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November 2016

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ffmpeg-devel.log.20161119
by burek 20 Nov '16

20 Nov '16

[00:12:12 CET] <rcombs> I know of a few random devices that support RS422; no idea what they do with it, though [00:12:33 CET] <rcombs> the Ki Pro supports it [00:12:44 CET] <rcombs> I guess it's used for shuttle control; probably common in tape decks? [00:14:02 CET] <Chloe> yes, I think it's related to tape hardware [00:14:50 CET] <llogan> maybe dericed has some [00:15:42 CET] <rcombs> I've got access to a few Ki Pros (and maybe some tape decks); why do you ask? [00:17:45 CET] <Chloe> I'm nearing the end of writing my avdevice for the macOS CoreMediaIO API, but I don't support much, due to me having only one piece of hardware (an old consumer level camcorder from 2005-ish), so I was wondering if anyone had something and would be interested in working with me to test the Interfacing/Control part of the CMIO API [00:18:15 CET] <TD-Linux> Chloe, I have a CRT with RS422, but a custom protocol that I reverse engineered [00:18:33 CET] <TD-Linux> very similar to the sony VHS control one though [00:50:27 CET] <Chloe> rcombs: the other issue is that you'd have to be integrated into the macOS environment to get any use from this. I'm not sure how the multimedia industry sways in terms of operating system usage with tape stuffs (I assume windows or maybe linux) [00:50:50 CET] <rcombs> I dunno what anyone still uses tape for [00:51:04 CET] <rcombs> nor the shuttle interfaces on HDD/SSD recorders [00:51:45 CET] <rcombs> I guess you could use those in a live-broadcast environment for replay or other video playback? [00:51:54 CET] <TD-Linux> vaporwave production [00:51:58 CET] <rcombs> but you'd integrate that with your switcher, not with a regular machine [00:54:32 CET] <Chloe> So leave out the interface stuff entirely? [01:01:29 CET] <beastd> lol @ "Map edited" makes us look like an OSM-like project ;-) [01:03:20 CET] <llogan> it was a profound edit [01:31:23 CET] <beastd> about passing encoder options to the turingcodec lib: i guess they do it that way for having a "cheap" C library interface. internally they have an Encoder class that takes a variable map (some boost data type), and they have a command line app which means they have the code for creating that variables map from the process arguments [01:32:52 CET] <beastd> anyway the code constructing the command line in ffmpeg encoder init looked strange, if it isn't wrong it is at least awkward i would say. but maybe i am missing things. [01:37:04 CET] <Chloe> I just love how the new apple developer website is absolutely, completely useless. The header files are 10000x better... [02:16:41 CET] <cone-306> ffmpeg 03Steven Liu 07master:4696f7639b22: avformat/flvdec: add debug message to list keyframes index metadata [02:18:51 CET] <cone-306> ffmpeg 03James Almer 07master:2ab50647ff65: avformat/utils: add av_stream_add_side_data() [02:22:11 CET] <cone-306> ffmpeg 03James Almer 07master:77f033eb98d0: avformat/mov: use av_stream_add_side_data() for displaymatrix side data [05:24:13 CET] <philipl> 5~ [15:53:34 CET] <Kuroe> Anyone know what '3G Text' subtitles are? [15:56:30 CET] <nevcairiel> https://en.wikipedia.org/wiki/MPEG-4_Part_17 probably [15:56:46 CET] <nevcairiel> sometimes called tx3g or 3gpp timed text [15:59:12 CET] <Kuroe> ah thanks, what's the CODEC_ID--I can't seem to find it? (surely ffmpeg supports it) [15:59:42 CET] <Compn> uhmmmm [15:59:57 CET] <nevcairiel> mov_text i think? [15:59:58 CET] <Compn> we have timed text samples, dunno about 3gpp timed text tho [16:00:45 CET] <Compn> CODEC_ID_MOV_TEXT [16:01:34 CET] <cone-757> ffmpeg 03James Almer 07master:eb3a59c903bd: avformat/mov: reuse existing err variable [16:03:45 CET] <Compn> http://samples.ffmpeg.org/mov/subtitles-embedded/ [16:05:32 CET] <Compn> Kuroe : but if you have sample files which do not work, let us know. :) [16:06:55 CET] <Kuroe> I'm trying to match up CoreMedia FourCCs to FFmpeg AVCodecIDs [16:07:23 CET] <Kuroe> I can do a lot of them, but some of them I don't know. [16:11:15 CET] <nevcairiel> if its an apple t hing its most definitely mov_text [16:22:13 CET] <cone-757> ffmpeg 03James Almer 07master:2343f23e4d7e: avformat/matroska: use av_stream_add_side_data() for stereo3d side data [17:21:47 CET] <durandal_1707> is having video to captions/codes filter OK? [17:22:25 CET] <durandal_1707> eia-608 [17:27:22 CET] <Kuroe> what's the 'normal' jpeg codecid? [17:27:47 CET] <Compn> normal as in mjpeg or single image ? [17:28:05 CET] <Compn> durandal_1707 : you mean ocr? or ? [17:28:18 CET] <Kuroe> single image [17:28:39 CET] <Kuroe> although this is probably mpjeg actually [17:29:01 CET] <Compn> ffplay says a .jpg file is mjpeg [17:29:21 CET] <Compn> the only other fourcc is 'ijpg' really, but there are a lot more [17:37:11 CET] <nevcairiel> jpg images usually just live in their own jpg files, not a typical fourcc for them [17:37:25 CET] <durandal_1707> Compn: no, it's coded as pixels on special rows [17:40:25 CET] <Kuroe> Well, posted it to the ML, can see what people think [17:43:27 CET] <nevcairiel> the real question is, what exactly is that good for [17:52:33 CET] <Kuroe> It would make working with coremedia + ffmpeg tons easier, although I guess it could be bundled in every application instead of added to ffmpeg. Also getting a AVCodecID from a CMDescriptionRef with the least amount of pain [17:52:59 CET] <Compn> durandal_1707 : oh you mean a filter to convert text to the analog video signal pixel rows ? [18:12:11 CET] <nevcairiel> Chloe: some of thoise don't seem particularly helpful, particularly muxed and metadata tags, we don't handle those as codecs (timed id3 is something else) [18:13:18 CET] <nevcairiel> and I would think t hose are practically the same as the mp4/isom mappings for other codecs? [18:23:20 CET] <durandal_1707> Compn: other way around [18:23:35 CET] <philipl> BtbN, nevcairiel: IT'S HAPPENING. [18:24:03 CET] <philipl> Latest nvidia drivers (375.20 on linux) have 12bit hevc decode and p010 (probably p016) output [18:24:16 CET] <BtbN> via cuvid? Or did they update vdpau? [18:24:28 CET] <philipl> cuvid [18:24:51 CET] <philipl> Undocumented without an sdk update, of course. [18:24:56 CET] <philipl> I just decided to try it out and it worked. [18:26:13 CET] <wm4> philipl: wow [18:26:37 CET] <wm4> so I guess nvidia don't care about vdpau anymore? [18:27:12 CET] <nevcairiel> i've been looking at their dxva2 devices if they added a new p016 device or such, but nothing yet [18:27:27 CET] <nevcairiel> there is a unnamed guid with p010 output, i was wondering if thats maybe the 12 bit device [18:27:32 CET] <nevcairiel> but without 12 bit output its meh [18:28:29 CET] <nevcairiel> and by try it you mean you just requested a output format "2" ? :p [18:28:58 CET] <philipl> output format [18:29:02 CET] <philipl> "1" [18:29:09 CET] <philipl> 0 is nv12. previously 1 didn't work [18:29:10 CET] <nevcairiel> oh right nv12 was 0 [18:29:32 CET] <philipl> and I tried a 12bit sample and it worked where that didn't before [18:33:46 CET] <BtbN> Yeah, I think we should get that set merged to get cuvid useable. Not looking good for vdpau. [18:34:53 CET] <philipl> quite so [18:35:41 CET] <philipl> do we have a p016 format? [18:35:58 CET] <philipl> I don't it or p012 [18:36:18 CET] <philipl> Unclear if 12bit is being dithered down to 10bit or whether they are populating the extra bits [18:37:06 CET] <philipl> running strings on the cuvid library only shows P016 so I'd like to think "1" is P016 [18:53:01 CET] <Chloe[m]> nevcairel: the muxed tags are the most useful [18:55:04 CET] <nevcairiel> philipl: p012 is not something i ever heard before [18:55:11 CET] <Chloe[m]> The CoreMediaIO API can output muxed data [18:55:26 CET] <nevcairiel> Chloe[m]: how is that, ffmepg cant handle it as that [18:58:54 CET] <Chloe[m]> Also it would be silly to have all the mappings except for one or two. A complete list would help to keep things together [18:59:25 CET] <nevcairiel> but it gives an illusion of being something ffmpeg understands [18:59:28 CET] <nevcairiel> which it is not [19:00:52 CET] <philipl> nevcairiel: true enough. It goes straight to p016 [19:01:38 CET] <nevcairiel> philipl: not that it really matters, the memory layout of those is the same since its a msb-packed format [19:01:46 CET] <nevcairiel> just more or less zeros [19:02:02 CET] <Chloe> nevcairiel: ffmpeg can't demux/decode DV? [19:02:23 CET] <nevcairiel> Chloe[m]: it can decode dv and it can demux dv, but it doesn't use codec ids for demuxing [19:02:37 CET] <nevcairiel> it needs an already demuxed track for the codecid to make sense [19:03:20 CET] <nevcairiel> same for the mpeg2ts codec id, thats just some hack for very special use-cases, ffmpeg cant do anything with it [19:05:57 CET] <Chloe> ok, so which would be useful? audio, video and subtitle/text/cc? [19:06:32 CET] <nevcairiel> yeah those things that are actually codecs [19:06:38 CET] <nevcairiel> mostly all of them except mux and meta [19:07:13 CET] <nevcairiel> although it feeld odd to split various incantations of subtitles into 3 [19:07:17 CET] <nevcairiel> but if apple has it that way [19:07:35 CET] <Chloe> yeah that's how it is, I thought it'd be better to leave it as is [19:07:42 CET] <Chloe> What would be the best way to handle muxed and metadata? [19:08:19 CET] <Chloe> AV_CODEC_ID_PROBE for muxed? [19:08:32 CET] <nevcairiel> just not bunch it in w ith codec ids [19:08:38 CET] <nevcairiel> there are none that make sense [19:08:51 CET] <nevcairiel> you need to use entirely different APIs to process muxed streams then codec streams [19:09:15 CET] <nevcairiel> avformat vs avcodec [19:09:19 CET] <Chloe> yes, ok [19:10:35 CET] <nevcairiel> and metadata is just an entirely different concept .. its not a stream of data, but like a dictionary [19:10:46 CET] <nevcairiel> avformat usually just extracts that from the container [21:06:22 CET] <philipl> I assume no one objects if I throw out support for 10bit -> nv12 dithering in cuvid. It'll be hard to handle the logic if we can't assume it goes to p010 [21:17:00 CET] <wm4> shouldn't the API user be able to ask for a pixfmt? [21:22:36 CET] <nevcairiel> on a decoder level that seems a bit silly, just let it output the native one and thats that [21:24:34 CET] <wm4> true [21:25:59 CET] <nevcairiel> people have been harassing me for cuvid hevc10 support and i told them no because 8-bit rounded output is dumb, guess when this hits an actual sdk i will need to implement that <.< [21:31:06 CET] <wm4> but why [21:31:14 CET] <wm4> it makes no sense since dxva2 works? [21:33:35 CET] <nevcairiel> thats what i keep telling them [21:34:01 CET] <nevcairiel> the only advantage cuvid has it that you get deinterlacing basically for free [21:34:09 CET] <nevcairiel> without needing a renderer to access that [21:35:11 CET] <wm4> you can deint with the d3d video processor [21:35:18 CET] <nevcairiel> i know [21:35:22 CET] <nevcairiel> would just need to build that [21:35:26 CET] <wm4> (although I haven't found out how to select the damn deint alg) [21:35:27 CET] <nevcairiel> with cuvid its like one flag [21:51:45 CET] <philipl> ff_get_format seems to do the wrong thing in the case where cuvid is being used with copy-back. [21:52:10 CET] <philipl> It doesn't try and choose the right pix fmt - it just picks the last entry in the list. [21:52:28 CET] <philipl> I can obviously work around it in cuvid.c easily enough but it seems dumb. [21:55:47 CET] <nevcairiel> you need to have two lists and only call it with the one that has the correct format at the end [21:55:52 CET] <nevcairiel> thats what hevc does, f.ex. [22:00:05 CET] <philipl> so check the input pix_fmt first and then call with the right list? [22:00:26 CET] <nevcairiel> pretty much [22:52:51 CET] <philipl> BtbN: https://github.com/philipl/FFmpeg/commit/5d678acc8983ca311dfe0c8eb3630cb3a6… [23:14:58 CET] <nevcairiel> adding p016 support wouldnt be terribly hard [23:15:10 CET] <nevcairiel> if p010 gets converted to yuv420p10le the 2 extra bits are just shifted out [23:18:45 CET] <nevcairiel> but being able to use them interchangabliy otherwise is part of their design, thats why they use a msb layout [23:21:13 CET] <philipl> small favours. [23:21:24 CET] <philipl> But this code isn't right yet :-( [23:21:34 CET] <philipl> It assumes the pix_fmt will be set when the decoder init is called. [23:21:39 CET] <philipl> That's true for mp4 containers; not for mkv [23:21:49 CET] <philipl> I'm not sure how I break the cycle here. [23:24:02 CET] <wm4> philipl: are you testing this with mpv? because the libavformat demuxer decodes some video to determine a pixfmt, while mpv's mkv demuxer doesn't [23:24:25 CET] <nevcairiel> a decoder like that really shouldnt rely on the pixfmt being set, it should set it itself [23:26:36 CET] <philipl> wm4: yes, I am. [23:26:53 CET] <philipl> nevcairiel: Yeah. I'm not sure how to bootstrap it though. [23:27:26 CET] <nevcairiel> just wait until the cuvid parser calls the config callback and set it then? :) [23:27:42 CET] <nevcairiel> by that time you have bistream info [23:27:59 CET] <philipl> I'm concerned that's too late. [23:28:12 CET] <philipl> but I can try [23:28:51 CET] <wm4> I don't know what callback cuvid has there, but why would it be too late? [23:29:25 CET] <nevcairiel> thats how h264/hevc etc work as well, they only figure out the pixfmt once it was determiend from the stream [23:29:40 CET] <philipl> Ok. I'll push on that angle. [00:00:00 CET] --- Sun Nov 20 2016

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ffmpeg.log.20161119
by burek 20 Nov '16

20 Nov '16

[02:16:30 CET] <yunhuali> hi, I can draw frame number with drawtext filter with text=%{n}, is there any way to draw fps? [02:16:44 CET] <yunhuali> I want to get Frame per second. [02:17:50 CET] <yunhuali> I want draw text like every 10 minutes, display 30second, then disappear, then show up again, 10 minutes later, [02:29:04 CET] <llamapixel> yunhuali: One idea would be to chop up the video and apply to the segments then put it all back together. [02:29:34 CET] <llamapixel> Idea 2 would be to make a black video with just the fps at the correct timeline locations and overlaying with additive. [02:30:07 CET] <yunhuali> llamapixel, Thanks, I wish have better solution. [02:30:31 CET] <llamapixel> What exactly do you want to do? that might encourage other ideas. [02:30:35 CET] <furq> yunhuali: https://ffmpeg.org/ffmpeg-filters.html#Timeline-editing [02:31:02 CET] <furq> i assume this is a vfr video and you can't just get it with ffprobe beforehand [02:31:07 CET] <furq> s/it/the fps/ [02:31:49 CET] <llamapixel> oh nice enable between feature [02:32:38 CET] <yunhuali> thank you furq, I'm actually reading that, and was trying to do it. My idea is use drawtext=enable='between(t,10,60)') [02:33:32 CET] <furq> http://ffmpeg.org/ffmpeg-utils.html#Expression-Evaluation [02:33:35 CET] <furq> you probably want to read that as well [02:33:38 CET] <yunhuali> Can I change it to between(mod(t, 600), t, t+30)' [02:34:13 CET] <furq> you probably want between(mod(t, 600), 0, 30) [02:35:05 CET] <llamapixel> clever [02:38:28 CET] <yunhuali> Thanks, Furq, I just tested with ffplay, it works [02:38:42 CET] <yunhuali> (mod(t, 60), 0, 10) [02:38:55 CET] <yunhuali> I changed it to smaller number, so I can verify it [02:41:02 CET] <yunhuali> Thanks!! have a nice weekend! [06:26:33 CET] <mantas322> Hi guys [06:26:35 CET] <mantas322> anyone here? [12:51:54 CET] <somaReverse> hello [12:51:58 CET] <somaReverse> I'm new to ffmpeg [12:52:14 CET] <somaReverse> I'm very confused by these options [12:52:23 CET] <somaReverse> https://ptpb.pw/iC51 [12:52:47 CET] <somaReverse> what do they mean? How can I make it require less cpu? [12:55:32 CET] <DHE> https://www.ffmpeg.org/ffmpeg-all.html while it's a lot to digest, when you need to search for what options mean this is the definitive guide [12:59:29 CET] <somaReverse> ok, is there a pre defined option set that is suit for low cost screencast? [13:00:28 CET] <DHE> you're using vp8, so find the libvpx section of the documentation and see what options it offers [13:02:13 CET] <DHE> I see options named "-quality" and "-speed" that look promising [13:08:07 CET] <somaReverse> yeah, I'm trying. thanks [13:38:16 CET] <relaxed> somaReverse: also, https://trac.ffmpeg.org/wiki [14:38:57 CET] <hoodedfigure> Hi, i am trying to get started with ffmpeg but it seems that the code examples at https://ffmpeg.org/doxygen/trunk/examples.html are outdated. Is that possible? [14:43:34 CET] <JEEB> those should still build. [14:43:52 CET] <JEEB> if they don't in the latest HEAD that's a problem [14:44:07 CET] <JEEB> (the examples are under docs/examples or so in the git repo) [14:44:19 CET] <hoodedfigure> I can't find function open_codec_context() anywhere in the rest of the code, actually. [14:44:42 CET] <hoodedfigure> (in v3.2) [14:45:49 CET] <BtbN> that's a function defines by that specific example itself [14:46:10 CET] <JEEB> yea [14:46:45 CET] <hoodedfigure> oh, sometimes I fee stupid [14:46:49 CET] <hoodedfigure> thanks [15:06:03 CET] <hoodedfigure> I basically want to be able to decode single specific frames throughout the input video. Can you give me a hint how to do that? [15:28:26 CET] <hoodedfigure> I'm getting the following two errors: /usr/bin/ld: video.o: undefined reference to symbol 'av_image_alloc@@LIBAVUTIL_55' /usr/lib/libavutil.so.55: error adding symbols: DSO missing from command line [15:28:35 CET] <hoodedfigure> Is my library compiled the wrong way? [16:51:25 CET] <DHE> hoodedfigure: compiling and linking against mismatched versions of the headers and libraries. check if you have ffmpeg installed properly [17:51:10 CET] <Alina-malina> Hello all, how can i add a file limit i have this <<ffmpeg -i "input" out.mp3>> but the file is getting too big, i need to split it to chunks [17:53:43 CET] <SchrodingersScat> does -segment_time 300 -f segment help? [17:54:58 CET] <furq> -segment_time 300 -f segment out%03d.mp3 [17:55:48 CET] <Alina-malina> oh cool let me try this please [18:00:53 CET] <Alina-malina> furq, oh this is so cool thanks!!! [18:11:30 CET] <SchrodingersScat> i was there too ;( [18:16:55 CET] <kerio> how's "too big" a thing [18:27:01 CET] <yalopov> hello! [18:27:22 CET] <durandal_1707> hello [18:28:40 CET] <yalopov> i want to learn about theory behind video codecs and containers, does anyone knows if there's a book about it? [18:32:43 CET] <furq> yalopov: http://www.dspguide.com/ [18:37:01 CET] <yalopov> impressive, that's what i was looking for [18:37:03 CET] <yalopov> thanks [18:43:30 CET] <jj15> So following from what I was trying to do the other day Ive done it :) Ive created interactive live streams where users get shoutouts for shating the live stream. Check it out (ffmpeg and PHP script to pull date from FB) https://www.facebook.com/ZestyBingo/videos/1343693932321218/ [18:43:44 CET] <jj15> s/shating/sharing [18:45:53 CET] <DHE> I liked shating better [18:54:32 CET] <jj15> ha ha [19:00:24 CET] <yalopov> it's pretty cool! [19:00:38 CET] <yalopov> i bit lagged i think [19:30:04 CET] <jj15> I think the lag is my connection. Do you mean with the sound? [19:30:11 CET] <jj15> Or the updating? [19:31:07 CET] <jj15> With the updating my script queries Facebook every 5 seconds otherwise I get errors from their API so the image is updated every 5 seconds. I want to now try overlay a movie of snow falling. [19:36:50 CET] <yalopov> updating [19:36:52 CET] <yalopov> oh, i see. [20:13:27 CET] <wouter> does it make sense to try a combination of -i "concat:foo.ts|bar.ts|baz.ts" and "-c:v copy" in the same command line, or is that totally not going to work [20:13:39 CET] <wouter> (assuming the three .ts files have the same settings etc) [20:44:43 CET] <DHE> wouter: if they literally are identical in codec and codec settings, then it should be safe [00:00:00 CET] --- Sun Nov 20 2016

