:37 CET] <durandal_1707> everything is wrong [23:13:11 CET] <GuiToris> which one should I use? I'm converting still jpeg images [23:14:01 CET] <durandal_1707> just ignore it, if you are not changing color range [23:14:51 CET] <GuiToris> well okay, thanks :) [00:00:00 CET] --- Sun Dec 9 2018

1 0

ffmpeg-devel.log.20181207
by burek 08 Dec '18

08 Dec '18

[06:07:11 CET] <TheAMM> Heh, those AC3 ticket update messages remind me of error spam when decoding broken stuff [10:22:34 CET] <cone-264> ffmpeg 03Zhong Li 07master:34cba6555767: lavc/qsvenc: remove redundant code [10:22:35 CET] <cone-264> ffmpeg 03Zhong Li 07master:33df1afe39fa: lavc/qsvenc: make hevc alignment same as h264 for 1.19+ version [10:22:36 CET] <cone-264> ffmpeg 03Zhong Li 07master:c4a4cfa7628b: lavc/qsvenc: dump log for frame rate [10:22:37 CET] <cone-264> ffmpeg 03Zhong Li 07master:c9f0cff5efd8: lavc/qsvenc: add an option to set h264 pps for every frame [12:37:13 CET] <durandal_1707> so nobody have v210 muxed in MXF? [13:28:18 CET] <cone-264> ffmpeg 03Gyan Doshi 07master:05a61a02d628: doc/general: rearrange libraries in alphabetical order [15:57:03 CET] <JEEB> lol, someone is using something i contribute to in a proprietary player [15:57:07 CET] <JEEB> their jenkins just sent me a CC [15:57:30 CET] <gnafu> JEEB: Are they abiding by the license, or are their a violator? [15:57:57 CET] <JEEB> I couldn't find the product on their site yet [15:58:23 CET] <JEEB> oh [15:58:33 CET] <JEEB> it's their player for Academy Awards screenings [15:58:43 CET] <JEEB> *screeners [16:00:30 CET] <JEEB> https://www.fortiumtech.com/products-solutions/mediaseal/ [16:00:56 CET] <kierank> hahahaha [16:00:58 CET] <kierank> drm product [16:03:34 CET] <JEEB> yup [16:03:37 CET] <JEEB> that made my evening [16:03:43 CET] <JEEB> I was down with flu for a while [16:04:02 CET] <JEEB> a private CI system sending me an e-mail telling me their build failed xD [16:04:41 CET] <rcombs> sounds less like DRM and more like extremely dumb disk encryption [19:45:21 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:0f8383005c97: avcodec/dpx: check version of format header too [19:45:22 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:abb0306bf301: avcodec/dpx: parse image creator [19:45:23 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:9c881cf9a57e: avcodec/dpx: parse input device name [19:45:24 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:f628970e409c: avcodec/dpx: add support for another gray10 variant [19:45:25 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:788e116c2f10: avcodec/dpx: improve decoding of 10 bit gray images [19:45:26 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:61e6226a5a22: avcodec/dpx: add support for special encoding [19:45:27 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:b88ed73be185: avcodec/dpx: fix spotted code style issues [20:07:26 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:3cc5576ee100: avcodec/proresdec2: add new line at end of debug message [20:12:57 CET] <cone-264> ffmpeg 03Paul B Mahol 07master:cc6208cde26a: avformat/mxfdec: set codec_tag for prores codec [22:02:50 CET] <Ingersol> hi. what is purpose of av_gettime_relative()? it was added after av_gettime() so seems it should fit unknown (for me) needs. [22:07:46 CET] <Ingersol> *got it. av_gettime() uses gettimeofday which may be affected by 3rd party forces. [00:00:00 CET] --- Sat Dec 8 2018

1 0

ffmpeg.log.20181207
by burek 08 Dec '18

08 Dec '18

[01:43:16 CET] <Soni> how do I use ffmpeg to mix multiple audio files together? [01:43:22 CET] <Soni> at different positions [01:43:27 CET] <Soni> y'know, like an audacity project [01:43:34 CET] <Soni> can ffmpeg do that? [01:44:00 CET] <Soni> can ffmpeg read stuff from a file? [01:44:20 CET] <Soni> (commands etc) [04:33:38 CET] <raytiley> is there a way to take something like `-vf scale=w=640:h=360:force_original_aspect_ratio=decrease` and make it round to even pixels? [04:47:23 CET] <furq> raytiley: scale=640:-2 or scale=-2:360 [04:47:48 CET] <furq> or if you need to handle any AR, scale="if(gt(a\,16/9)\,640\,-2):if(gt(a\,16/9)\,-2\,360)" [04:48:42 CET] <raytiley> furq: thanks, just stumbled on that [11:20:23 CET] <dv_> hmm what parts of a video codec benefit most from hw acceleration on embedded devices? [11:20:34 CET] <Mavrik> All of them? :) [11:20:43 CET] <Mavrik> It's usually a full decoding / encoding block on the SoC [11:20:51 CET] <dv_> I mean, assume that you have a hw codec whose driver is fully open source, so you can access all of the hardware's facilities. [11:21:10 CET] <Mavrik> You're assuming that the pipeline is programmable. [11:21:11 CET] <dv_> isnt the hardware usually made up of units that do individual steps, like DCT, motion compensation, deblocking etc.? [11:21:18 CET] <Mavrik> But it's usually a hardwired block that does a full decode [11:21:20 CET] <Mavrik> And a full encode [11:21:47 CET] <Mavrik> Which is configurable to some extent but not really programmable [11:22:05 CET] <JEEB> at some point intel noted that they might open up APIs for specific parts of H.264, but then that person was mysteriously vanished, never to be seen again [11:28:07 CET] <dv_> hmm [11:28:15 CET] <dv_> Mavrik: ever seen anything about Hantro hw codecs? [13:52:20 CET] <alonex> . [13:52:56 CET] <alonex> Hello, any one know right syntax of QSV h264 encoding? [14:18:43 CET] <alonex> i want to use ffmpeg for screen capturing with QSV instead h264 lossless, but i can't. With bandicam qsv working. [14:20:30 CET] <JEEB> pretty sure QSV doesn't support lossless H.264? [14:22:10 CET] <alonex> hello jeeb, i guess qsv does not supporting lossless with h264, but with bandicam i use 100% of quolity with 5 frames per second rate [14:22:24 CET] <alonex> and h264 with QSV - it's ok. [14:22:33 CET] <alonex> CPU almost iddle [14:22:44 CET] <alonex> but i want my lovely ffmpeg for that [14:57:28 CET] <alonex> . [14:57:39 CET] <alonex> Jeeb u here? [14:57:56 CET] <alonex> hello jeeb, i guess qsv does not supporting lossless with h264, but with bandicam i use 100% of quolity with 5 frames per second rate [14:58:02 CET] <alonex> and h264 with QSV - it's ok. [14:58:05 CET] <alonex> but i want my lovely ffmpeg for that [18:15:10 CET] <brimestone> Hello everyone.. I'm getting an issue when transcoding an XDCAM proxy into mp4 where the resulting mp4 has a few seconds of repeated last frame. [21:14:19 CET] <petecouture> Has anyone run into an issue of coverting a webcam -> wowza recorded flv1 video to mp4 using FFMPEG. And during the process theres a timing issue where 3 minutes are stretched out into 40 minutes? Like suddenly the video and audio slow down to 1/16 speed. [21:14:37 CET] <petecouture> Its a 5 minute recording and the first minute and last minute play fine. [21:18:55 CET] <djk> I had an ffmpeg setup to stream to facebook at 720 unfortunately lost the that working configuration. The stream I have setup now Facebook only show 480 as the highest option. any suggestions? https://pastebin.com/dQfRbSVC [21:19:21 CET] <JEEB> note: you don't need 44.1kHz unless Facebook requires that [21:19:44 CET] <JEEB> post the full log for that? [21:19:47 CET] <JEEB> on a pastebin [21:19:49 CET] <JEEB> and link [21:19:59 CET] <JEEB> because that looks as if you're setting 1280x720 and then not scaling [21:20:05 CET] <JEEB> so if something's doing something, it's facebook :P [21:20:36 CET] <djk> JEEB: so it looks right to you? [21:21:25 CET] <djk> and the 44.1 I recall is Facebook requirement have to double check. It is silent anyway [21:21:33 CET] <JEEB> can't say anything from just the command, but if you look at the log and it clearly encodes 720p then you can only ask facebook [21:21:46 CET] <JEEB> yea facebook might be requiring it due to a stupid wording in the FLV spec :P [21:21:49 CET] <JEEB> the spec literally says [21:21:57 CET] <djk> https://www.facebook.com/BRPCwebcam [21:22:06 CET] <JEEB> "for AAC we want 44.1kHz as the value written in the FLV header" [21:22:12 CET] <JEEB> and then the next sentence [21:22:22 CET] <JEEB> "we will promptly ignore this value in players and trust the AAC stream instead" [21:22:29 CET] <djk> that is the live stream. [21:22:37 CET] <JEEB> cue over 9000 vendors which limited AAC to 44.1kHz :P [21:22:40 CET] <JEEB> (and stereo) [21:22:55 CET] <atomnuker> JEEB: so the aac samplerate? [21:22:55 CET] <JEEB> even though the spec says "we promptly ignore these two values and trust the AAC stream" [21:23:06 CET] <JEEB> atomnuker: yes, AAC configuration [21:23:11 CET] <atomnuker> lul [21:23:12 CET] <JEEB> as yoü'd expect [21:23:31 CET] <JEEB> only the FLV header contains the pre-defined values (stereo, 44.1kHz) [21:23:52 CET] <JEEB> but yea, that's why all those video services with FLV ended up with 44.1kHz AAC [21:23:59 CET] <JEEB> because they skipped the second sentence [21:24:12 CET] <djk> :-/ [21:26:11 CET] <djk> JEEB: what else do you need or does what I have setup look right? [21:26:58 CET] <JEEB> I have no idea what *facebook* needs [21:27:00 CET] <djk> JEEB: and it is just Facebook messing with the stream making 480 the best view [21:27:11 CET] <JEEB> if you see in your log that the encoded resolution is 720p [21:27:11 CET] <petecouture> Ive figured out the answer to my question. If anyone else runs into this legacy issue with Flash -> Wowza. The video was recorded with a variable framerate which players handle differently from flv to mp4 when you re-encode it. Most players keep th einitial framerate and then adjust the playback speed to match any variences. The key to fixing it is to define a flat rate or -r 30 when youre reencoding from flv to mp4 [21:27:27 CET] <JEEB> atomnuker: https://wwwimages2.adobe.com/content/dam/acom/en/devnet/flv/video_file_form… [21:27:30 CET] <JEEB> page 71 [21:27:37 CET] <JEEB> fourth paragraph [21:28:15 CET] <djk> JEEB: https://www.facebook.com/help/publisher/162540111070395?helpref=page_content [21:28:16 CET] <JEEB> petecouture: if the timestamps make sense it should be remuxable [21:41:03 CET] <djk> JEEB: I updated the with the nohup.out log https://pastebin.com/dQfRbSVC [22:43:53 CET] <alonex> i want to use ffmpeg for screen capturing with QSV instead h264 lossless, but i can't. With bandicam qsv working. [22:44:06 CET] <alonex> Hello, any one know right syntax of QSV h264 encoding? [00:00:00 CET] --- Sat Dec 8 2018