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ffmpeg-devel.log.20161118
by burek 19 Nov '16

19 Nov '16

[00:06:56 CET] <wm4> jamrial_: what are those about? gapless stuff? [00:08:44 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:1af7ddecda33: apng: use side data to pass extradata to muxer [00:08:45 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:e622d7723bcf: fate: add apng encoding/muxing test [00:08:46 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:e92f585bd90e: configure: make sure LTO does not optimize out the test functions [00:08:47 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:346fa70bb88a: doc: fix spelling errors [00:08:48 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:c90d521f161b: interplayacm: check for too large b [00:08:49 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:266cf258cc93: interplayacm: validate number of channels [00:08:50 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:6e5ccabbe846: interplayacm: increase bitstream buffer size by AV_INPUT_BUFFER_PADDING_SIZE [00:08:51 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:d95568f9a2d3: mov: immediately return from mov_fix_index without old index entries [00:08:52 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:3e33685892f9: apngdec: use side data to pass extradata to the decoder [00:08:53 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:35db873534a8: fate: add streamcopy test for apng [00:08:54 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:dc2942bbc889: avformat: close parser if codec changed [00:08:55 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:35cb0c47bc89: ppc: pixblockdsp: do unaligned block accesses correctly again [00:08:56 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:08b1fd6afb3d: diracdec: use correct buffer for slice_params_buf realloc [00:08:57 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:b9a24cee3ba0: diracdec: clear slice_params_num_buf on allocation failure [00:08:58 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:31cebfe789ff: diracdec: check return code of get_buffer_with_edge [00:08:59 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:c72ac9ffd017: lzf: update pointer p after realloc [00:09:00 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:c11fd9de7653: mxfdec: fix NULL pointer dereference [00:09:01 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:1e4979f780a8: mpegaudio_parser: don't return AVERROR_PATCHWELCOME [00:09:02 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:fa24e3780b77: matroskadec: fix NULL pointer dereference in webm_dash_manifest_read_header [00:09:03 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:9dee25fbc796: mpegts: prevent division by zero [00:09:04 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:72a2d6ff56ce: escape124: reject codebook size 0 [00:09:05 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:1ed4b527321f: proresdec_lgpl: explicitly check coff[3] against slice_data_size [00:09:06 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:581cce0cca80: dvbsubdec: fix division by zero in compute_default_clut [00:09:07 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:41359d381a2b: icodec: fix leaking pkt on error [00:09:08 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:1615d83dcf6e: icodec: correctly check avio_read return value [00:09:09 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:d8affeea82e5: smvjpegdec: make sure cur_frame is not negative [00:09:10 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:039a3e6db892: pnmdec: make sure v is capped by maxval [00:09:11 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:b526958ca4a1: filmstripdec: correctly check image dimensions [00:09:13 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:c8f5154fc1ca: softfloat: handle -INT_MAX correctly [00:09:14 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:a2c7840a6b69: mlz: limit next_code to data buffer size [00:09:14 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:598016b85f29: dds: limit 4 bpp handling to AV_PIX_FMT_PAL8 [00:09:15 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:2b863d4e9b10: libschroedingerdec: don't produce empty frames [00:09:17 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:cbc9d46066b6: libschroedingerdec: fix leaking of framewithpts [00:09:18 CET] <cone-780> ffmpeg 03Andreas Cadhalpun 07release/3.2:c7d38efbc26b: exr: fix out-of-bounds read [00:09:19 CET] <cone-780> ffmpeg 03Martin Vignali 07release/3.2:08f26d99b52f: libavcodec/exr : fix channel size calculation for uint32 channel [00:09:19 CET] <cone-780> ffmpeg 03Stefano Sabatini 07release/3.2:31c9c7ad826e: ffprobe: fix crash in case -of is specified with an empty string [00:13:58 CET] <Chloe> jeez. that's a lot of commits [00:21:11 CET] <BBB> michaelni: tnx for the fix [00:23:23 CET] <atomnuker> michaelni: is cehoyos' commit still needed after those 2 commits? [00:36:05 CET] <michaelni> it would affect cases for which we do not have a explicit pixel format (we cannot represent these without extra lsbs, i think theres no way to encode these atm either), future ffmpeg versions could support encoding these though and old ffmpeg could decode it by adding lsbs. [00:36:40 CET] <nevcairiel> but dont you agree that adding random bits into the format is anything but not lossless? [00:37:20 CET] <michaelni> i belive the changed codepath is unused except such future (not explicitly supported formats) [00:38:07 CET] <Chloe> if it's unused then there's no reason why it should be there? [00:38:10 CET] <michaelni> nevcairiel, i dont know, adding 0 lsbs is not really lossless either [00:38:35 CET] <Chloe> michaelni: why not? [00:38:37 CET] <nevcairiel> well this only happens if you encode a format you cant decode to, doesnt it [00:38:51 CET] <nevcairiel> obvious solution: make sure its implemented at both ends [00:38:53 CET] <nevcairiel> no need for hackery [00:40:43 CET] <nevcairiel> half-assed "support" for gray10 is what got us here, why not simply enforce adding both parts in the future, the decoder patch doesn't look particularly complicated now [00:41:56 CET] <michaelni> the ffv1 spec does not restrict things to 8,10,12,16 bits, our decoder should support as much as it can even if our encoder doesnt IMO [00:43:10 CET] <nevcairiel> but scaling it up to some other bitdepth and adding bits that didnt exist from the start is just not lossless, is all [00:43:34 CET] <nevcairiel> the decoder shouldnt decode a format it doesnt know to output unmodified, then [00:44:28 CET] <nevcairiel> to some degree its probably nice to do that, but on a more stricter level it may cause issues with people assuming its properly lossless/bitexact when it isnt [00:46:26 CET] <nevcairiel> but alas thanks for fixing the 10-bit case, i'll leave it to you to decide what to do about those "fallback" decoding modes [00:58:21 CET] <jamrial_> i need to revert a patch, which will break a fate test, to apply another, which will make the test pass again [00:58:26 CET] <jamrial_> should i instead revert and apply the new code all in one commit to avoid breaking fate for the sake of bisects? [00:58:47 CET] <jamrial_> such a commit will look ugly and hard to read i guess [00:59:46 CET] <jamrial_> maybe i could disable the test within the revert commit instead [00:59:46 CET] <nevcairiel> you could put a fate change into the commit, but thats probably still ugly [01:03:53 CET] <jamrial_> nevcairiel: afaik without either of the patches the test would fail, not just output wrong data [01:04:03 CET] <nevcairiel> i see [01:04:21 CET] <nevcairiel> well disabling it in the revert is probably the best choice, squashing the two will likely be a very ugly diff [01:04:52 CET] <nevcairiel> you wouldnt be reverting if the two approaches were at all similar [01:05:29 CET] <jamrial_> yeah, i'll do that [03:38:57 CET] <kinetik> jamrial_: ping [03:39:19 CET] <jamrial_> kinetik: pong [03:40:33 CET] <kinetik> jamrial_: hi! just looking at fixing the comment you made on the FLAC MP4 muxer patch. I cribbed the experimental flag check from the VP9 code (which has the same issue, it'll mux into mov without the strict flag) [03:41:47 CET] <kinetik> jamrial_: are you saying that i should remove it from ff_codec_movaudio_tags or just make the strict flag check exclude non-MODE_MP4 entirely? [03:42:10 CET] <jamrial_> i tried removing the line from ff_codec_movaudio_tags but it broke demuxing [03:44:58 CET] <jamrial_> maybe just return EINVAL with an error message saying flac is not supported if mode is not mp4 [03:45:20 CET] <jamrial_> i'd prefer the movenc maintainer to chime in, but he hasn't replied when i CCd him [03:46:08 CET] <jamrial_> trying to mux to m4a (ipod muxer) correctly fails with your patch as is, at least [03:46:42 CET] <kinetik> removing it doesn't seem to stop it being muxed into mov either [03:48:41 CET] <jamrial_> it makes the muxer use "MS style audio codec tag" instead of your code [03:49:06 CET] <jamrial_> which probably just creates a broken file [03:49:06 CET] <kinetik> ah, i see [04:00:11 CET] <kinetik> jamrial_: thanks, i'll post an updated patch shortly [10:17:29 CET] <cone-106> ffmpeg 03Vittorio Giovara 07master:390b95b88b2b: fate: Add a mixed NAL coding sample [10:17:30 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:9e1ddc08208d: Merge commit '390b95b88b2b896b63f257f69e434dfc0111e076' [10:22:55 CET] <cone-106> ffmpeg 03Vittorio Giovara 07master:0b1bd1b2057d: lavd: Drop unneeded av_init_packet()s [10:22:56 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:d7d6f9c78272: Merge commit '0b1bd1b2057d41fd0ccba7317911c484a50f9207' [10:44:32 CET] <cone-106> ffmpeg 03Matthieu Bouron 07master:61cb9fac4749: mov: fix stream extradata_size allocation [10:44:33 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:8a91452e83e7: Merge commit '61cb9fac47498a38dfe7623f66aa1f3696e9158c' [10:44:46 CET] <cone-106> ffmpeg 03Martin Storsjö 07master:d0b1e6049b06: libopenh264dec: Fix cleanup if the init failed early [10:44:47 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:6fb07c7d85c0: Merge commit 'd0b1e6049b06eeeeca146ece4d2f199c5dba1565' [10:45:16 CET] <cone-106> ffmpeg 03Martin Storsjö 07master:36b380dcd52e: libopenh264dec: Simplify the init thanks to FF_CODEC_CAP_INIT_CLEANUP being set [10:45:17 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:0bd76401d166: Merge commit '36b380dcd52ef47d7ba0559ed51192c88d82a9bd' [10:46:37 CET] <cone-106> ffmpeg 03Martin Storsjö 07master:2d097c16b833: libopenh264enc: Return a more sensible error code in some init failure paths [10:46:38 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:7e9474ca47fd: Merge commit '2d097c16b833c532ac974a7f1fd05c0a1f3b7675' [10:47:37 CET] <cone-106> ffmpeg 03Martin Storsjö 07master:7a76371437f9: libopenh264enc: Simplify init by setting FF_CODEC_CAP_INIT_CLEANUP [10:47:38 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:07502e473f0c: Merge commit '7a76371437f9562c3414f985523f883489e3936a' [10:52:20 CET] <cone-106> ffmpeg 03Anton Khirnov 07master:d60c2d521693: mp3dec: read the initial/trailing padding from the LAME tag [10:52:21 CET] <cone-106> ffmpeg 03Anton Khirnov 07master:46278ec90ac5: mp3enc: write trailing padding [10:52:22 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:f4c3aa7ade13: Merge commit 'd60c2d5216930ef98c7d4d6837d6229b37e0dcb3' [10:52:23 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:5e78126bbdd5: Merge commit '46278ec90ac5ad1dab5e85991f176afe49003fee' [11:37:33 CET] <cone-106> ffmpeg 03Anton Khirnov 07master:8d07e941b04d: FATE: add a test of H.264 SEI recovery in an intra refresh stream [11:37:34 CET] <cone-106> ffmpeg 03Hendrik Leppkes 07master:127cc6dd3d0b: Merge commit '8d07e941b04d63fc4443dd986e3dc7b69cdcca43' [13:47:15 CET] <BBB> michaelni: can you add support for specifying color_range, colorspace, color_trc and color_matrix to ffv1? [13:47:38 CET] <BBB> michaelni: or alternatively, can you add support for this being specified in the container (I believe webm/mp4 support that)? [13:48:02 CET] <BBB> michaelni: maybe chroma location also [13:49:04 CET] <wm4> is color_matrix = primaries? [13:53:46 CET] <BtbN> it should be, yes. At least from what I remember from VAAPI [14:06:56 CET] <nevcairiel> matrix is color space [14:07:33 CET] <nevcairiel> (ie. the most important of them all) [14:08:23 CET] <nevcairiel> to avoid confusion just use the terms the api uses, range, primaries, trc, space [14:10:02 CET] <nevcairiel> BBB: matroskaenc (and as such webm) can already write thsoe elements, but the spec for those is not formally finalized yet, so it requires a -strict flag [14:10:30 CET] <BBB> with matrix, I meant primaries, yes [14:10:43 CET] <BBB> ok, strict might be useful [14:10:54 CET] <nevcairiel> -f_strict -1 [14:10:55 CET] <nevcairiel> or so [14:10:59 CET] <BBB> cool, ok [14:11:03 CET] <BBB> tnx [14:26:45 CET] <wm4> this should probably include some HDR stuff as well then [14:44:47 CET] <nevcairiel> which mkv also has [14:45:18 CET] <nevcairiel> its included in the specs here now https://www.matroska.org/technical/specs/index.html .. i dont know if that means its official now [14:45:38 CET] <nevcairiel> the development of mkv has splintered onto a second mailing list which i dont follow [14:45:40 CET] <nevcairiel> so :p [16:58:58 CET] <cone-306> ffmpeg 03James Almer 07master:16c429166ddf: Revert "apngdec: use side data to pass extradata to the decoder" [16:58:59 CET] <cone-306> ffmpeg 03James Almer 07master:50e0e96f3f8e: avformat/apngenc: use the stream parameters extradata if available [16:59:00 CET] <cone-306> ffmpeg 03James Almer 07release/3.2:cf655d164334: Revert "apngdec: use side data to pass extradata to the decoder" [16:59:01 CET] <cone-306> ffmpeg 03James Almer 07release/3.2:f97bee9ad565: avformat/apngenc: use the stream parameters extradata if available [17:24:47 CET] <Chloe> Why is there yet another HEVC library... [17:29:57 CET] <Chloe> Oh hmm, I was under the impression there were quite a few libraries, looks like there's only a few. (was probably thinking of h264) [17:33:29 CET] <RiCON> there's only two non-hardware for h264, same as hevc [17:37:15 CET] <Chloe> RiCON: I see that now too. [17:45:44 CET] <atomnuker> reynaldo: what's happening with ffsever? [17:53:16 CET] <Chloe> atomnuker: I believe it was being continued in a separate repo, so dropping it in ffmpeg was going ahead as planned [17:54:16 CET] <atomnuker> I know that, but reynaldo said it wouldn't take him long but just a couple of days a couple of weeks ago [18:19:00 CET] <cone-306> ffmpeg 03Michael Niedermayer 07master:d7a3bb208867: avcodec/ffv1dec: Support gray 10/12/16 explicitly avoid shifts [18:19:01 CET] <cone-306> ffmpeg 03Michael Niedermayer 07master:d1d18de6ade3: avcodec/ffv1dec: Set packed_at_lsb for 16bit YUV [19:14:53 CET] <ubitux> nevcairiel: "Our H264 decoder drops 3 frames from the beginning of the stream" why? [19:15:14 CET] <nevcairiel> if you find out, you can tell me [19:15:30 CET] <ubitux> i thought you knew, ok :) [19:34:54 CET] <jamrial> that libturing library needs boost [19:37:41 CET] <RiCON> it comes with boost in the repo too, in case you don't want or have it installed [19:37:50 CET] <jamrial> it also has the weirdest method to send parameters to an encoder i've seen so far [19:38:21 CET] <atomnuker> it reconstructs a command line [19:38:33 CET] <Chloe> RiCON: yeah, I think I cried at that [19:38:34 CET] <atomnuker> complete with the executable name in the front [19:40:18 CET] <llogan> boost is the best [19:40:50 CET] <nevcairiel> boost isnt too bad if you're a c++ project, but linking C++ libs to C projects is always a bit terrible =p [19:41:14 CET] <nevcairiel> but as long as these shenangians are hidden through pkg_config or something, not hardcoded in configure.. [19:43:11 CET] <llogan> plus you get to install a million billion files for boost [19:43:22 CET] <BtbN> Did they turn a command-line encoder into a library, or what the hell is that interface? [19:46:02 CET] <nevcairiel> probably [20:12:31 CET] <wm4> the overflow checks seem to be severely broken in that patch too [20:15:10 CET] <jamrial> kinetik: i'm trying files created with your patch on Firefox 51 beta 1, and i get no video if the file also has a flac stream [20:15:19 CET] <jamrial> the audio plays, though [20:35:07 CET] <ubitux> BBB: libavcodec/x86/vp9itxfm.asm:2151: error: instruction expected after label [20:35:15 CET] <ubitux> (yasm 0.8.0) [20:35:23 CET] <BBB> add HAVE_EXTERNAL_AVX2 around the function [20:35:26 CET] <nevcairiel> update that shit =p [20:35:27 CET] <ubitux> (doesn't matter much but i have a fate instance with that as requested) [20:35:28 CET] <BBB> someone else reported the same bug yesterday [20:35:38 CET] <ubitux> probably Carl reading FATE [20:35:40 CET] <ubitux> ? [20:36:03 CET] <ubitux> i might be wrong but i think he's the one who asked me to add that instance [20:36:12 CET] <ubitux> maybe because a stable distro was using this release [20:36:32 CET] <ubitux> i must say i don't care much, i was just raising the problem [20:36:52 CET] <nevcairiel> its an easy fix either way, but at some point we might as well bump the requirement [20:37:11 CET] <BBB> can you guys add the right %if HAVE_EXTERNAL_AVX2 around the given functions? [20:37:14 CET] <BBB> makes my life easy [20:39:16 CET] <jamrial> we could just bump the minimum yasm version to 1.1.0. that's what debian oldstable uses [20:39:22 CET] <cone-306> ffmpeg 03Clément BSsch 07master:b9c26aed4e3c: ffmpeg: remove dead code out of the video filter loop [20:39:23 CET] <cone-306> ffmpeg 03Clément BSsch 07master:825f6fbfac6c: ffmpeg: factor out sending frame to filters [20:39:36 CET] <jamrial> it will not address this, though, since avx2 was added in 1.2.0 [20:40:21 CET] <jamrial> but it would let us remove guards for other sets [20:51:40 CET] <BBB> up to you guys [20:51:58 CET] <BBB> I dont mind the %if HAVE_EXTERNAL_AVX2 if people think its useful [20:52:06 CET] <BBB> but if youd rather remove it thats obviously fine with me also [20:53:35 CET] <jamrial> BBB: i'll add the guards for now [20:53:53 CET] <RiCON> BBB: it was wm4 [20:54:00 CET] <RiCON> ffmpeg-git was breaking mpv's travis build [20:54:19 CET] <nevcairiel> if we ever get avx512 we'll need to switch to recommending nasm anyway, at which time we might as well require a more recent version [20:54:33 CET] <RiCON> it was "fixed" by adding --disable-asm [20:56:06 CET] <BBB> RiCON: omg [20:56:25 CET] <BBB> RiCON: Im not sure that was the best solution [20:56:27 CET] <RiCON> travis is still on ubuntu 12.04, so yasm 1.1.0 [20:56:49 CET] <BBB> I hope that thing isnt running cpu-intesive unit tests using ffmpeg [20:57:12 CET] <RiCON> it's not, it's just building ffmpeg to build mpv, not running any test [20:57:55 CET] <BBB> ok then [20:57:57 CET] <RiCON> fwiw, libass also disables asm on its own [20:57:58 CET] <BBB> sorry about the breakage [20:59:17 CET] <llogan> nasm is dead, let's use yasm. yasm is dead, let's use nasm. [21:00:33 CET] <BBB> thats pretty much the cycle of life, yes [21:01:13 CET] <BBB> the sad thing is that for software to succeed, you pretty much have to give up your life and any other amibitions for it [21:01:29 CET] <BBB> the yasm author got distracted; yasm dead [21:03:25 CET] <cone-306> ffmpeg 03James Almer 07master:2de1c79b61b6: x86/vp9itxfm: add missing AVX2 guards [21:07:52 CET] <BBB> jamrial: ty! [21:09:01 CET] <wm4> mpv's travis build uses ffmpeg only to the extent of testing header files and linking against it, so --disable-asm is fine [21:09:13 CET] <wm4> trying to use it with a shit-old nasm is only a side-effect [22:45:04 CET] <cone-306> ffmpeg 03Andreas Cadhalpun 07master:7289aa2d71a8: options_table: limit codec parameters to sane values [23:46:04 CET] <Chloe> Anyone got any AVC or RS422 hardware? [23:46:34 CET] <Chloe> AVC being 'AV/C Digital Interface Command Set General Specification Version 4.1' [00:00:00 CET] --- Sat Nov 19 2016