1 0

ffmpeg-devel.log.20181206
by burek 07 Dec '18

07 Dec '18

[12:53:51 CET] <JEEB> beep boop for https://patchwork.ffmpeg.org/patch/11258/ , https://patchwork.ffmpeg.org/patch/11257/ [15:15:41 CET] <durandal_1707> h264 code is awfull [15:16:19 CET] <alexpacio_> @durandal_1707: what do you mean? [16:06:26 CET] <JEEB> durandal_1707: yea it has a lot of older stuff and it might not be simple to "get" into [16:09:04 CET] <jamrial> plus years and years of "fix this one broken sample made by this one long discontinued cheap cctv camera" lines :p [16:09:32 CET] <JEEB> yup [16:09:50 CET] <jamrial> or workarounds so broken streams saved through udp can still output something somewhat recognizable as a video frame [16:10:18 CET] <JEEB> thankfully a lot of the UDP I find is generally OK and the most I want is for a decoder to recover after the input un-fucks itself [16:10:19 CET] <nevcairiel> good error resilience is really a good thing to have [16:10:39 CET] <jamrial> sure, but doesn't change the fact it can make the code look awful [16:10:45 CET] <JEEB> yup [17:57:40 CET] <durandal_1707> people wrote shitty dav parsers clones, and now ffmpeg need to fight back! [20:15:05 CET] <cone-766> ffmpeg 03Carl Eugen Hoyos 07master:b44a571dc4af: lavc/proresdec.h: Include a missing header [22:04:47 CET] <atomnuker> I don't like opusenc_psy now, I want to start from scratch [22:10:00 CET] <durandal_1707> anyone have link to v210 in mxf? [22:15:26 CET] <jamrial> kierank may [22:16:15 CET] <durandal_1707> sample [23:46:58 CET] <durandal_1707> i got seizures when dealing with Carl [00:00:00 CET] --- Fri Dec 7 2018