1 0

ffmpeg.log.20161118
by burek 19 Nov '16

19 Nov '16

[00:16:34 CET] <SteenJobs> hey all - having a lot of trouble installing the ffmpeg cookbook - i ran into a yasm issue but fixed it by adding package yasm manually to my chef recipe, but doing the same for libx264 is not working. despite it installing im still getting a not found error [00:31:11 CET] <llogan> what's a cookbook? [00:32:39 CET] <SteenJobs> llogan: its basically ruby code that dictates how to configure the server [00:37:18 CET] <llogan> we don't really provide support for such things here [00:47:09 CET] <Daniel__> llogan, I got it working but it's not pretty :D http://pastebin.com/9uT8D7Ac [01:08:57 CET] <llamapixel> Is there a preferred editor for ffmpeg string construction? something that elevates the process? [01:10:04 CET] <Chloe[m]> SteenJobs: probably missing the devel package [01:10:57 CET] <SteenJobs> Chloe[m]: tried adding it, didnt work. instead of using the ffmpeg cookbook i decided to just take the installation guide and turn it into my own recipe [01:18:45 CET] <llamapixel> cheeky monkeys, http://www.iffmpeg.com | http://www.myffmpeg.com/ [01:32:45 CET] <llogan> it appears they leave it up for the user to find a binary [05:51:39 CET] <hiby007> jj15 you did it? :) [10:43:22 CET] <teamcoltra> Hey guys, I feel like what I want is super simple but I can't figure it out. I want to crop my video to a 1:1 aspect ratio, have it keep whatever is in the dead centre of the video and be done. :P Nothing fancy [10:44:15 CET] <kerio> can the video be vertical [10:44:20 CET] <kerio> the input [10:45:52 CET] <kerio> try -vf crop=min(in_w,in_h):min(in_w,in_h) [11:04:38 CET] <teamcoltra> Thanks kerio that was helpful :) [11:06:58 CET] <kerio> did it work? [11:17:40 CET] <Eiken> anyone got some idea why scale messes up the alpha in a filter_complex with overlay [11:17:55 CET] <Eiken> if i remove the scale the alpha works as expected and is faded togheter with rgb [11:20:03 CET] <Eiken> i tried adding format=yuva420p to the scale but same result [11:21:43 CET] <Eiken> [1:v]fade=in:0:10[v1]; [0:v]setpts=PTS-STARTPTS[v0]; [v1]setpts=PTS-STARTPTS+26/TB[v3];[v0][v3]overlay=eof_action=pass[out1] this works [11:22:08 CET] <Eiken> [0:v]scale=640:360,format=yuva420p[scaled];[1:v]scale=640:360,format=yuva420p[scaledlogo];[scaledlogo]fade=in:0:10[v1]; [scaled]setpts=PTS-STARTPTS[v0]; [v1]setpts=PTS-STARTPTS+26/TB[v3];[v0][v3]overlay=eof_action=pass[out1] this don't [11:24:22 CET] <Eiken> maybe i can put the scale afterwards instead [11:27:37 CET] <Eiken> that worked like a charm [11:27:38 CET] <Eiken> yay [12:14:08 CET] <hay> hi all.. I have two inputs for an output and would like that ffmpeg process stops if any of those two streams fails... I have tried with -shortest output option and it doesn't seem to work (the process doesn't stop), also with -xerror -err_detect explode it still continues... any ideas on how to achieve this? thanks [13:35:56 CET] <DanteD> Any idea if it can be avoided that picture gets a little darker with rgb->yuv420p conversion e.g. when making a video of gameplay? [13:41:07 CET] <Mavrik> Not really, you can convert to yuv444 if you think players will handle that [13:41:40 CET] <DanteD> Mavrik: I wonder if yuv444 can be uploaded to youtube? [13:41:58 CET] <Mavrik> it can, but they'll just use ffmpeg to downconvert it back to yuv420 anyway :) [13:42:42 CET] <DanteD> Ok. I will just live with this then, maybe do some post processing to to try to limit the proble [13:43:10 CET] <DanteD> brighten rgb -> convert to yuv , maybe it will help [13:44:10 CET] <kerio> is it really darker tho [13:47:37 CET] <DanteD> kerio: yes, it is, dark brown -> black [13:48:48 CET] <DanteD> orange -> red [13:48:53 CET] <ZexaronS> I have a bit of a problem with dropping frames from video [13:49:33 CET] <ZexaronS> It's kinda goes slow it wants to recalculate ofcourse, i have various video from 30-6 FPS, heavily variating, im not sure if it's variating inside the stream [13:49:47 CET] <ZexaronS> I've tried -r 1 as suggested [13:51:57 CET] <ZexaronS> But, a video is only 85 MB in side and it took ... i don't know the latest ffmpeg doesn't report time for some reason [13:52:04 CET] <ZexaronS> the video is twice as large as well [13:52:18 CET] <ZexaronS> I tried using -vsync drop but it said invalid argument [13:53:05 CET] <ZexaronS> I'd just like to dump all reference frames, and make a video out of those, there's no audio and all are mp4 AVC files [13:53:32 CET] <ZexaronS> but i never done that [14:51:21 CET] <ZexaronS> most of these files are badly created in terms of container/codec compatability [14:51:40 CET] <ZexaronS> the time is stretched, I can't get rid of the timecodes [14:54:15 CET] <ZexaronS> basically trying to extract all I frames at maximum of 1 or less fps [14:54:38 CET] <ZexaronS> then i'll make a new video off those I frames [15:07:11 CET] <ZexaronS> some progress finally, i have been able to lower size from 85MB to some 30 MB with this command [15:07:54 CET] <ZexaronS> but the timecodes still stuck, it's not really 1 fps, it's like a lot of time between frames, so I have to experiment with setpts right ? [15:10:34 CET] <ZexaronS> the source files are incredibly buggy, bitrate at the start of the video or in some areas is reported as 61 million, and there's a lot of warnings and errors about data being corrupt in streams [15:13:17 CET] <kerio> does the source actually have variable framerate? [15:20:48 CET] <ZexaronS> I do not know for sure, some of the files are properly detected as 720p and 29,9 or 23,9 or 24 FPS in the windows explorer, and media info doesn't have a problem, but majority of the files are also mp4 but were are kinda weirldy muxed or generated [15:21:13 CET] <ZexaronS> It's 250GB total but I have removed many similar ones and probably down to 50 GB now [15:21:35 CET] <ZexaronS> and these roughly 50 GB of some 100 files, I chooses a bit of these and a bit of those [15:21:52 CET] <ZexaronS> but I can't see for sure if they're variable FPS inside [15:22:28 CET] <ZexaronS> Some videos are time-stretched, for example the 6 FPS ones, almost all videos are muxed to 15:19 minutes [15:22:55 CET] <ZexaronS> everything I open, whether it's 29, 23, 26, 25, 20, or 6 fps, it's always 15:19 minutes long [15:26:01 CET] <ZexaronS> it's all over the place, there's different resolutions, 3 groups, and another time group of 59:56 minutes, there's also 28,673 framerate in one of them [15:27:13 CET] <ZexaronS> there are already many 1 fps ones which are small I just saved them in whole no need to fix those [15:28:51 CET] <ZexaronS> snd ofcourse bitrates range from 5 to 30k [15:29:24 CET] <kerio> sounds fun [15:31:47 CET] <ZexaronS> kerio: I'm trying to use CRF to preserve quality and I simply want to extract 1 key frame per 2 seconds diregarding input framerate, resoultion, or anything else and dump those I frames into a new video which plays back these frames at fixed 30 fps, disregarding any timecodes, could you help ? [15:32:15 CET] <kerio> i could [15:32:26 CET] <kerio> but then again, all i do is just read the docs [15:32:29 CET] <kerio> so you could too [15:32:47 CET] <ZexaronS> I am but -vsync drop doesn't work [15:36:13 CET] <ZexaronS> I am at this for a day, i never dealt with something like this, I know how to use ffmpeg I did DVD remux and archiving [15:39:09 CET] <furq> if you just want to dump iframes then use -skip_frame nokey [15:39:39 CET] <furq> if you really need those time constraints then you'll need to mess around with -vf select [15:39:58 CET] <furq> that sounds less good if you're getting decoder errors though [15:41:17 CET] <furq> also it sounds like you want -vsync cfr [15:41:26 CET] <furq> -vsync drop will speed the result up [15:42:05 CET] <ZexaronS> furq, i'm already using select, for %%a in ("*.mp4") do ffmpeg -v 9 -loglevel verbose -i "%%a" -vcodec libx264 -preset fast -x264-params crf=23 -vsync 2 -vf select=eq(pict_type\,I) -an -threads 3 -r 1 -f mp4 "aaa_Transcode_OUT\%%~na.mp4" [15:42:36 CET] <furq> well get rid of -v 9 for starters [15:42:39 CET] <ZexaronS> This made the video a lot smaller just what I wanted, but the timecodes are still that the time is fixed at 15 minutes [15:42:40 CET] <furq> that doesn't do anything [15:43:11 CET] <ZexaronS> I like to use verbose all the time, I tested it a year ago and found it did a bit more [15:43:26 CET] <ZexaronS> showed a bit more [15:43:31 CET] <furq> i didn't say to remove that [15:43:41 CET] <furq> -v and -loglevel are aliases of the same option [15:44:03 CET] <furq> also 9 isn't a valid value [15:44:10 CET] <furq> verbose is 40 [15:44:34 CET] <ZexaronS> this may have since changed ... [15:44:41 CET] <furq> but yeah vsync vfr will preserve the length, it'll just drop frames with duplicate PTS [15:45:08 CET] <ZexaronS> I saw other's using it as well, like in 4chan-webm-maker [15:45:44 CET] <ZexaronS> furq, so drop option isn't valid anymore or another number ? [15:45:54 CET] <furq> -vsync drop [15:46:04 CET] <ZexaronS> says invalid argument [15:46:11 CET] <furq> what ffmpeg version [15:46:15 CET] <ZexaronS> I use x64 zeranoe btw [15:46:19 CET] <ZexaronS> latest [15:46:44 CET] <furq> shrug [15:46:50 CET] <furq> works for me [15:47:02 CET] <furq> pastebin the command and output [15:47:17 CET] <ZexaronS> N-82314-g863ebe6 [15:48:25 CET] <ZexaronS> I'm doing this over remote desktop, the execution is on another PC, since the whole data is offloaded there on spare space on HDD, so i need to go back and forth it may take a bit longer [15:49:01 CET] <ZexaronS> For some reason running bat on my PC won't launch ffmpeg on the target PC [15:53:13 CET] <ZexaronS> furq; http://pastebin.com/tn0qTfhn [15:56:22 CET] <ZexaronS> value 3 doesn't work either [15:57:07 CET] <ZexaronS> with vfr frames get duplicated, i'll try what happens, but does duplication increase size or just makes the same frame replay? [15:59:13 CET] <ZexaronS> dup thing definitely doesn't work, that's the default option before I didn't use vsync command, it makes the file twice as large [16:01:01 CET] <ZexaronS> oh you said yes vfr will preserve lenght ... well the other command also does, and the drop doesn't work [16:01:53 CET] <ZexaronS> what about dumping frames as PNGs ? [16:12:47 CET] <PaulePanter> Hi. Is there a way to install the build FFmeg files to a directory other than `PREFIX`? Thatd be useful for distributions, and test builds. [16:13:12 CET] <PaulePanter> `$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"` is in the Makefile. [16:13:41 CET] <PaulePanter> In GNU Autotools you have `make install DESTDIR="${D}"` where `D` is the destination. [16:18:11 CET] <PaulePanter> Hmm, there is `DESTDIR` in `config.mak`. [16:18:20 CET] <BtbN> DESTDIR works fine for me. [16:20:07 CET] <furq> PaulePanter: make install-progs prefix=/foo [16:21:51 CET] <PaulePanter> BtbN: `MANDIR` is not set correctly for me. [16:22:00 CET] <PaulePanter> furq: Lets see. [16:22:19 CET] <PaulePanter> & but isnt prefix normally something independent from DESTDIR? [16:22:47 CET] <PaulePanter> I probably got it wrong. Let me take a step back, and try again. [16:23:27 CET] <ZexaronS> so furq, this is dead end, if vsync drop doesn't work, I am out of options? [16:26:18 CET] <PaulePanter> BtbN: It dosent for me. [16:26:47 CET] <PaulePanter> BtbN: Sorry, let me try again. [16:31:04 CET] <persina> I'm trying to convert images to a video but the resulting video is just black...any ideas? console output: http://pastebin.com/paKUsYBz [16:56:47 CET] <persina> hmm...still can't figure out why I'm getting a black screen video. [17:24:25 CET] <PaulePanter> BtbN, furq: I called it with `make -j install PREFIX=/usr EPREFIX=/usr BINDIR=/usr/bin SBINDIR=/usr/sbin LIBEXECDIR=/usr/libexec SYSCONFDIR=/etc SHAREDSTATEDIR=/var LOCALSTATEDIR=/var LIBDIR=/usr/lib INCLUDEDIR=/usr/include DATAROOTDIR=/usr/share DATADIR=/usr/share INFODIR=/usr/share/info LOCALEDIR=/usr/share/locale MANDIR=/usr/share/man DOCDIR=/usr/share/doc/ffmpeg DESTDIR=/dev/shm/bee-root/ffmpeg/ffm [17:24:31 CET] <PaulePanter> peg-3.1.4-0/image`. [17:24:57 CET] <PaulePanter> But in contrast to GNU Autotools, FFmpegs build system includes DESTDIR in MANDIR as can bee seen in `config.mak`. [17:25:19 CET] <PaulePanter> So setting for example `MANDIR=/usr/share/man` does not work with FFmpeg. [17:29:12 CET] <kepstin> hmm? autotools should be putting DESTDIR in front of all install paths when set, anything else is a bug (probably in someone Makefile.am, rather than autotools itself) [17:29:46 CET] <kepstin> normally DESTDIR is used for stuff like packaging, where all the files get installed to a temporary tree that's used to make an archive. [17:30:40 CET] <kepstin> so if anything is installed outside of DESTDIR when it's being used, that's a problem [17:36:59 CET] <BtbN> PaulePanter, most of these variables do not exist for ffmpeg. [17:37:03 CET] <BtbN> ffmpeg does not use autotools. [17:41:26 CET] <persina> What does stream_loop -1 do? [17:42:52 CET] <PaulePanter> BtbN: Yes, I knew that. I thought it would follow it in all ways though. [17:43:07 CET] <BtbN> DESTDIR definitely does work perfectly fine though. [17:43:25 CET] <BtbN> And most of those variables are configure options. [17:43:26 CET] <PaulePanter> kepstin: Yes, Autotools puts $DESTDIR$MANDIR in the Makefiles. [17:44:02 CET] <PaulePanter> BtbN: Yes, thats the main problem. I dont see why DESTDIR should be set during configure time. [17:44:15 CET] <kepstin> PaulePanter: ffmpeg's configure is significantly different from autotools [17:44:30 CET] <PaulePanter> kepstin: Yes, I had to learn that the hard way. [17:44:48 CET] <PaulePanter> Thank you for all your help. [17:45:04 CET] <hiby007> hey jj15 any updates? [17:45:23 CET] <kepstin> i'm pretty sure DESTDIR can be set at install time with ffmpeg? the other paths probably can't or shouldn't be. [17:45:46 CET] <kepstin> (that's true of autotools as well, of course) [17:55:12 CET] <BtbN> PaulePanter, it shouldn't. DESTDIR works just fine as it is. [18:01:47 CET] <PaulePanter> kepstin, BtbN: Sorry, you are right indeed. [18:01:59 CET] <PaulePanter> By the way, the version in the topic should probably be updated. [18:03:19 CET] <kepstin> hey, it's not /wrong/ ;) [18:03:41 CET] <PaulePanter> No its not. ;-) [18:03:58 CET] <PaulePanter> By the way, can I specify a different build directory from the source directory? [18:04:13 CET] <BtbN> The CWD is the build directory. [18:05:15 CET] <PaulePanter> BtbN: Good to konw. Thank you. [18:05:26 CET] <babadoc> excuse me [18:05:59 CET] <babadoc> can ffmpeg fix errors in a ts video? [18:06:04 CET] <babadoc> (without encoding it) [18:07:28 CET] <babadoc> whenever i watch a video from my server that is a ts file, it seems to be malformed until i encode it into mkv; [ffmpeg/video] h264: co located POCs unavailable [ffmpeg/video] h264: co located POCs unavailable [18:07:35 CET] <babadoc> those errors are thrown [18:07:48 CET] <babadoc> is this an error i should be concerned about? [18:08:41 CET] <babadoc> here is a link to show what my error is http://i.imgur.com/LVdOKRK.png [18:08:53 CET] <babadoc> it repeats that over and over and over [18:10:14 CET] <babadoc> here it is more detailed: http://i.imgur.com/iijf524.png [18:10:58 CET] <babadoc> Does anyone know if this is fixable via ffmpeg without encoding the ts video into mkv? [18:11:54 CET] <babadoc> I can still watch the video, if not for some skipping along the way [18:12:08 CET] <babadoc> but the red errors are still there. [18:21:49 CET] <babadoc> i also have another question [18:21:54 CET] <babadoc> i have an mkv file on my desktop [18:22:07 CET] <babadoc> i want to encode it to a format that can be played in a browser with audio [18:22:10 CET] <babadoc> such as mp4 [18:22:20 CET] <babadoc> how can i do this in a timely manner? [18:22:29 CET] <kerio> what are the actual codecs? [18:22:32 CET] <kerio> you can use ffprobe to check [18:22:40 CET] <babadoc> lemme check [18:22:56 CET] <kerio> if it's h264/aac you can just ffmpeg -i foo.mkv -c:v copy -c:a copy foo.mp4 [18:23:08 CET] <kerio> i mean, i guess you could just do that and then test [18:23:23 CET] <babadoc> http://pastebin.com/TADv0Xk8 [18:23:34 CET] <babadoc> its a movie [18:23:51 CET] <kerio> video's good, audio's probably good [18:23:59 CET] <babadoc> okay, i will try that now [18:24:42 CET] <babadoc> Nice [18:24:47 CET] <babadoc> thank you so much [18:26:12 CET] <kerio> does the audio work? [18:27:14 CET] <babadoc> no [18:27:17 CET] <babadoc> :( [18:27:22 CET] <babadoc> it plays in my computer [18:27:27 CET] <babadoc> but not on the browser [18:27:44 CET] <babadoc> hmm.. [18:27:49 CET] <kerio> you'll have to reencode the audio, probably in stereo [18:27:58 CET] <babadoc> ffmpeg can do this, yes [18:28:00 CET] <babadoc> or no [18:28:16 CET] <kerio> yes [18:28:39 CET] <babadoc> im looking at this right now https://trac.ffmpeg.org/wiki/Encode/AAC [18:29:25 CET] <kerio> ffmpeg -i foo.mkv -c:v copy -ac 2 -c:a libfdk_aac -b:a 256k foo.mp4 [18:29:39 CET] <furq> he probably doesn't have fdk [18:29:50 CET] <babadoc> i can install it [18:29:58 CET] <furq> you'd need to recompile ffmpeg with it [18:30:04 CET] <babadoc> wait [18:30:06 CET] <furq> just use the builtin aac encoder, there'll be no difference at 256k [18:30:07 CET] <babadoc> maybe i do have it [18:30:07 CET] <kerio> if you don't have it, you can use the builtin aac encoder instead [18:30:18 CET] <kerio> well i said 256k because i like to waste space [18:30:26 CET] <babadoc> lol [18:30:46 CET] <kerio> your original stream was 384k for 6 channels [18:30:59 CET] <kerio> i guess it doesn't make much sense to go over 128k for two then [18:31:15 CET] <babadoc> ffmpeg -i foo.mkv -c:v copy -ac 2 -c:a ffmpeg_aac -b:a 256k foo.mp4 would this be correct? [18:31:19 CET] <furq> just use -vbr 5 [18:31:30 CET] <kerio> i think you mean 0 >:c [18:31:39 CET] <furq> there is no 0 [18:31:48 CET] <kerio> you can't -v:a 0? :( [18:31:50 CET] <furq> oh nvm i'm back on fdk [18:32:12 CET] <furq> iirc vbr with the builtin encoder isn't very good [18:32:36 CET] <furq> with fdk you should just use -vbr 5 (which is highest quality) [18:32:48 CET] <babadoc> i have fdk [18:32:54 CET] <babadoc> and i am runnig the command currently [18:32:58 CET] <babadoc> ffmpeg -i foo.mkv -c:v copy -ac 2 -c:a libfdk_aac -b:a 256k foo.mp4 [18:33:18 CET] <furq> that'll be fine [18:33:25 CET] <babadoc> the speed is slooooooooow [18:33:36 CET] <babadoc> by this rate, itll take like 4 minutes [18:33:38 CET] <babadoc> so long [18:33:39 CET] <babadoc> lol [18:33:44 CET] <furq> but if you're using unnecessarily high bitrates then it's worth knowing that fdk will still run a lowpass filter in cbr mode [18:33:56 CET] <furq> there's some switch to disable it but i forget what it is now [18:34:03 CET] <furq> either way it's disabled with -vbr 5 [18:36:17 CET] <babadoc> audio works [18:36:21 CET] <babadoc> thank you so much :D [18:36:24 CET] <babadoc> both of you [18:37:07 CET] <babadoc> i dont know why it matters that I needed to encode it into 256k to play audio in browsers, but nevertheless it works [18:37:27 CET] <furq> 128k would probably have been fine [18:38:06 CET] <babadoc> but why even encode it into 128k in the first place lol [18:38:14 CET] <furq> your original file was ac3 [18:38:25 CET] <furq> which is somehow still considered exotica in an mp4 [18:38:39 CET] <babadoc> really [18:38:40 CET] <babadoc> wow [18:39:00 CET] <furq> mp4's codec support is generally a nightmare [18:39:23 CET] <furq> i remember browsers used to have issues playing mp3 audio in mp4s [18:39:25 CET] <babadoc> oh god [18:39:35 CET] <babadoc> o.o..... [18:39:38 CET] <ritsuka> no one use mp3 in mp4& [18:39:40 CET] <ritsuka> why would you& [18:39:47 CET] <furq> because it's supported [18:40:03 CET] <babadoc> isnt the mp4 basic audio format m4a [18:40:04 CET] <ritsuka> in the specs yes [18:40:07 CET] <babadoc> or am i wrong [18:40:13 CET] <ritsuka> in the real world not very much [18:40:32 CET] <furq> babadoc: m4a is just another name for mp4 [18:40:44 CET] <furq> usually it uses aac audio [18:40:51 CET] <babadoc> oh i see [18:41:00 CET] <furq> but yeah firefox could play mp3, it could play mp4, but it couldn't play mp3 in mp4 [18:41:17 CET] <furq> i don't see how i was wrong to assume it would work [18:41:20 CET] <furq> (which it does now) [18:41:49 CET] <babadoc> i am so uneducated in this stuff [18:41:54 CET] <babadoc> i have a lot to learn [18:42:20 CET] <babadoc> like, whats the difference between aac audio and mp3? [18:42:31 CET] <furq> aac is a newer codec [18:43:25 CET] <babadoc> alright, well thanks a lt [18:43:26 CET] <babadoc> alot [18:43:31 CET] <babadoc> you have been very helpful :) [18:43:39 CET] <furq> "newer" meaning 1997 vs 1993 [18:43:48 CET] <babadoc> ha [18:44:23 CET] <furq> i'm pretty sure aac predates the first hardware mp3 player [18:45:03 CET] <babadoc> all these formats D: [18:45:09 CET] <babadoc> why cant we just use one or two D: D: D: [18:45:22 CET] <babadoc> why does it always have to be so complicated ;( [18:45:26 CET] <BtbN> because codecs are getting better. [18:45:32 CET] <furq> run `ffmpeg -codecs` [18:45:36 CET] <furq> and revel in the simplicity [18:45:44 CET] <BtbN> mp3 is horribly bad compared to the most recent codecs like opus. [18:46:05 CET] <furq> it's not that bad [18:46:38 CET] <babadoc> in my book, if it can play audio, then its good [18:46:51 CET] <BtbN> juse use wav then [18:47:07 CET] <furq> there aren't really that many audio codecs you need to give a shit about [18:47:18 CET] <BtbN> yeah, opus works for everything nowdays [18:47:30 CET] <furq> gone are the days when musepack had to occupy space in your brain [18:47:52 CET] <ritsuka> if you want browsers compatibility, you don't have many choices: mp4 + h.264 + aac and webm + vp9/vp8 + vorbis/opus [18:48:33 CET] <furq> saying that, i still have some wma kicking about [18:48:42 CET] <kerio> babadoc: because there's audio and audio [18:48:59 CET] <furq> could be worse, it could be atrac3+ [18:48:59 CET] <kerio> especially when you're trading quality with space [18:49:02 CET] <furq> the devil's codec [18:50:42 CET] <kerio> babadoc: oh btw you probably also want -movflags +faststart [18:51:11 CET] <kerio> if you already did the video you can just ffmpeg -i foo1.mp4 -c:v copy -c:a copy -movflags +faststart foo2.mp4 [18:51:46 CET] <kerio> rip [18:52:12 CET] <kerio> BtbN: does opus actually work for music [18:52:24 CET] <furq> opus is designed to work for everything [18:52:29 CET] <kerio> fucking magic ;o [18:52:51 CET] <furq> i keep seeing it peddled that it's specifically only for low bitrates, but that's not true [18:53:12 CET] <furq> i wonder if it has anything to do with that garbage "artifact amplification" listening test site [18:53:24 CET] <kerio> maybe it's because everyone knows opus through mumble? [18:53:37 CET] <furq> are they using opus now [18:53:46 CET] <kerio> mumble has been using opus for quite a while now [18:53:51 CET] <furq> i've not looked in the options since it was speex [18:54:20 CET] <furq> http://soundexpert.org/encoders-128-kbps [18:54:22 CET] <furq> this is the one [18:54:34 CET] <kerio> it defaults to using speex if there's any client that doesn't support opus [18:54:39 CET] <kerio> but you can configure murmur to tell that client to fuck off [18:55:41 CET] <furq> http://soundexpert.org/encoders-320-kbps [18:55:43 CET] <furq> look at this [18:55:56 CET] <furq> original minidisc atrac ranked above opus [18:56:03 CET] <furq> it's like they've got a vendetta or something [18:56:10 CET] <kerio> D; [18:56:29 CET] <kerio> >mfw i only store lossless anyway [18:56:41 CET] <kerio> >mfw 1.8mbps song [18:57:04 CET] <furq> http://soundexpert.org/testing-methodology [18:57:11 CET] <furq> in case you were wondering why all of these results are bullshit [18:57:46 CET] <kerio> i don't get it [18:57:53 CET] <furq> For tested devices with hard to distinguish artifacts this methodology is used in combination with sound artifacts amplification (SARTAMP) special technology aimed to gain artifacts to some controlled level. [18:58:03 CET] <kerio> lmao [18:58:16 CET] <furq> in layman's terms, that acronym should be SHARTAMP [18:58:27 CET] <kerio> sooo [18:58:31 CET] <kerio> opus is worse if you crank the artifacts up? [18:58:47 CET] <furq> something like that [18:58:48 CET] <kerio> i didn't know codecs should optimize for cranked up artifacts [19:02:17 CET] <furq> http://listening-test.coresv.net/s/scores_by_tracks_en.png [19:02:27 CET] <furq> this is the only recent multiformat listening test i know of where the samples haven't been fucked with [19:02:47 CET] <furq> going higher than 128k would probably just result in all 5s for opus and qaac [19:29:59 CET] <persina> If I have images with diff file extensions must I convert them to be the same before using ffmpeg to make a video? [19:33:55 CET] <llogan> anyone recall if the -crf values for x264 are still "offset" for 10-bit? or was that changed later? i'm too damned lazy to look. [19:34:04 CET] <furq> they should be the same [19:35:53 CET] <kohanz> hi all, is this the place to ask a question about an ffmpeg error message? [19:39:38 CET] <llogan> yes [20:02:08 CET] <kohanz> i posted my question to the mailing list [20:02:48 CET] <kohanz> basically I'm trying to join some mp4 files via concat, but get the error message "moov atom not found". However, the mp4 files seem to be fine (playable, FFMpeg does not report errors on them) [20:03:30 CET] <kohanz> if I want to join mp4 files and eventually have transitions in between, is concat even what I should be looking at? [20:08:35 CET] <BtbN> mp4 files are not exactly good for concat [20:08:43 CET] <BtbN> transmux them to ts first [20:10:02 CET] <kohanz> ah, thanks. Also what would be the best practice for joining two files of different resolutions? [20:10:30 CET] <kohanz> i guess transmux to the same resolution, but i guess i'm wondering what the best way is to choose that res [20:11:03 CET] <llogan> use various filters to make them all the same and then concat with concat filter. you can do it all in one command. [20:11:23 CET] <BtbN> you can't transmux and change the resolutions [20:11:42 CET] <BtbN> if you want to concat files without re-encoding, they have to be identical codec and parameter wise [20:12:30 CET] <kohanz> i see, thanks. They will sometimes be of varying size, I don't think I can avoid that. So it sounds like I have to re-encode? [20:13:17 CET] <kohanz> if i went down the reencode path, would you still recommend ts as the intermediate format? [20:14:02 CET] <kohanz> so my workflow would be 1) reencode each video to have the same format + parameters 2) join via concat [20:16:52 CET] <BtbN> if you re encode it should not matter, but you can't use the concat demuxer for that [20:17:21 CET] <kerio> you can use the concat filter tho, can't you [20:17:55 CET] <BtbN> that allways works, yes. [20:18:04 CET] <BtbN> but can be a bit clunky to setup with a lot of files [20:18:52 CET] <BtbN> you basically need a script to generate your commandline if it's more than a handfull of files [20:20:27 CET] <BtbN> It might be easier to just transcode all the various files to a common format in mpegts, and then just concat those [21:18:24 CET] <hay> I have two inputs for an output and would like that ffmpeg process stops if any of those two streams fails... I have tried with -shortest output option and it doesn't seem to work (the process doesn't stop), also with -xerror -err_detect explode it still continues... any ideas on how to achieve this? thanks [21:19:41 CET] <kam187> hi guys [21:20:53 CET] <kam187> how can i make a 3 second 1x1 pixel video with no sound. I want to use it as a data uri video/mp4 [21:22:12 CET] <kam187> it will never be displayed so fps can be set to something silly like 1 [21:26:12 CET] <llogan> i was going to suggest color source filter, but apparently it won't allow 1x1 [21:26:31 CET] <furq> it'll probably need to be 2x2 if you want browsers to actually display it [21:26:41 CET] <llogan> ffmpeg -y -f lavfi -i color=s=2x2:r=1:d=3 output [21:26:45 CET] <kam187> basically i need a video that will be as small as possible and in fact almost any format will do [21:26:46 CET] <kepstin> yeah, since you'll need yuv420p [21:26:48 CET] <kam187> but 3 second slong [21:26:51 CET] <furq> ^ [21:27:37 CET] <kepstin> i don't think you'll get any improvements from smaller than 16x16, which gives me a 1599byte file. [21:27:42 CET] <kam187> 1604 bytes [21:28:00 CET] <furq> i got 1510 with r=1/3 [21:28:29 CET] <kepstin> i get 1503 with ffmpeg -filter_complex color=c=black:s=16x16:r=1/3 -crf 64 -t 3 blank.mp4 :) [21:28:45 CET] <kepstin> nopretty silly video [21:28:46 CET] <furq> you don't need -t 3 [21:28:49 CET] <kepstin> 1 frame long [21:29:01 CET] <furq> well you do but you can pass d=3 to color [21:29:13 CET] <kam187> yep 1503 is the smallest so far [21:29:16 CET] <kepstin> i'd expect the result to be identical, but yeah. [21:29:27 CET] <furq> it's one character fewer [21:29:30 CET] <furq> we all know how important that is [21:29:42 CET] <kepstin> no idea if the browser will actually consider a 1 frame video to have 3s duration, of course. [21:30:02 CET] <furq> kam187: 2x2 will give the exact same filesize as 16x16 fwiw [21:30:16 CET] <furq> in case the display size actually matters [21:30:40 CET] <kam187> i'll base64 encoding it into the page, so the smaller the better, the video will never be displayed [21:30:58 CET] <furq> or maybe not [21:31:00 CET] <furq> it's two bytes bigger [21:31:27 CET] <furq> oh hey [21:31:32 CET] <furq> -preset ultrafast saves ~100b [21:31:38 CET] <furq> s/b/B/ [21:31:44 CET] <hay> I find it hard to understand that no better solution than this presented here: http://video.stackexchange.com/questions/10346/restart-encoding-with-ffmpeg…... :/ [21:32:19 CET] <kam187> 1456 :) [21:32:34 CET] <kam187> err 1406 i mean [21:32:59 CET] <furq> actually isn't there a flag to remove the x264 encoder settings [21:33:06 CET] <furq> that's probably taking up half that size [21:34:34 CET] <kam187> hmm yeah [21:36:14 CET] <furq> ha [21:36:20 CET] <furq> 497-byte webm [21:36:48 CET] <furq> that's with default settings and -crf 63, you can probably do better [21:37:02 CET] <kam187> 497 bytes? [21:37:11 CET] <furq> yeah [21:37:18 CET] <kam187> whats your command line? [21:37:41 CET] <kam187> oh webm format [21:37:54 CET] <furq> ffmpeg -f lavfi -i color=s=16x16:r=1/3:d=3 -crf 63 -c:v libvpx-vp9 tiny.webm [21:38:45 CET] <furq> x264 writes about 500 bytes of encoder settings into the bitstream and there's no flag to remove it afaik [21:38:57 CET] <kam187> hmm i get an error [21:39:06 CET] <kam187> [libvpx-vp9 @ 0x14bf7a0] Failed to initialize encoder: Invalid parameter [21:39:06 CET] <kam187> [libvpx-vp9 @ 0x14bf7a0] Additional information: g_timebase.num out of range [1..cfg->g_timebase.den] [21:39:18 CET] <furq> weird [21:40:05 CET] <llogan> old libvpx? [21:40:34 CET] <furq> kam187: http://vpaste.net/vfIu1 [21:40:49 CET] <kam187> thanks [21:40:52 CET] <kam187> lets see if it works ok [21:41:27 CET] <furq> there's a whole lot of \0s in there [21:41:35 CET] <furq> i wonder if this could go smaller [21:42:58 CET] <furq> https://github.com/mathiasbynens/small/issues/10#issuecomment-95160907 [21:43:00 CET] <furq> apparently it can [21:45:45 CET] <kam187> hmm nope ios doesn't like it [21:45:56 CET] <furq> iOS doesn't support webm [21:47:37 CET] <furq> there's an ancient patch with a bitstream filter that removes x264 settings but it never got merged [21:48:58 CET] <kam187> ahh [21:49:07 CET] <kam187> the 1406 mp4 works ok [21:51:20 CET] <furq> kam187: http://vpaste.net/7D6zU [21:51:22 CET] <furq> 768 bytes [21:51:42 CET] <kam187> oh wow, how? [21:51:50 CET] <furq> http://forum.doom9.org/archive/index.php/t-152419.html [21:52:02 CET] <furq> there's an ancient build with that patch merged [21:52:06 CET] <kam187> ahh [21:52:09 CET] <kam187> lets try it [21:53:48 CET] <kam187> works :) [21:54:00 CET] <TD-Linux> that libvpx error actually looks like a fffmpeg error [21:54:14 CET] <kam187> i'm using a stupid trick to figure out when the user pulls up the control panel in ios [21:54:35 CET] <kam187> there's no page events, but any running media gets sent a 'pause' [21:55:30 CET] <kam187> so on touch i start a small video and if it pasues before getting to the end i know they went to the control panel, i wait for the play event and know they came back. [21:55:32 CET] <llogan> TD-Linux: i am unable to duplicate (assuming the same command mentioned previously) [21:59:43 CET] <ZexaronS> hello [22:00:36 CET] <ZexaronS> I have error i can't output frames to BMP, says "I/O" error cannot find file and it points to the a file that uses the same name as the batch file with an added .bmp [22:00:46 CET] <ZexaronS> cant' find file exec.bat.bmp [22:01:12 CET] <kam187> so next issue, i have a h264 stream i'm saving in a container but there's no keyframes [22:01:29 CET] <kam187> so seeking makes a mess on the screen [22:01:33 CET] <kam187> what's the best thing to do? [22:01:41 CET] <kam187> i guess re-encode is a last resort [22:04:02 CET] <ZexaronS> http://pastebin.com/27Qbjejw [22:04:21 CET] <ZexaronS> im trying to extract all I frames [22:05:05 CET] <ZexaronS> I was trying to do it directly but the vsync drop command doesn't work any nobody had any other ideas, that solution would have worked and be faster from before [22:05:52 CET] <ZexaronS> now I have to get out all I frames, preferrably I could set some command to make it "every 2 seconds" [22:06:12 CET] <ZexaronS> i don't need all I frames if there's more than one in 2 second [22:08:42 CET] <ZexaronS> outputtin non-I frames would take longer I guess, and they will probably have most qualit [22:13:08 CET] <ZexaronS> llogan have you found anything? [22:13:16 CET] <ZexaronS> this is broken, mp4 doesn't work [22:14:03 CET] <ZexaronS> I will have to stop before I lose my mind today this is ridicolous nothing is working what the hell is going on, why is vsync drop not working ? [22:16:00 CET] <ZexaronS> I wanted to convert some of these files I needed space, I ended up picking the best ones and deleting the space, 200 GBs cleaned out, this data was sitting there for months, I got some space I needed, this 50 GB can sit another year before this nonsense gets fixed [22:19:57 CET] <ZexaronS> I found this to be the best lifestyle, dig in for 2 days if the sucker doesn't work, throw it away for a month or 2, problem solved [22:21:26 CET] <ZexaronS> not gonna lose my nerves for this seriously, how can they release a program with a major feature not working ... that's like Ferrari putting a car out that has no rear breaks [22:37:22 CET] <momomo> is it possible to generate hls live stream from a local file so it becomes a live stream, just as if it was a url ? [22:37:59 CET] <momomo> i am able ot generate a hls stream, but it is happening at full speed, so that the playlist is replaced faster than actual playtime [22:38:31 CET] <kepstin> momomo: sure. If you use the '-re' input option, it slows down the input file reading to realtime [22:43:34 CET] <momomo> kepstin: ok, i will try that .. however, isn't that a bug to be defualt to the other way ... since the m3u8 will be replaced immediately ? [22:44:00 CET] <kepstin> the default for ffmpeg is to run as fast as possible, it's designed mostly as a batch processing tool [22:44:17 CET] <kepstin> and it's pretty common to pre-generated static hls files for the entire length of a media file, too [22:44:24 CET] <kepstin> rather than "live" streaming it [22:46:55 CET] <momomo> kepstin: so when it is an external file ... it will act as the -re option by default ? [22:47:02 CET] <momomo> or should I use that then too ? [22:47:41 CET] <kepstin> if your source is live (e.g. a remote media stream or capture card), then ffmpeg will default read it as fast as possible ... which is realtime. You should not use -re in that case. [22:48:44 CET] <kepstin> Use -re only when you are making a fake live stream, where you have to slow ffmpeg down to run at realtime speeds. [23:03:46 CET] <DarkUranium> hey [23:04:09 CET] <DarkUranium> for the file list (e.g. -f concat -i list.txt), is there a way to specify a start/end time for a segment? [23:11:15 CET] <klaxa> and how would that work? [23:13:20 CET] <DarkUranium> klaxa, since each file has a `file '...'` line, could do something like [23:13:25 CET] <DarkUranium> `file '...' -ss ... -t ...` [23:13:36 CET] <DarkUranium> so it *COULD* work. [23:13:43 CET] <DarkUranium> also unrelated, is there a way to cut a video to the end? [23:14:39 CET] <klaxa> i don't understand anything you said [23:14:47 CET] <klaxa> what kind of video do you want? [23:15:00 CET] <klaxa> concat means glue one video after another [23:15:06 CET] <klaxa> i don't get what you want with timestamps [23:15:17 CET] <DarkUranium> klaxa, some of the parts, I don't want the entire thing [23:15:19 CET] <klaxa> and can you explain the "cut a video to the end" thing? [23:15:25 CET] <klaxa> aah [23:15:31 CET] <DarkUranium> just (say) 00:05:13 to 00:13:12 [23:15:33 CET] <DarkUranium> for file A [23:15:37 CET] <DarkUranium> but file B needs to be whole [23:15:42 CET] <klaxa> ok i see now [23:15:47 CET] <DarkUranium> file C needs to start at 00:50:00 but go to the end [23:15:59 CET] <DarkUranium> but nvm, I've just decided to cut in a separate step [23:16:11 CET] <klaxa> i don't think there is an option for that and this usecase is quite specific, but it shouldn't be too hard to think of some combinations of multiple ffmpeg runs [23:16:13 CET] <furq> if you need such precise cuts you'll probably need to reencode anyway [23:16:32 CET] <furq> in which case you can do it in one pass with the select/trim and concat filters or something along those lines [23:16:32 CET] <DarkUranium> furq, is there a way to force it to do it lossless? [23:16:42 CET] <DarkUranium> it doesn't have to be that precise, they were just examples. [23:16:56 CET] <klaxa> if you cut at keyframes you can use -c copy [23:16:59 CET] <furq> use -c copy if you don't want to reencode, but yeah [23:17:04 CET] <furq> you can only cut at keyframes if you do that [23:17:23 CET] <DarkUranium> and if I don't? Will the entire video get messed up (for h264, if it matters), or will it only be a lossy cut until the first keyframe? [23:17:31 CET] <furq> it'll seek to the nearest keyframe [23:17:40 CET] <DarkUranium> that's CloseEnough(TM) [23:17:55 CET] <furq> with h264 it'll probably be within about five seconds [23:18:33 CET] <DarkUranium> actually, looks like my time just happened to hit it exactly [23:18:40 CET] <DarkUranium> that works *shrug* [23:19:02 CET] <furq> if you're cutting on a scene change or something then that'll often be a keyframe [23:19:17 CET] <DarkUranium> for future reference, is there a way to do this in a file list, in the same step? [23:19:22 CET] <DarkUranium> (provide start/end times, I mean) [23:19:55 CET] <furq> https://ffmpeg.org/ffmpeg-formats.html#concat [23:19:57 CET] <furq> apparently there is [23:20:30 CET] <DarkUranium> ah, I was reading https://trac.ffmpeg.org/wiki/Concatenate [23:20:30 CET] <furq> file 'foo.mp4'\ninpoint 01:23:45\nduration 00:01:23 [23:21:01 CET] <furq> or you might need outpoint instead of duration [23:21:17 CET] <DarkUranium> mm. [23:21:22 CET] <DarkUranium> and how do I reset it for the next file? [23:21:39 CET] <DarkUranium> (though I suspect I may need to provide inpoint *BEFORE* the file; I'll experiment!) [23:22:22 CET] <furq> All subsequent file-related directives apply to that file. [23:23:17 CET] <DarkUranium> where does it say that? [23:23:26 CET] <DarkUranium> meh, I'll just try it. [23:24:06 CET] <kam187> hmm so i'm using av_interleaved_write_frame to write audio [23:24:41 CET] <kam187> but if there's a gap, i.e. stream_audio_1 <10 second gap of no audio> stream_audio_2 it ignores the gap [23:24:47 CET] <kam187> the pts is correct [23:32:42 CET] <kam187> is there a way to fill the gaps with silence using ffmpeg? [00:00:00 CET] --- Sat Nov 19 2016