1 0

ffmpeg.log.20181206
by burek 07 Dec '18

07 Dec '18

[01:04:50 CET] <amflir> `ffmpeg -i foo.mov bar.mov` changes the size of file from 200mb to 15mb for me. I can hardly see any quality difference by eyes. I am just deciding to which version to store? [01:05:06 CET] <devp> Hello. I've got a MPD file with segmets of video and audio, there's some way to take just the audio segments? [01:05:17 CET] <amflir> actually what happens? why it compresses the file so much? and what I lose? [01:35:50 CET] <amflir> why concating videos with ffmpeg concat filter (and -c copy) produces a larger output than cat a.mts b.mts>c.mts ? [01:38:04 CET] <DHE> amflir: if you don't specify codecs and options thereof, ffmpeg uses defaults. they might not be smart defaults, but they're functional... [01:41:09 CET] <amflir> DHE, when I don't reencode (use -c copy) what exactly changes that makes the output larger? [01:42:51 CET] <DHE> mpegts has a LOT of muxing overhead. 10% is somewhat normal, and some types are optional or just repeated frequently. [01:43:28 CET] <DHE> eg: ffmpeg's mpegts defaults are designed to be friendly to streaming/broadcast uses. you can knock a bit off the file size if you change that. [04:40:15 CET] <fraktor> I'm trying to extract some audio from another file using ffmpeg to do some processing with it, but it's quite long, so WAV files don't quite work properly. What's another uncompressed format that I can use? [04:42:52 CET] <furq> flac? [04:43:04 CET] <furq> if "don't work properly" means you're running out of disk space [04:44:58 CET] <fraktor> WAV headers can only support files up to 4gb, so it's more that the file format is problematic. [04:45:07 CET] <furq> oh right yeah [04:45:09 CET] <furq> well flac should work fine [04:45:33 CET] <furq> or mka if you specifically need uncompressed [04:45:35 CET] <fraktor> I wanted something uncompressed, so that conversion would be fast. I found out how to use raw pcm data, which audacity can read, so I went with that. [09:33:23 CET] <th3_v0ice> How can I get how long in seconds a video is, AVFormatContext->duration / AV_TIME_BASE should do it, right? [12:15:54 CET] <X-Kent> hi, I am using ffplay to play hls. source m3u8 has 10 segments available. ffplay starts playing from the last 3 segments. Can I somehow force it to start from the first segments ? (as in case of last 3 it stalls if any of the segments doesn't arrive in time) [12:16:04 CET] <X-Kent> it's a live stream [12:29:17 CET] <X-Kent> found it, it's live_start_index [14:16:47 CET] <DHE> X-Kent: sounds like the playlist is badly formatted. it should indicate it's an EVENT/VOD type, or have an EXT-X-ENDLIST tag indicating it's complete. otherwise it's assumed to be a live/updating event and played near the end [14:17:57 CET] <X-Kent> DHE, it is a live playlist, I found the solution, it's "-live_start_index", it wasn't in the manpage so took time to figure out [14:21:01 CET] <DHE> oh, you did say that. oops... [14:21:23 CET] <DHE> yeah default behaviour is to start at close to live as realistically possible. 3 segments are for buffering purposes. [15:15:56 CET] <alexpacio_> how can I tell ffmpeg to use libva (VA-API acceleration) by default? [16:22:31 CET] <devp> Hello. I need split an audio from video source, I did download it from a streaming. The audio is encoded into the video source with a range of bytes indicated through of a .mpd file. How i could get just the audio ? [16:23:37 CET] <pink_mist> ffmpeg -i file.ext audio.mp3 [16:24:51 CET] <DHE> that transcodes the audio to mp3, which may or may not be what you want [16:26:31 CET] <pink_mist> yeah, alternatively you could figure out what the actual encoding is and use -c:a copy foo.