1 0

ffmpeg-devel.log.20161117
by burek 18 Nov '16

18 Nov '16

[00:38:41 CET] <muxketeer> attempting to make use of the zimg lib in ffmpeg. Im interested in using the st_2084 color transfer function available in the latest version of zimg. I have compiled the latest version of zimg and compiled ffmpeg pointing to that latest version of zimg. however the following command doesnt work root@71cd9cfee65a:~/ffmpeg_sources/ffmpeg# ./ffmpeg -loglevel debug -y -start_number 86400 -f image2 -r 23.976 -i /path/to/ti [00:38:55 CET] <muxketeer> os+accurate_rnd+print_info+full_chroma_int -vf showinfo,zscale=rangein=full:primariesin=2020:transferin=2020_10:matrixin=2020_ncl:range=full:primaries=2020:transfer=st_2084:matrix=2020_ncl,format=yuv420p10 -c:v rawvideo /path/to/rawframesYUV.yuv [00:39:14 CET] <muxketeer> with output being http://pastebin.com/LNhj9Jg3 according to zimg git hub page st_2084 is a valid enum for that argument. is there something Im doing wrong? [00:41:00 CET] <wm4> might need update of the zimg ffmpeg wrapper? [00:47:19 CET] <muxketeer> is that the case? I'm hoping it isn't . [00:50:12 CET] <muxketeer> was thinking that the 'st_2084' would just pass through to the lib without the ffmpeg wrapper needing to have the allowed enums duplicated. [00:50:17 CET] <kierank> nevcairiel: https://ffmpeg.org/pipermail/ffmpeg-devel/2016-November/202859.html [00:50:20 CET] <kierank> wtf is this about? [00:50:45 CET] <nevcairiel> i have no clue, it just seemed fishy to somehow modify the output of a supposed lossless format [00:50:50 CET] <nevcairiel> so i commented appropriately [00:51:05 CET] <nevcairiel> i didnt understand his response one bit, or the passive-aggressive nature of it [00:51:13 CET] <kierank> yes he is wrong [00:51:21 CET] <kierank> he's doing some weird rgb full range shifting [00:51:59 CET] <kierank> black needs to stay black, not end up with weird noise [01:02:46 CET] <iive> doesn't this use the msb as filler to the empty bits? [01:03:08 CET] <iive> aka, 00->0000 ; 7F->7F7F ; FF->FFFF ? [02:36:08 CET] <llogan> BBB: is VP an acronym, and if it is what does it mean? [02:58:28 CET] <Guest31671> hello, one point troubles me. why *(AVClass **)a not (AVClass *) directly? [03:31:13 CET] <BBB> llogan: I dont tihnk it stands for anything; vp3 was basically assume the m in mp3 stands for music, then vp3 is the same but v being video [03:31:27 CET] <BBB> llogan: and then all subsequent versions where vpn with n being >3 [03:35:03 CET] <llogan> ah, i had a feeling it didn't really mean anything [03:40:07 CET] <klaxa> the m in mp3 is from mpeg though, isn't it? [03:47:05 CET] <BBB> KLAXA: I didnt say it means music, I said assume it means music [03:47:36 CET] <BBB> going to bed, gnite :) [03:49:57 CET] <klaxa> ok then, well that's one explanation more [04:51:22 CET] <Compn> Guest31671 : ask michaelni when he returns [04:51:33 CET] <Compn> or ask on ml [12:22:17 CET] <cone-780> ffmpeg 03Michael Niedermayer 07master:85407c7e6372: avcodec/mpegvideo: Fix edge emu buffer overlap with interlaced mpeg4 [12:22:18 CET] <cone-780> ffmpeg 03Michael Niedermayer 07master:2c9106257ffc: avcodec/mpeg4videodec: Workaround interlaced mpeg4 edge MC bug [12:50:44 CET] <cone-780> ffmpeg 03Carl Eugen Hoyos 07master:f8247c0ccebc: lavc/ffv1enc: Support pix_fmt GRAY10. [13:01:51 CET] <cone-780> ffmpeg 03Carl Eugen Hoyos 07master:55a424c5a836: lavc/ffv1dec: Scale output for msb-packed compression to full 16bit. [13:13:49 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:48ee545d11e6: avformat/flvdec: Fix regression loosing streams [13:13:50 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:c9c619e667e1: avcodec/8bps: Check side data size before use [13:13:51 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:d8db018e313d: avcodec/cinepak: Check side data size before use [13:13:52 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:be756396b579: avcodec/idcinvideo: Check side data size before use [13:13:53 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:dc692ae1b727: avcodec/kmvc: Check side data size before use [13:13:54 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:3c1eb57d1eb5: avcodec/msrle: Check side data size before use [13:13:55 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:0c0aa5ebbaf7: avcodec/qtrle: Check side data size before use [13:13:56 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:7821c96dd050: avcodec/qpeg: Check side data size before use [13:13:57 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:956407b5df45: avcodec/msvideo1: Check side data size before use [13:13:58 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:aa896c182dae: avcodec/rscc: Check side data size before use [13:13:59 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:e8b93372819b: avcodec/rawdec: Check side data size before use [13:14:00 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:fa1ee9602632: avcodec/rscc: Fix constant [13:14:01 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:487accbf19be: avcodec/tscc: Check side data size before use [13:14:02 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:7521d5b8daae: avcodec/sunrast: Fix input buffer pointer check [13:14:03 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:07c5e65e6d6b: ffmpeg: Fix bsf corrupting merged side data [13:14:04 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:20d0f3201284: avcodec/movtextdec: Fix potential integer overflow [13:14:05 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:a0c6b4cfd169: avcodec/movtextdec: Fix tsmb_size check==0 check [13:14:06 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:8deaed3b12bf: avcodec/movtextdec: Add error message for tsmb_size check [13:14:07 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:7e8eb30f4002: avcodec/ituh263dec: Avoid spending a long time in slice sync [13:14:08 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:0f8de7a3db4a: avcodec/rv40: Test remaining space in loop of get_dimension() [13:14:09 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:3f6aae377a09: avcodec/mpegvideo: Fix edge emu buffer overlap with interlaced mpeg4 [13:14:10 CET] <cone-780> ffmpeg 03Michael Niedermayer 07release/3.2:b9a01722608a: avcodec/mpeg4videodec: Workaround interlaced mpeg4 edge MC bug [13:51:00 CET] <BBB> michaelni: ping [13:51:12 CET] <BBB> michaelni: when I ffprobe a mxf file, I dont get a fps, why? [13:51:19 CET] <BBB> it has an average fps, right? [13:57:46 CET] <michaelni> BBB, trying with a random mxf i get fps & tbr values [13:58:10 CET] <michaelni> Stream #0:2: Video: dnxhd (DNXHD), yuv422p(bt709/unknown/unknown, top first), 1920x1080, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 29.97 tbn, 29.97 tbc [13:59:20 CET] <BBB> I get Stream #0:0: Video: jpeg2000, yuv422p10le(progressive), 3840x2160, SAR 1:1 DAR 16:9, 59.94 tbr, 59.94 tbn, 59.94 tbc [13:59:27 CET] <BBB> for https://media.xiph.org/video/derf/meridian/MERIDIAN_SHR_C_EN-XX_US-NR_51_LT… [13:59:32 CET] <BBB> no fps there afaics [14:04:23 CET] <BBB> the r_frame_rate in -show_streams is set, though [14:04:26 CET] <BBB> and seems correct [14:04:31 CET] <BBB> so not sure why its not reporting that value [14:06:31 CET] <wm4> libavcodec/x86/vp9itxfm.asm:2149: error: instruction expected after label [14:06:37 CET] <wm4> with ffmpeg git [14:07:40 CET] <wm4> libass does it correctly: configure: WARNING: yasm is too old (found yasm 1.1.0.2352); ASM functions are disabled. [14:32:10 CET] <BBB> wm4: probably a missing HAVE_EXTERNAL_AVX2 or so [14:32:33 CET] <BBB> wm4: if you add that around the newly added iadst16 avx2 function (in .asm) and assignment (in _init.c), it should work [14:34:19 CET] <wm4> well it fetches directly from git [14:53:57 CET] <wm4> meh building ffmpeg qsv with mingw seems like a real PITA [14:58:05 CET] <RiCON> wm4: how so? [14:58:34 CET] <nevcairiel> probably want to use lucas libmfx thing with another build system [14:59:01 CET] <cone-780> ffmpeg 03Martin Storsjö 07master:4f7723cb3b91: movenc: Add an option for skipping writing the mfra/tfra/mfro trailer [14:59:02 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:4a485daa7fc5: Merge commit '4f7723cb3b913c577842a5bb088c804ddacac8df' [14:59:12 CET] <wm4> RiCON: include locations (well, general windows pain), msvc-only import libs, etc [14:59:30 CET] <RiCON> oh, you're not using libmfx [14:59:33 CET] <nevcairiel> the source of the library is shipped with the sdk [14:59:36 CET] <nevcairiel> you can just rebuild it [14:59:50 CET] <nevcairiel> or use lucas fork of it, he tacked on some build system [15:00:06 CET] <nevcairiel> https://github.com/lu-zero/mfx_dispatch [15:00:27 CET] <wm4> weird shit [15:01:43 CET] <cone-780> ffmpeg 03Martin Storsjö 07master:d825b1a53065: libopenh264: Support building with the 1.6 release [15:01:44 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:da97b244b04b: Merge commit 'd825b1a5306576dcd0553b7d0d24a3a46ad92864' [15:04:44 CET] <cone-780> ffmpeg 03Janne Grunau 07master:fc5cdc0d5372: doc: escape left brace in texi2pod.pl regex [15:04:45 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:711b7b77763c: Merge commit 'fc5cdc0d5372f5103c71d5dede296734fe71ead2' [15:05:16 CET] <cone-780> ffmpeg 03Janne Grunau 07master:5f74bd31a9bd: vp8/armv6: mc: avoid boolean expression in calculation [15:05:17 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:2818aaaba09d: Merge commit '5f74bd31a9bd1ac7655103b11743c12d38e0419f' [15:06:02 CET] <cone-780> ffmpeg 03Janne Grunau 07master:ec32574209f3: checkasm: vp8: mc: test unequal width/height for partitions [15:06:03 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:4fe013fc701a: Merge commit 'ec32574209f36467ef0d22c21a7e811ba98c15b6' [15:06:18 CET] <cone-780> ffmpeg 03Janne Grunau 07master:8c816c0c9b12: checkasm/arm: align the clobber check data properly for ldrd [15:06:19 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:90b72f6bdac9: Merge commit '8c816c0c9b12fdefd9046415e97df299880bc9b8' [15:06:36 CET] <cone-780> ffmpeg 03Martin Storsjö 07master:2866d108c9e9: vp8dsp: Remove the comment saying that the height is equal to the width [15:06:37 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:e999a4ed6c63: Merge commit '2866d108c9e9da7baf53ff57a51d470691049a57' [15:06:52 CET] <cone-780> ffmpeg 03Steve Lhomme 07master:99cf943339a2: d3d11va: don't keep the context lock while waiting for a frame [15:06:53 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:4c5c522fc1a5: Merge commit '99cf943339a2e5171863c48cd1a73dd43dc243e1' [15:18:53 CET] <cone-780> ffmpeg 03Anton Khirnov 07master:a8cbe5a0cceb: h264_ps: export actual height in MBs as SPS.mb_height [15:18:54 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:cca4fd477851: Merge commit 'a8cbe5a0ccebf60a8a8b0aba5d5716dd54c1595c' [15:19:58 CET] <cone-780> ffmpeg 03Janne Grunau 07master:17c99b6158f2: h2645_parse: handle embedded Annex B NAL units in size prefixed NAL units [15:19:59 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:62d9b7a69b9b: Merge commit '17c99b6158f2c6720af74e81ee727ee50d2e7e96' [15:21:23 CET] <cone-780> ffmpeg 03Janne Grunau 07master:80fbb7becae5: checkasm: vp8.mc: initialize the full src buffer after ec32574209f [15:21:24 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:c0af1ee90dcb: Merge commit '80fbb7becae530167373fe5178966b7d7604306e' [15:21:41 CET] <cone-780> ffmpeg 03Janne Grunau 07master:7b1ae0e73ab7: checkasm/arm: preserve the stack alignment checkasm_checked_call [15:21:42 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:286d8bae61e8: Merge commit '7b1ae0e73ab7f7c5eabc70dbe2e579127c6e154f' [15:25:05 CET] <cone-780> ffmpeg 03Vittorio Giovara 07master:61bd0ed781b5: h264: Log more information about invalid NALu size [15:25:06 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:2f1a539d4b90: Merge commit '61bd0ed781b56eea1e8e851aab34a2ee3b59fbac' [15:27:43 CET] <cone-780> ffmpeg 03Vittorio Giovara 07master:cbbb40405587: fate: Restore order of h264 entries [15:27:44 CET] <cone-780> ffmpeg 03Hendrik Leppkes 07master:1398ded7a77b: Merge commit 'cbbb404055877e3beb9890ffe22784a6a100963e' [17:24:39 CET] <kierank> "So AFAIK the encoder pushes the values to the LSBs but the decoder didn't [17:24:39 CET] <kierank> shift them back up?" [17:24:43 CET] <kierank> atomnuker: no [17:25:00 CET] <kierank> the decoder is fine [18:05:46 CET] <cone-780> ffmpeg 03Michael Niedermayer 07master:ae514b125431: avcodec/ass_split: Change order of operations in ass_split_section() [18:10:22 CET] <gnafu> Hehe, "ass_split". [18:26:06 CET] <microchip_> :D [19:03:52 CET] <kierank> wtf is wrong with carl [19:04:06 CET] <kierank> https://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=55a424c5a836523828b3b… [19:04:09 CET] <kierank> that's clearly wrong [19:06:41 CET] <wm4> but but it's still lossless! [19:07:03 CET] <kierank> on what planet is adding noise to the lsbs lossless [19:07:11 CET] <durandal_1707> was it approved by Michael? [19:08:27 CET] <kierank> this is literally cargo cult programming [19:08:31 CET] <wm4> he replied "probably ok" [19:08:33 CET] <kierank> adding random junk to the LSBs [19:10:14 CET] <jamrial_> so he pushed code that brings no benefits and makes the decoder slower at the same time? [19:10:34 CET] <kierank> correct [19:11:00 CET] <wm4> well... he said "Attached patch improves output for some ffv1 files imo." [19:11:29 CET] <wm4> (improves output of a lossless decoder, I still don't get it) [19:11:33 CET] <durandal_1707> some color issue [19:12:06 CET] <Chloe[m]> shouldnt colour issues be fixed using post processing/filters? [19:12:29 CET] <durandal_1707> probably related what he uses to display video [19:12:30 CET] <wm4> it should produce exactly what was input? [19:14:24 CET] <Chloe[m]> wm4: ofc, but if there an issue with the original source (or his display) then it should have been fixed through a filter (or similar) [19:14:45 CET] <Chloe[m]> kierank: why are you waiting two hours? [19:17:40 CET] <wm4> because just reverting a patch is offensive, no matter how wrong it is [19:18:25 CET] <wm4> or whether he was allowed to apply the patch at all (it was pushed before the doubts of others were resolved and only a "probably ok" was given) [19:18:58 CET] <Chloe[m]> ah ok, makes sense. (I didn't see it as offensive, but if other people think so then that's fine) [19:21:45 CET] <cone-780> ffmpeg 03Michael Niedermayer 07master:709c87109dc8: avformat/movenc: Check frame rate before use. [19:24:18 CET] <Chloe[m]> how does reverting a library version bump work? Is it bumped up one further? [19:26:18 CET] <BBB> I agree with kierank that the patch is ridiculous and wrong [19:26:26 CET] <BBB> even if it wasnt wrong, it would still be ridiculous [19:26:37 CET] <BBB> (but its actually wrong because of what he pointed out it not being RGB) [19:26:47 CET] <BBB> or, rather, fullrange [19:27:46 CET] <kierank> whoops yeah [19:29:03 CET] <BBB> its ok, we know what you mean [19:29:24 CET] <jamrial_> then please reply to that thread stating this, to reduce the chances of carl starting a revert war [19:30:26 CET] <kierank> well rgb implies fullrange but fullrange doesn't imply rgb [19:32:23 CET] <durandal_1707> Chloe[m]: not at all, also something like that should not happen [19:34:07 CET] <Chloe[m]> the main reason I was wondering is because if it's a straight revert then you run the risk of having two versions of the same version (only in tree, but that's still an issue I think) [19:34:11 CET] <Chloe[m]> s/tree/history/ [20:44:14 CET] <cone-780> ffmpeg 03Stefano Sabatini 07master:427a47abcdda: ffprobe: fix crash in case -of is specified with an empty string [20:45:47 CET] <iive> and what is the correct way to extend ranged data to 16 bits? [20:50:41 CET] <cone-780> ffmpeg 03Moritz Barsnick 07master:c493a531edfd: doc/bsfs: various improvements [20:51:28 CET] <drv> this is unrelated to full range vs video range anyway [20:54:18 CET] <ubitux> nevcairiel: were you able to entangle yourself from the merge madness? [20:54:35 CET] <ubitux> untangle* [20:56:58 CET] <Chloe[m]> ubitux: why does the EBU R128 filter have a video output? [20:57:21 CET] <BBB> iive: just upshift and leave lower bits to zero [20:57:45 CET] <BBB> iive: its kind of counterintuitive, but Im pretty sure its correct [20:57:46 CET] <ubitux> Chloe[m]: because it's cool, you can monitor the evolution of the loudness of a stream [20:57:51 CET] <ubitux> Chloe[m]: did you try it? [20:58:00 CET] <iive> no it's not. [20:58:17 CET] <BBB> its not counterintuitive? or its not correct? [20:58:32 CET] <BBB> (or both?) [20:59:27 CET] <ubitux> Chloe[m]: example from the doc: ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]" [20:59:28 CET] <drv> if anything is looking at the low bits of these formats, that code is the real bug (imo) [21:07:21 CET] <jamrial_> ubitux: yeah, once he got past the avconv.c stuff it was a breeze. noops and onliners galore [21:12:07 CET] <Chloe[m]> ubitux: hmm, I guess it's cool. Why couldn't it be integrated into the new ebur128 filter? [21:12:26 CET] <jamrial_> hah, carl waited exactly two hours to send that reply [21:13:02 CET] <kierank> I'm reverting [21:13:15 CET] <ubitux> Chloe[m]: i have no objection to integrating it to the new filter, but i don't want the "old" ebur128 to depend on new code dumped randomly [21:13:21 CET] <jamrial_> kierank: poke michaelni first [21:13:48 CET] <kierank> why should I do that when I wasn't given the courtesy of having my objection acknowledged [21:14:21 CET] <Chloe[m]> kierank: because you're not the maintainer either? [21:14:29 CET] <kierank> the maintainer system is a joke [21:14:38 CET] <llogan> it was "probably" reviewed [21:15:12 CET] <jamrial_> kierank: because he's is indeed the maintainer. and he will listen to your objections, unlike carl [21:15:18 CET] <kierank> he didn't [21:16:43 CET] <ubitux> kierank is getting more acid with time [21:17:02 CET] <durandal_1707> illusions [21:22:03 CET] <BBB> yo guys, some people (like me) are actually using ffv1 in production settings [21:22:08 CET] <BBB> Im expecting kieran does so too [21:22:14 CET] <BBB> its an international standard, after all [21:22:20 CET] <jamrial_> michaelni: could you confirm if kierank and BBB's arguments against carl's patch are right or not? [21:22:36 CET] <jamrial_> carl seems adamant about it being correct [21:23:31 CET] <jamrial_> BBB: that's also true. ffv1 is supposedly being standarized, so this kind of change can't be done lightly [21:23:55 CET] <BBB> Im certainly not happy about decoder behaviour changing for stuff I use in prod settings :) [21:24:09 CET] <BBB> but I can simply not git update for a while; we dont all have that luxury [21:25:14 CET] <BBB> (plus Im mostly using it for 8bpc, but thats a different issue :-p) [21:26:14 CET] <JEEB> wtf that mail thread [21:27:07 CET] <durandal_1707> the Carl code looks strange and explanation is weak imho [21:27:39 CET] <wm4> ubitux: not surprising [21:27:49 CET] <JEEB> is that some PAL pix_fmt specific code because he keeps mentioning it? can't be seen from the context of the patch [21:28:09 CET] <wm4> ffmpeg development is a total joke management-wise [21:28:15 CET] <JEEB> because he keeps mentioning 10bit PAL pix_fmt (?!) [21:28:21 CET] <wm4> it's pathetic and you should be ashamed [21:28:28 CET] <JEEB> or well, gray [21:29:14 CET] <ubitux> wm4: you're also part of the project you know [21:29:31 CET] <Chloe> wasn't there a meeting earlier this year to sort it all out? [21:30:02 CET] <jamrial_> we could ban carl if people actually voted in a relevant vote [21:30:36 CET] <Chloe> It still comes down to 'who will do the bug tracker then' iirc [21:30:46 CET] <wm4> ubitux: am I [21:30:49 CET] <Chloe> or at least that was what was said last time this came up [21:31:11 CET] <wm4> and you didn't notice how I was absent for months etc. [21:31:31 CET] <ubitux> wm4: you have several commits, you have write permissions (afaik), you are part of the voting comittee, and you're generally active in the project [21:31:56 CET] <ubitux> even if you're actively spreading toxicity on your bad days ;) [21:31:57 CET] <jamrial_> Chloe: last time a vote showed up, things got muddled and nobody voted [21:32:04 CET] <durandal_1707> I did do bugs, many of them on tracker [21:32:23 CET] <Chloe> durandal_1707: note that I'm not saying anyone doesn't [21:33:24 CET] <durandal_1707> Carl contributions are mostly controversial most of time [21:33:32 CET] <BBB> Im confused [21:33:36 CET] <BBB> so maybe I have an old checkout [21:33:49 CET] <BBB> but for me, packed_at_lsb is always set for bits_per_raw_sample > 8 && bits_per_raw_sample < 16 [21:33:55 CET] <BBB> so I dont see when that code ever executes [21:33:59 CET] <BBB> am I crazy? [21:34:23 CET] <JEEB> so it's for 8-15 bps? [21:34:26 CET] <JEEB> uhh [21:34:28 CET] <JEEB> 9-15 [21:35:20 CET] <BBB> look at http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/ffv1dec.c;h=0719e… [21:35:29 CET] <BBB> line 600-627 [21:36:09 CET] <BBB> thats the only cases where bits_per_raw_sample is between 9 and 15 [21:36:13 CET] <BBB> and they all set packed_at_lsb=1 [21:36:20 CET] <JEEB> hah [21:36:20 CET] <wm4> now I'm engaging cehoyos [21:36:22 CET] <wm4> just kill me [21:36:31 CET] Action: JEEB RIPs wm4 [21:36:32 CET] <wm4> might as well try to convince a wall while hitting myself with a hammer [21:36:41 CET] <BBB> oh I see its for grayscale only [21:36:48 CET] <wm4> (this is what talking to some ffmpeg "devs" feels like) [21:36:54 CET] <JEEB> yeah, he was mentioning gray [21:37:07 CET] <BBB> I still think it should respect fullrange [21:37:09 CET] <wm4> GRAY is considered YUV, so same rules [21:37:18 CET] <BBB> theres sufficiently many people that transform gray by taking y plane from yuv [21:37:25 CET] <BBB> (like, thats what I do) [21:37:33 CET] <BBB> so it should at least respect the range flag [21:37:37 CET] <wm4> well it is _actually_ not marked as rgb in ffmpeg's pixdesc [21:37:37 CET] <JEEB> ye [21:37:55 CET] <wm4> hm I don't think full vs. limited range has to do with this [21:38:27 CET] <wm4> hm I guess my argumentation makes no sense for the 16 bit case, whatever, bye [21:38:34 CET] <durandal_1707> pixel format should not have anything with range [21:39:01 CET] <JEEB> yes, they are orthogonal [21:39:07 CET] <JEEB> if that was the right word for it [21:39:46 CET] <wm4> if you have a 10 bit format and expand it to 16 bit, it will still have only 2^10 range [21:39:56 CET] <wm4> and the LSBs ought to be 0 [21:40:17 CET] <durandal_1707> but/and swscale is big mess [21:40:31 CET] <wm4> (and full range vs. limited range still happens within the 2^10 range) [21:40:49 CET] <durandal_1707> yes [21:42:11 CET] <wm4> oh I see ffv1 supports only gray8 and gray16 anyway [21:43:34 CET] <Chloe> huh, it's as simple as a shift? i.e. 10 0000 0000 -> 1000 0000 0000 0000 [21:43:44 CET] <JEEB> wm4: he added gray10 [21:43:51 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=f8247c0ccebc2a153eb76a8296… [21:44:16 CET] <wm4> hm [21:44:38 CET] <wm4> so basically for gray10 the decoder adds noise to the LSBs? [21:44:49 CET] <durandal_1707> and even with gray10 patch is still weird [21:44:52 CET] <wm4> while not for yuv410p10 [21:45:30 CET] <atomnuker> Chloe: not really, its better if you apply a dither after you shift [21:45:50 CET] <atomnuker> but yeah, you can get away with just a shif [21:45:58 CET] <atomnuker> *shift [21:47:28 CET] <Chloe> 512/1024 -> 32768/65536, ah. ok, I see how this works [21:49:09 CET] <Chloe> why is dithering better if an (expanding) shift is lossless? [21:49:30 CET] <durandal_1707> both Carl recent 2 patches to ffv1 are very ugly imho [21:51:30 CET] <durandal_1707> if one encodes gray10 he should get gray10 back, not something else [21:53:11 CET] <JEEB> Chloe: in real life your sources can have banding so that improves it as you widen the range [21:53:30 CET] <JEEB> you usually also dither audio when you resample [21:55:40 CET] <BBB> dithering makes sense when you reduce resolution (e.g. packing after applying a filter, or downsamplign from 10bit to 8bit) [21:55:53 CET] <BBB> when strictly upsampling, I dont think theres much of a case for dithering [21:55:57 CET] <JEEB> yes [21:56:03 CET] <JEEB> when downsampling it makes much more sense [21:56:26 CET] <BBB> and its true that swscale kinda sucks, the colorspace filter implements FSB dithering if you want [21:56:39 CET] <BBB> its obviously not fast [22:00:47 CET] <Chloe> chloe-: quit [22:02:14 CET] <Chloe> I can see how dithering on upsampling would make sense, but not for downsampling. If you are already losing information, then aren't you losing more by dithering on an already compressed sample? [22:04:50 CET] <atomnuker> you'll reduce banding if you dither when downconverting [22:04:52 CET] <BBB> Chloe: see wikipedia ;) [22:05:11 CET] <BBB> https://en.wikipedia.org/wiki/Dither [22:05:12 CET] <Chloe> BBB: yeah I'm reading now [22:05:26 CET] <BBB> Chloe: see pictures at bottom, going to 1bit makes the issue pretty clear and explains why dithering is useful [22:05:33 CET] <BBB> 8bit makes the issue less severe but its still there [22:10:11 CET] <Chloe> the 1bit example was very useful [22:12:41 CET] <wm4> isn't "dithering" on upsampling literally just adding noise [22:13:09 CET] <BBB> I dont know what youd dither to [22:13:18 CET] <BBB> there is no error diffusion since there is no error [22:13:22 CET] <wm4> yes [22:13:29 CET] <drv> it's essentially implementing the multiplication for range conversion in fixed point [22:13:34 CET] <wm4> so all you could do is adding noise [22:13:37 CET] <BBB> yeah [22:13:40 CET] <BBB> right [22:13:45 CET] <BBB> I think were saying the same thing ;) [22:14:06 CET] <BBB> its hard if theres no strict terminology that you have to abide by :-p [22:28:43 CET] <Chloe> Only when writing the email does my client wrap, but after it sends it doesn't wrap. Weird. [23:07:34 CET] <nevcairiel> ubitux: untangle? :) Skipped a few that didn't apply to FFmpeg.c without a lot of work, and bow there is a lot of more or less simple ones [23:08:52 CET] <ubitux> nothing particularly blocking in current state? [23:08:59 CET] <nevcairiel> Nah [23:09:05 CET] <ubitux> ok :) [23:09:08 CET] <ubitux> thanks for your work [23:09:15 CET] <ubitux> again i'm sorry not to be of any help currently [23:09:27 CET] <nevcairiel> I uploaded some new fate samples earlier so I can merge the new tests tomorrow [23:09:53 CET] <nevcairiel> There is one or two h264 changes soon that may take some exploring, but no huge sets [23:20:46 CET] <jamrial_> the two upcoming mp3 de/muxer commits will probably have to be noop'd. they clash with our implementation for the same thing [23:21:39 CET] <jamrial_> wonder if theirs is more correct. it is after all stream trailing padding, so using packet side data to signal packet trailing padding may not be "proper" [23:23:25 CET] <nevcairiel> I'll let someone fix that if its wrong, doesn't feel merge appropriate [00:00:00 CET] --- Fri Nov 18 2016