<appropriate audio extension> but I didn't want to haveto write that much, but now you made me do it anyway =( [16:26:53 CET] <devp> exactly.. The source has the audio encoded just in a little bytes and that bytes are indicated in a mpd file [16:27:11 CET] <pink_mist> "in a little bytes" is not an audio encoding [16:32:07 CET] <devp> And i could split a video/audio from encrypted source ? [17:16:28 CET] <DHE> devp: ffmepg has some encryption support. if the key is obtainable, maybe. [17:41:25 CET] <FurretUber> Is there a way to record a specific window with FFmpeg? Using Xubuntu 18.04, Xorg. I know there is a way to set resolution and coordinate, but I wanted to set by window title, if possible [17:42:28 CET] <DHE> doesn't look like it. see: ffmpeg -h demuxer=x11grab [17:42:49 CET] <DHE> none of these options seems appropriate, other than helping you line it up with a capture border [17:55:28 CET] <BtbN> There isn't really an API to do that either [17:55:47 CET] <BtbN> OBS does it by just tracking position and size and then cropping a full screen capture [17:56:07 CET] <BtbN> Or via (ab)using XComposite extensions to grab a specific windows texture [17:57:51 CET] <Alina-malina> can someone give me the command of how correctly to download the desktop and sound? [18:00:24 CET] <Zgrokl> is it possible to pipe screen shot >> ffmpeg in realtime with multithread [18:00:31 CET] <Zgrokl> actually it's working monothread [18:00:39 CET] <Zgrokl> maybe because ffmpeg wait on the screenshot app [18:01:16 CET] <Zgrokl> but i want to habe speed = 1x [18:02:15 CET] <BtbN> That sounds like worlds most inefficient screen recording. [18:02:35 CET] <Alina-malina> well i need to record the sound from firefox browser [18:02:37 CET] <Alina-malina> maybe it has some pipe? [18:02:52 CET] <BtbN> Use OBS [18:42:42 CET] <Alina-malina> ye tried it, but its not convenient, i wish i could record the same with ffmpeg :-/ but thanks for tip bro, btw mate this is the quality i made with OBS https://youtu.be/4FlfLesMzLc BtbN [18:43:32 CET] <BtbN> so? Encoder settings are identical for obs and ffmpeg. Since obs uses ffmpeg as encoder. [18:44:01 CET] <Alina-malina> what about CPU usage? [18:44:31 CET] <Alina-malina> i thought that ffmpeg is less eating resources, anyways thanks for tip, this is great program, i think i can stream with it as wel? [18:44:34 CET] <Alina-malina> well* [18:45:05 CET] <Alina-malina> with ffmpeg i could add slides fade in/ fade out and some other effects, so much get used to it already [18:47:46 CET] <BtbN> That's usually what you use OBS for, not ffmpeg. Since doing that is insanely annoying with just ffmpeg cli [18:50:39 CET] <Alina-malina> well i already got used to it hehe [20:25:15 CET] <halvard> Hi I tried to convert a mkv to mp4, but got the error "Could not write to header for output file". I used just the simple command ffmpeg -i <input> -codec copy <output> [20:25:35 CET] <JEEB> the actual error is somewhere before that [20:25:44 CET] <JEEB> post full terminal log on pastebin or similar [20:25:46 CET] <JEEB> link here [20:26:10 CET] <halvard> Codec frame size is not set, probably. [20:26:14 CET] <halvard> Will do :) [20:27:40 CET] <halvard> https://pastebin.com/S5K6FeZx [20:29:17 CET] <DHE> line 31 pretty much says it all [20:29:44 CET] <JEEB> except of course everyone prefers -strict experimental instead of -2 :P [20:29:52 CET] <DHE> either add the -strict option just before the output filename, or transcode the audio to another code [20:29:54 CET] <DHE> *codec [20:30:12 CET] <JEEB> also I'm not sure if the opus muxing of 3.4.x is up to snuff [20:32:34 CET] <JEEB> seemms like there were some improvements semi-recently [20:32:43 CET] <JEEB> this probably the biggest one http://git.videolan.org/?p=ffmpeg.git;a=commit;h=7890181d7e76a74fdb7a50785f… [20:34:02 CET] <halvard> At least one output file must be specified [20:34:42 CET] <kepstin> maybe that message ffmpeg prints for the "support is experimental" should have an "or upgrade to a newer ffmpeg version" added to the end of it :) [20:41:38 CET] <pink_mist> kepstin: how can an old version of ffmpeg know that a newer one will not have the support be experiemental still? [20:41:48 CET] <pink_mist> *experimental [20:43:04 CET] <halvard> Artificial Intelligence [21:24:52 CET] <Seethe> Hi, i'm trying to add two audio streams (both work independently, encoded in reaper as pcm) into an mp4 container. ffmpeg seems to be setting the codec improperly to an ascii square. here's the command: https://pastebin.com/MhJwLHLe (ffmpeg.pastebin is down?) [21:57:39 CET] <BtbN> I'm pretty sure mp4 can't contain wav [21:59:44 CET] <JEEB> there's raw audio defined for it, but it's not often supported by other stuff [00:00:00 CET] --- Fri Dec 7 2018

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