1 0

ffmpeg.log.20161117
by burek 18 Nov '16

18 Nov '16

[00:29:43 CET] <llamapixel> Is it possible to generate a video from ffmpeg without a starting video. Example I want to generate time code on nothing and end up with a black video of set size and a time code stamp across the media? [00:30:05 CET] <c_14> -f lavfi -i color=black [00:30:25 CET] <c_14> Check ffmpeg-filters(1) for Video Sources [00:30:49 CET] <c_14> https://ffmpeg.org/ffmpeg-filters.html#Video-Sources [00:36:13 CET] <llamapixel> thanks c_14 [00:37:46 CET] <klaxa> c_14: you know what the vp in vp8 stands for? [00:37:50 CET] <klaxa> video... p??? [00:38:03 CET] <llamapixel> progressive? [00:38:03 CET] <klaxa> can't find it on wikipedia :V [00:39:02 CET] <klaxa> looking at the rfc now [00:41:06 CET] <llamapixel> c_14 and how to timestamp that black video if you have a quick reference [00:41:39 CET] <c_14> drawtext filter [00:41:42 CET] <c_14> an example should be in the docs [00:42:04 CET] <llamapixel> thanks [00:43:33 CET] <c_14> klaxa: Video Processor? [00:44:29 CET] <c_14> I'm not sure it actually says anywhere [00:44:42 CET] <klaxa> would make sense, i also can't find any source [00:45:09 CET] <klaxa> if i can't find one by next week i'll put an email in my gmail draft folder, maybe a google employee answers it [01:38:45 CET] <llamapixel> c_14: thanks I got it with ffmpeg -f lavfi -i color=black -vf "drawtext=fontfile=/Library/Fonts/Arial.ttf: text='%{localtime}': x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1: boxcolor=0x00000000@1: fontsize=30" -r 25 -t 5 out.mp4 [01:51:09 CET] <llamapixel> c_14: Conways life is perfect for what I was hoping to do now I have the basics, thanks again for the head start. [02:35:17 CET] <llogan> klaxa: nobody know 11 years ago either: http://multimedia.cx/eggs/vp7-on2-just-wont-quit/ [02:56:52 CET] <klaxa> ha wow [02:56:57 CET] <klaxa> thanks for the link llogan [03:29:48 CET] <daemon> hey all how would I convert mp3 to ulaw [03:29:50 CET] <daemon> for asterisk [03:29:59 CET] <daemon> mono mp3 format [03:33:52 CET] <c_14> ffmpeg -i mp3 -f mulaw outfile [03:34:02 CET] <daemon> cool [03:36:16 CET] <llogan> klaxa: "bbb: I dont tihnk it stands for anything; vp3 was basically 'assume the m in mp3 stands for music, then vp3 is the same but v bring video', and then all subsequent versions where vpn with n being >3" [03:36:17 CET] <lOrD_dAeMoN> Can I convert .mov format? [03:36:42 CET] <c_14> lOrD_dAeMoN: don't see why not. You'll have to be a bit more specific than that though [03:38:17 CET] <lOrD_dAeMoN> Anyone got it? Fmpeg -i movie.mov -vcodec copy -acodec cop out.mp4 [03:39:19 CET] <llogan> what? [03:39:55 CET] <lOrD_dAeMoN> convert .mov format for mp4 [03:40:01 CET] <lOrD_dAeMoN> to mp4 [03:40:51 CET] <c_14> What's wrong with that? (besides ffmpeg vs Fmpeg) [03:42:06 CET] <lOrD_dAeMoN> I want to know if someone has already converted .mov to .mp4, that's it. [03:42:11 CET] <c_14> yes [03:42:25 CET] <lOrD_dAeMoN> What is the right command? [03:42:34 CET] <lOrD_dAeMoN> the best way? [03:43:03 CET] <c_14> what you wrote is 98% right. ffmpeg -i mov -c copy -map 0 out.mp4 is better because it maps all the streams (in case of multiple audio/subtitle/whatever) [03:43:17 CET] <c_14> You can also add -movflags +faststart depending on your use-case [03:44:53 CET] <lOrD_dAeMoN> thank you. I have this task to do at work and I have to use opensource tool. thanks again. I had never used it for .mov files [03:45:30 CET] <furq> i shudder to think what closed-source tool you were using for this [03:46:47 CET] <lOrD_dAeMoN> I have not used it, but I know that aTube Catcher does [03:47:06 CET] <lOrD_dAeMoN> :) [03:49:21 CET] <kurufu> Do I have to close AVFormatContext::pb to close the underlying rtp connection for an RTSP format? [03:50:01 CET] <kurufu> av_write_trailer() and avformat_free_context() both finish without error but the listening ffmpeg keeps listening like its expecting more frames.. [03:53:49 CET] <kurufu> Testing shows that rtsp is AVFMT_NOFILE like i thought... [04:22:57 CET] <kurufu> Looks like it was somethin weird with ffmpeg not sure. [04:35:57 CET] <mozzarella> hello [04:36:02 CET] <Sashmo_> can someone tell me how I can add frames to my clock that I am trying to overlay on my video? it looks like this '%{localtime\:%T}' [04:36:55 CET] <mozzarella> I created a video using ffmpeg and I can't play it on iOS, why is this? [04:41:08 CET] <mozzarella> shouldn't it be able to watch h.264 videos properly? [04:46:30 CET] <kurufu> what container did you put it in? [04:46:51 CET] <mozzarella> kurufu: mp4 [04:53:25 CET] <kurufu> sorry im not to sure about the intricacies of playing stuff on iphone... But h264 and mp4 sounds about right [04:53:43 CET] <kurufu> assuming you didnt do something silly like using 10bit encoding or something [04:56:53 CET] <mozzarella> I can watch it on linux, I can watch it on windows& [04:57:20 CET] <furq> mozzarella: pastebin the output of ffprobe -show_streams [04:59:40 CET] <mozzarella> furq: http://paste.fedoraproject.org/483505/55148147/ [05:00:57 CET] <furq> i suspect it's either the video pixel format or the audio codec [05:01:20 CET] <furq> the latter is easier to change so try that first i guess [05:03:55 CET] <mozzarella> what do I change it to? it's weird, nautilus says it's MP1, ffprobe says it's MP2 [05:03:59 CET] <furq> aac [05:06:56 CET] <mozzarella> how do I reencode it? sorry, I'm a newb with ffmpeg [05:07:54 CET] <mozzarella> I'm trying something [05:08:09 CET] <mozzarella> I'll let you know if it works [05:08:54 CET] <mozzarella> ffmpeg -i possible_plmd.mp4 -c:a aac possible_plmd2.mp4 [05:09:04 CET] <furq> that'll reencode the video as well [05:09:09 CET] <furq> -c:v copy -c:a aac [05:09:28 CET] <mozzarella> aaaww shit [05:09:33 CET] <mozzarella> must be why it took so long [05:13:06 CET] <mozzarella> the original video used aac, not sure what happened [05:15:20 CET] <furq> you probably just want to copy it from the original file then [05:15:37 CET] <furq> -i encoded.mp4 -i original.mp4 -map 0:v -map 1:a -c copy out.mp4 [05:18:50 CET] <mozzarella> furq: still can't watch the video from my ipod touch [05:18:55 CET] <mozzarella> with AAC sound [05:21:14 CET] <furq> if that's an old ipod touch you might need to use baseline h264 [05:21:26 CET] <mozzarella> it's the 5th gen [05:21:37 CET] <mozzarella> what's baseline h264? [05:21:54 CET] <furq> Note: iPad, iPhone 3G, and iPod touch (2nd generation and later) support H.264 Baseline 3.1. If your app runs on older versions of iPhone or iPod touch, however, you should use H.264 Baseline 3.0 for compatibility. If your content is intended solely for iPad, Apple TV, iPhone 4 and later, and Mac OS X computers, you should use Main Level 3.1. [05:22:00 CET] <furq> that's not really clear whether you should use main or baseline [05:22:11 CET] <furq> baseline will definitely work but it'll be larger and/or lower quality [05:22:18 CET] <furq> -c:v libx264 -profile:v baseline [05:22:36 CET] <furq> you probably also want -pix_fmt yuv420p [05:23:14 CET] <mozzarella> well, iOS is up-to-date enough that it should support main level, I think [05:23:24 CET] <furq> it depends on the hardware, not the os [05:24:04 CET] <furq> the builtin player uses a hardware decoder, so you're stuck with whatever that supports [05:24:25 CET] <furq> or a thirdparty player that decodes in software, but that'll eat your battery [05:24:32 CET] <mozzarella> what about itunes? I can't play the video from inside itunes [05:24:41 CET] <furq> no idea, i've never used apple stuff [05:25:26 CET] <mozzarella> good, stay away from them, that's what I wish I had done [05:27:25 CET] <furq> according to some forum post you should be able to use -profile:v main -level 3.1 [05:27:53 CET] <furq> you might want to scale the video to 1136x640, it'll encode faster and that's the native res anyway [05:28:05 CET] <furq> -vf scale=1136:640 [05:28:16 CET] <mozzarella> ffmpeg -i sleep/sleep11.mp4 -ss 06:35:45 -t 00:02:00 -c copy possible_plmd.mp4 [05:28:29 CET] <mozzarella> would that not keep the same audio codec? [05:28:34 CET] <furq> it should [05:28:46 CET] <mozzarella> well, it generated a file with a MP2 codec [05:28:51 CET] <mozzarella> while the original file is AAC [05:28:57 CET] <furq> weird [05:29:33 CET] <furq> does the original have multiple audio tracks [05:30:06 CET] <furq> that's the only way i can imagine that would happen [05:34:21 CET] <mozzarella> wait nevermind [05:36:54 CET] <mozzarella> ffmpeg -i sleep/sleep11.mp4 -ss 06:35:45 -t 00:02:00 -c:a AAC -c:v libx264 -profile:v main -level 3.1 copy possible_plmd3.mp4 [05:36:58 CET] <mozzarella> is that good? [05:37:44 CET] <furq> aac should be lowercase and delete that stray "copy" [05:38:09 CET] <mozzarella> oops [06:00:27 CET] <mozzarella> furq: thank you! it finally worked [09:05:14 CET] <mog> if i want to make a timelapse of a ton of image files i have how do i glob correctly? [09:05:22 CET] <mog> i tried image_%d.png [09:05:28 CET] <mog> but it didnt seem to match [09:05:44 CET] <mog> my images are formatted as image_UNIXTIME.png [09:22:13 CET] <Mysoft> mog you can create mylist.txt [09:22:21 CET] <Mysoft> with the images on the order you want [09:22:29 CET] <Mysoft> to be used as input for the concatenate [09:22:43 CET] <Mysoft> https://trac.ffmpeg.org/wiki/Concatenate [09:25:23 CET] <mog> hanks Mysoft [10:42:01 CET] <jj15> I want to stream a still image as a video. Ive managed to do this. Except, I also want to be able to modify the image and have the changes reflected in the video. For example, change the text in the image and have that update. Currently this doesnt work. Is it possible to do this with ffmpeg? [10:42:09 CET] <jj15> My current command: ffmpeg -re -loop 1 -i ./cam.jpg -r 10 -vcodec mpeg4 -f mpegts cam.mp4 [10:46:02 CET] <Mysoft> hum... i guess not directly [10:47:02 CET] <Mysoft> but what about this... trying to using a list.txt with two .jpg [10:47:06 CET] <Mysoft> and loop the "txt" file [10:47:17 CET] <Mysoft> may force it to reload from disk everytime like you want [10:47:37 CET] <jj15> Good point. Will try that. [10:48:37 CET] <Mysoft> otherwise you would need to have something like -f image2pipe and piping the .jpg yourself with a tiny program/script (at correct rate?) [10:48:43 CET] <Mysoft> unless someone else has some other idea :] [10:54:13 CET] <Mysoft> ---------- [10:55:58 CET] <jj15> Thanks Mysoft Ill look at both of these [10:56:13 CET] <Mysoft> oh and btw people... i have this situation i want to stream some video to ffmpeg with a raw yuv420p format... i dunno exactly how i should create an .avi for this purpose (with interleaved raw PCM/yuv420p chunks... i tried to get ffmpeg to generate an .avi of sorts, like -f avi -c:v rawvideo -pix_fmt yuv420p (but it ended as RGB32... probabily because the source for testing was a .gif file), any hints? [10:56:18 CET] <Mysoft> -- [10:56:41 CET] <Mysoft> and jj15, just remember to test with two different .jpg files on the .txt to make sure that it's looping the file list, not the first file on the list :P [10:58:37 CET] <Mysoft> ----- [10:59:18 CET] <Mysoft> ok people... got the answer of my own question... hahahha first generating a mp4 .avi and then with that avi as source generaing a rawvideo codec .avi did the trick to have yuv420p raw frames on .avi ^__^ [11:01:34 CET] <somaReverse> hi, how can I stop a screencast started by ffcast asynchronously [11:24:42 CET] <jj15> I tried doing the following & but it doesnt seem to work when replacing the images. I think the images must be loaded when the command is ran: ffmpeg -loop 1 -r 60 -f image2 -i img_%01d.jpg -vcodec mpeg4 test.avi [11:25:36 CET] <Mysoft> and looping the .txt didnt worked? [11:26:30 CET] <jj15> Oh darn it. Thats what I should have tried. Will try. Interesting though, when I view the video after stopping the command the change is visible so maybe it will work when streaming. [11:27:13 CET] <Mysoft> i think -f image2 is "autodetect" when you have a image as input [11:27:24 CET] <Mysoft> i mentioned -f image2pipe because it allows to use - as input [11:27:34 CET] <Mysoft> (if you're on linux) [11:27:57 CET] <Mysoft> and that allows you to cat the images yourself [11:28:15 CET] <jj15> Yeah Im on linux and I think youre right. [13:15:47 CET] <ZexaronS> hello [13:16:01 CET] <ZexaronS> I have many videos of several GB [13:16:18 CET] <ZexaronS> the nature of video is such that I do not need all of it [13:16:58 CET] <ZexaronS> infact, I was looking how to basically take 1 frame per second and make a new video , to basically throw away 90% of the frames [13:18:52 CET] <ZexaronS> however framerates reported in mediainfo and win explorer are heavily variated, there's 29, 23, 20, 26, 6, etc [13:19:22 CET] <ZexaronS> so I'm looking not to have to make a batch command for each of these cases [13:23:56 CET] <jkqxz> ZexaronS: The -r option on output files does exactly what you want, dropping frames to create output at the given framerate regardless of the input framerate. [13:26:35 CET] <ZexaronS> jkqxz, I guess indeed, however ofcourse more data will technically be lost in low-fps videos, but I think this is okay since most of them are above 20 so it's not a big deal to just separate those who are below 20, thanks [13:46:11 CET] <jj15> Confused. Im able to stream image to video to Facebook Live. I can see the preview fine. When I press Go Live it takes a while and then ffmpeg just quits. No errors or warnings. [13:48:14 CET] <jj15> Output: http://pastebin.com/U9DBfqfZ [13:48:20 CET] <Mysoft> so jj15 the .txt method worked? or you had to use something else? [13:48:30 CET] <jj15> The text method works fine :) [13:48:38 CET] <Mysoft> ok :] [13:49:07 CET] <jj15> Both methods work out ok. Im a ffmpeg noob so getting my head around all the options is daunting. Getting there slowly. [13:49:39 CET] <Mysoft> you should remember to not paste your key when doing such pastebins :) [13:49:57 CET] <jj15> Oh damn! [13:51:10 CET] <Mysoft> maybe facebook is closing the connection [13:51:43 CET] <Mysoft> i know that twitch do that when you send too little data [13:51:59 CET] <Mysoft> or invalid framedata [13:52:09 CET] <Mysoft> but... you're already changing your jpg? [13:52:15 CET] <Mysoft> you tried it without changing the jpg? [13:56:03 CET] <jj15> Mysoft: testing now. [14:00:33 CET] <Mysoft> that's a weird pixel format you're using there [14:01:05 CET] <Mysoft> for twitch i usually use -pix_fmt yuv420p / yuv422p [14:03:44 CET] <jj15> Oh, I thought I was using yuv420p & -pix_fmt yuv420p \ [14:04:48 CET] <jj15> Heres a script I created for a single image to test: http://pastebin.com/JjPVApAn [14:07:03 CET] <Mysoft> heh [14:07:13 CET] <Mysoft> then i dunno why the warning there [14:07:33 CET] <jj15> I dont get why the video ends on Facebook but previews fine. [14:08:00 CET] <Mysoft> ah... that's from input :) [14:08:23 CET] <Mysoft> it's scaling on the "weird" jpg input format ok [14:08:31 CET] <jj15> hmm [14:09:48 CET] <Mysoft> heh i don't know... [14:09:57 CET] <Mysoft> can you check if it's sending the data [14:10:15 CET] <Mysoft> because the upload of ffmpeg rtmp is SYNC [14:10:20 CET] <Mysoft> (or at least i have this problem here on windows) [14:10:36 CET] <Mysoft> so i used a "tunnel" to buffer the RTMP output [14:12:09 CET] <Mysoft> also you seem to have a huge resolution there [14:12:20 CET] <Mysoft> and in your "script" i don't see which profile you used [14:12:36 CET] <Mysoft> *preset [14:13:13 CET] <Mysoft> so using a preset "medium" (that is the default), for 720p can be heavy [14:13:21 CET] <Mysoft> and would cause it to not have time to send everything [14:13:58 CET] <Mysoft> (nothing of this explains why the preview works but the real thing doesnt... but, if there's a problem with that...) [14:14:03 CET] <relaxed> jj15: remove -r 30 [14:15:22 CET] <Mysoft> but since you're just sending repeated jpeg's none of that suppose to be a problem... [14:17:22 CET] <relaxed> jj15: did you see https://www.facebook.com/facebookmedia/get-started/live ? [14:19:58 CET] <jj15> Sorry back. Will read through responses. [14:28:40 CET] <jj15> Im going to try with OBS just to make sure I can definatly stream without issues. [14:29:57 CET] <somaReverse> hello, This configure makes my computer slow and unresponsive https://ptpb.pw/iC51 . How can I tune it to be more performant ? [14:31:49 CET] <DHE> launch ffmpeg with `nice` at the front of the command (unix-based systems only) ? [14:33:27 CET] <ZexaronS> hi again [14:34:16 CET] <ZexaronS> now I'm trying to figure out how to specify variable for bitrate, all the videos have different bitrates, if I pick a fixed one some videos will have a lot more detail cut out than others [14:39:34 CET] <DHE> do you have file size constraints? -crf is intended to allow you to specify a target "quality" and ideally every video will look about the same -- leaves bitrate out of it [14:40:33 CET] <ZexaronS> oh stupid me lol [14:41:25 CET] <ZexaronS> I had big operation saving my DVDs to newer formats and archiving, I just haven't done this type of "FPS reduction" procedure I'm attempting now wiht totally different kind of data so I kind forgot a few things [14:42:21 CET] <ZexaronS> yes, it's 250GB I'm trying to redcue framerate but keep quality the "same" as original, to lower the space taken because the nature of data is very repetitive and slow [14:43:12 CET] <DHE> framerate reduction? [14:43:13 CET] <Mysoft> well you would need two pass... to do it with "variable bit rate" [14:43:28 CET] <Mysoft> so i think -crf is the answer [14:43:28 CET] <Mysoft> BUT [14:43:56 CET] <Mysoft> reducing framerate wont necessarily reduce size... if the source motion is already low [14:44:56 CET] <DHE> -crf is for maintaining a constant (perceived) quality and let the bitrate do whatever it needs to. target bitrate is for when you have a specific file size in mind. 2-pass is usually needed for bitrate mode to turn out okay [14:44:58 CET] <ZexaronS> well going from 30 to 1 surely will be some ... actually is it possible to go even lower like 0.5 and it would take 1 frame per 2 seconds ? , im still writting the batch file so I didn't test anything yet [14:45:23 CET] <DHE> good god what are you doing that you're okay with DVDs being 2 fps when they're usually around 30 ? [14:45:32 CET] <flux> zexarons, so you're saying the picture changes very little? [14:45:56 CET] <DHE> some kind of security camera feed archival? [14:46:03 CET] <Mysoft> or slideshow :P [14:46:15 CET] <flux> zexarons, perhaps having a big i-frame interval would be effective, ie. re-encode with -g 300 [14:47:05 CET] <ZexaronS> These aren't DVDs, this data is recordings from camera for some scientific stuff, I don't need it anymore but I don't mind if it takes 20GB and sits there for a few more years for reference [14:47:46 CET] <flux> (or maybe even -g 3000) [14:47:50 CET] <ZexaronS> yes ... but I don't need it to be a video, it's okay if it looks like a high-speed slideshow [14:48:04 CET] <Mysoft> well i think -crf 23 will be fine your needs [14:48:21 CET] <jj15> Someone here having the same problem re: Facebook preview works but live doesnt. Im wondering if not sending audio could be the problem. http://stackoverflow.com/questions/40644596/using-ffmpeg-to-stream-to-faceb… [14:48:35 CET] <Mysoft> you can use -maxrate to make sure it wont pass a certain limit (so automatically reducing crf) [14:48:46 CET] <Mysoft> *increasing crf [14:50:42 CET] <ZexaronS> .. i'll be making a test with these in mind [14:51:29 CET] <flux> I highly suggest you do time-limited trials firs t:) [14:51:38 CET] <Mysoft> indeed :) [14:51:53 CET] <ZexaronS> oh yeh ofcourse [14:52:01 CET] <Mysoft> altough he can press Q at any moment... so this isnt big deal :P [14:52:24 CET] <flux> sure, but comparing sizes is easier if the test sequences are all the same length ;) [14:52:36 CET] <Mysoft> heh good point :) [14:53:14 CET] <Mysoft> jj15 the audio part... could be relevant indeed [14:53:20 CET] <Mysoft> with twitch if i stram without audio [14:53:27 CET] <Mysoft> it seems to play "as fast as possible" [14:53:36 CET] <jj15> Im going to try send some audio with it and see if that helps. [14:53:38 CET] <Mysoft> ignoring the framerate [14:58:51 CET] <ZexaronS> there is no audio [14:59:05 CET] <ZexaronS> no streams [14:59:57 CET] <jj15> Forgot to compile with aac. Woops. [15:33:58 CET] <jj15> frame=167216 fps=617 q=-1.0 Lsize= 11409kB time=01:32:53.76 bitrate= 16.8kbits/s speed=20.6x [15:34:22 CET] <jj15> That doesnt look right. [15:36:16 CET] <Mysoft> that's with 1 fps output? [15:37:16 CET] <Mysoft> it seems a bit low for 720p output [15:37:40 CET] <Mysoft> but [15:37:49 CET] <jj15> Im getting a little confused now. fr is set to 30 [15:37:59 CET] <Mysoft> yes [15:38:05 CET] <Mysoft> but the frames as repeating on and on [15:38:17 CET] <Mysoft> so the effective kbits/s is very slow [15:38:31 CET] <jj15> Heres where Im at http://pastebin.com/zYN6Ae12 [15:39:01 CET] <Mysoft> -framerate 1 [15:39:06 CET] <Mysoft> so 1 fps input [15:39:16 CET] <jj15> Ah sorry, that was 30 I just changed it when you mentioned to test. [15:39:29 CET] <Mysoft> still it doesnt change [15:39:33 CET] <Mysoft> so -preset medium [15:39:39 CET] <jj15> Before I added the audio I got a preview. I dont get that, or errors. When changing fr to 1 I get errors on FB [15:39:46 CET] <Mysoft> will detect and discard repetitions [15:40:50 CET] <Mysoft> you could check using video.flv [15:40:53 CET] <Mysoft> instead of rtmp:// [15:41:01 CET] <Mysoft> to see how it outputs locally [15:41:29 CET] <jj15> I did output local and it worked ok. I got the image plus the audio. [15:42:31 CET] <Mysoft> you know it's interesting that on facebook how to [15:42:34 CET] <Mysoft> they say ffmpeg as option :) [15:42:38 CET] <Mysoft> twitch didnt bothered to do that [15:42:44 CET] <Mysoft> and using any other program doesnt work on windows :P [15:42:52 CET] <Mysoft> but back to the problem let's see... [15:43:31 CET] <jj15> I wish FB would report the errors. It only reported an error when I set framerate to 1 & told me its too low. But like now, it doesnt report anything. [15:43:44 CET] <Mysoft> yeah [15:43:48 CET] <Mysoft> because on the requirements [15:43:51 CET] <Mysoft> it said it has to be 30 fps [15:43:57 CET] <Mysoft> i.e. here [15:43:57 CET] <Mysoft> https://www.facebook.com/facebookmedia/get-started/live [15:44:17 CET] <jj15> Yeah I had read that. My audio encoding seems to be fine. Its only since adding the audio that I get an issue. [15:44:33 CET] <jj15> Whats a good thread size? Im getting this Thread message queue blocking; consider raising the thread_queue_size option (current value: 512) [15:45:00 CET] <Mysoft> you probabily should include -g 60 [15:45:08 CET] <Mysoft> to enforce the "keyframe every 2 seconds" [15:45:49 CET] <mlgh> Hi. I have a command: ffmpeg -ss 00:00:42 -i bug.mp4 -t 00:00:08 -c copy 11_bug.mp4 but it cuts out a video of length of 15 seconds. How should I convert bug.mp4 to some raw video format so that it would allow me precise time cutting fragments? [15:47:38 CET] <Mysoft> but hum... weird jj15 i don't see anything causing trouble... probabily later today i will try a live stream myself... and then i will know more :) [15:48:03 CET] <jj15> Mysoft: No worries. Youve been very helpful already :) [15:51:31 CET] <mlgh> Ah, got it, I must not do codec copy when seeking [16:23:25 CET] <IDWMaster> Hi. What does av_mediacodec_default_init take as its third parameter? Is the surface referring to the JNI object reference, or a native surface handle? [16:38:50 CET] <jj15> Posted here as a last ditched attempt. Really bugging me now http://unix.stackexchange.com/questions/324037/problems-streaming-to-facebo… [16:40:30 CET] <jj15> Oh wtf [16:40:34 CET] <jj15> I just tried again and it worked [16:40:38 CET] <jj15> This is so strange [16:40:51 CET] Action: jj15 presses go live [16:41:08 CET] <jj15> nope :( [16:41:50 CET] <bencoh> people use facebook to stream live videos nowadays? [16:44:01 CET] <jj15> There's nothing to worry about it's because the video may contain music, audio or a video clip that belongs to someone else: & whaaat thats my own music. Liars. [16:44:57 CET] <furq> i'd be pretty worried if my music belonged to someone else [16:55:36 CET] <jj15> Yeah same. Apparently it belongs to someone else & yeah me! https://soundcloud.com/imbase2/marys-stash [17:00:32 CET] <jj15> @Mysoft -re fixed it :D [17:00:42 CET] <Mysoft> what was it? [17:01:08 CET] <jj15> simulate real-time output, slow down read from file [17:01:46 CET] <jj15> My audio has some crazy VHS thing going on. Slowing and speeding. Sounds very nice actually, but not what I want. [17:06:28 CET] <bencoh> jj15: :D [17:09:12 CET] <jj15> @Mysoft also, streaming the single image and modifying it changes automatically on a stream. It didnt when I was watching the .mp4 stream locally. [17:09:32 CET] <jj15> Now just need to loop the audio and I think Im done. Ill put the script on github if anyone wants to play with it [17:12:28 CET] <jj15> Wow, looping audio looks complex. Ill probably make a super long audio file. [17:37:44 CET] <Sashmo_> can anyone let me know if its possible to use ffplay and ffmepg at the same time when capturing from a decklink source? I would like to preview whats coming in on my screen as I capture it. [17:39:29 CET] <c_14> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs [17:39:52 CET] <c_14> (one of your outputs will either be a pipe into the video player of your choice, or potentially -f sdl) [17:41:14 CET] <hamsheet> does anyone know of a way to delete "all videos that are less than a minute" type thing? [17:41:32 CET] <Mysoft> heh [17:41:57 CET] <Mysoft> jj15 i see what you mean by the lack of information from facebook... it's uploading... but facebook just say "disconnected" :P [17:42:27 CET] <jj15> Yeah its very frustrating. Sec Ill gist the code [17:42:36 CET] <jj15> s/code/script [17:42:36 CET] <Sashmo_> c_14: yeah I checked that but ffmpeg does do output to screen, I need ffplay for that, and I wasnt able to make that work from the examples [17:43:05 CET] <jj15> @Mysoft https://gist.github.com/JamesTheHacker/56fb9caf36244de582540e7d5c2fd6a5 [17:43:16 CET] <jj15> Could probably be done better but works [17:43:55 CET] <Mysoft> yeah this is what i'm doing pretty much [17:44:29 CET] <c_14> hamsheet: ffmpeg -f decklink -i whatever -c:v libx264 -preset ultrafast -f matroska - -c:v libx264 out.mkv | ffplay - [17:45:05 CET] <Mysoft> except i'm using just "aac" as encoder because i don't have the newer lib... and i'm using my proxy instead of letting ffmpeg connect directly [17:45:13 CET] <hamsheet> c_14 wrong guy? [17:45:30 CET] <c_14> hamsheet: right, I meant Sashmo_ . you have the same color [17:45:39 CET] <jj15> I use a proxy through an ssh tunnel seems to work ok. [17:45:50 CET] <c_14> hamsheet: as to your question, you'll need a script that uses ffprobe to get the duration and then checks if that's greater than a minute and then deletes it [17:46:51 CET] <hamsheet> c_14: let me see the ffprobe [17:47:51 CET] <hamsheet> c_14: ffprobe output is a single chunk of text ;( [17:48:21 CET] <c_14> hamsheet: `ffprobe -v quiet -hide_banner -show_entries format=duration -of compact=nk=1:p=0' that'll output just the duration [17:48:33 CET] <Mysoft> i didnt got it working even on preview so far... closest i got was "too low framerate" when i had a bad config :P [17:48:34 CET] <hamsheet> hmm let me see [17:48:35 CET] <c_14> (assuming the duration is stored in the format metadata which it usually is) [17:50:46 CET] <hamsheet> c_14: thanks this might be helpful but it is not working well on some videos, assuming what you said has exceptions [17:51:26 CET] <hamsheet> however it is a great start for me [17:54:51 CET] <jj15> @Mysoft very strange. Ive had issues with OBS in the past even though the settings were correct. [17:54:58 CET] <jj15> It was very hit and miss [17:55:16 CET] <Mysoft> i see [18:33:40 CET] <Mysoft> oh btw jj15 include -vsync 1 [18:34:13 CET] <Mysoft> in your github... because if the source is <30fps instead of duplicating frames ffmpeg is just stretching the timeframe [18:34:28 CET] <Mysoft> and then facebook will still detect as "source" framerate instead of 30 [18:34:46 CET] <jj15> Ahhh that explains the warped sounding audio then [18:35:01 CET] <Mysoft> with -vsync 1... it now duplicates the frames as it should [18:35:16 CET] <jj15> Added :) Thanks. [18:35:39 CET] <Mysoft> i had a .gif (10fps) as input, so it wasnt working because of that [18:35:42 CET] <jj15> So far its been streaming for over an hour with no issues. Just moving onto the next bit. Grabbing page data using the SDK and updating the image :D [18:38:04 CET] <Mysoft> ^^ [18:38:21 CET] <Mysoft> also the problem you had that it wasnt showing anything [18:38:31 CET] <Mysoft> is that facebook blocks input if invalid stuff happens [18:38:48 CET] <Mysoft> so any invalid attempt and then you may need to try again later [18:39:09 CET] <Mysoft> (but would be good if it would SAY anything.. instead of silently ignore) [18:39:43 CET] <jj15> Ah so maybe because it flagged my track as belonging to someone else that would have put me on a temp. block or something? [18:39:55 CET] <Mysoft> could be [18:40:13 CET] <Mysoft> altough it would make more sense to just mute it... like youtube does [18:40:26 CET] <jj15> Yeah true. [18:40:56 CET] <Mysoft> i just made it capture my muted mic [18:40:56 CET] <Mysoft> :P [18:41:27 CET] <Mysoft> i remember people on twitch had such problems before [18:41:32 CET] <Mysoft> but they relaxed the requirement [18:41:43 CET] <Mysoft> now instead automatically do nonsense crap [18:41:57 CET] <Mysoft> they accept "copyright infrigment" requests [18:42:12 CET] <jj15> Muted mic & thats a good idea! [18:42:13 CET] <Mysoft> that temp ban the source... but that rarely happens [19:28:35 CET] <cluelessperson> ffmpeg -f x11grab -framerate 2 -r 2 -s 1920x1080 -i :0.0 -t 01:00:00.000 -vf \ [19:28:35 CET] <cluelessperson> "drawtext=fontfile=/usr/share/fonts/truetype/ttf-dejavu/DejaVuSans-Bold.ttf: \ [19:28:36 CET] <cluelessperson> text='\%T': fontcolor=white(a)0.8: x=0: y=0" "$name.mp4" [19:28:43 CET] <cluelessperson> I cannot seem to get drawtext to work. [19:38:19 CET] <llogan> also, remove the "-r 2" and change "-s" to "-video_size". [19:54:29 CET] <jj15> @Mysoft done it :) https://www.facebook.com/OfficialUniversityRooms/videos/399534653710985/ [19:54:58 CET] <jj15> the video has a slight delay, but if you do a reaction it will increase the counter on the video [19:57:37 CET] <jj15> Ok I have to work something out. I think if the image is being saved as ffmpeg is reading it, it will mess things up [19:58:51 CET] <persina> What is the command to set the frame rate per second? [19:59:15 CET] <persina> I have a bunch of images and I want to convert to a .mp4 where 60 images are displayed per second... [19:59:54 CET] <furq> persina: -framerate 60 -i ... [20:00:34 CET] <persina> furq: That returns "Unrecognized option 'framerate' [20:00:59 CET] <furq> pastebin the full command [20:01:43 CET] <persina> furq: http://pastebin.com/F3P40ZYD [20:06:23 CET] <llogan> persina: also pastebin the complete console output [20:08:51 CET] <persina> llogan: http://pastebin.com/KGZ9rbxh [20:09:15 CET] <persina> Am I using an old version that doesn't recognize cetain flags? [20:09:24 CET] <llogan> absolutely ancient [20:10:33 CET] <llogan> where do you even find something so old and ratty? [20:10:47 CET] <furq> the clue is in parentheses [20:11:21 CET] <llogan> also, IIRC, 26402 was the last SVN version, so I'm guessing blindly following some "guide" [20:11:39 CET] <furq> nice [20:11:44 CET] <furq> persina: https://www.johnvansickle.com/ffmpeg/ [20:16:54 CET] <hiby007> Hello guys, I am recording a website using phantomjs, and then broadcasting it to facebook, using ffmpeg, here is the code. [20:16:55 CET] <hiby007> https://gist.github.com/bhargavjoshi/6d9866af4ac3d6e23d7049bfc9961535 [20:17:12 CET] <hiby007> however I am not hitting the minimum facebook mark or 30 FPS [20:17:16 CET] <jj15> Live stream is down for the moment. I need to work out a way to save update the image without ffmpeg going crazy. I think as its saving, if it tries to read it an error occurs and it stops the live stream. [20:17:36 CET] <jj15> hiby007: I seen your post on StackOverflow. Same person? [20:17:48 CET] <Mysoft> jj15 saving as a temp name [20:17:50 CET] <Mysoft> and quickly rename? [20:17:57 CET] <hiby007> this post ? http://stackoverflow.com/questions/40644596/using-ffmpeg-to-stream-to-faceb… [20:18:14 CET] <jj15> Mysoft: Thats what Im thinking. I was thinking of having PHP output an image stream and feed that into ffmpeg but not sure that will work. [20:18:35 CET] <Mysoft> it would [20:18:35 CET] <jj15> @hiby007 yes Im BugHunterUK [20:18:49 CET] <Mysoft> that's closer to the -f image2pipe option that i suggested :P [20:18:51 CET] <hiby007> I did not post it, i am Bhargav joshi . [20:19:15 CET] <jj15> @Mysoft Ah & yes youre right actually. [20:19:17 CET] <jj15> Will do that. [20:19:31 CET] <hiby007> jj15 your solution loops over the same image right ? [20:19:37 CET] <jj15> Yes [20:19:41 CET] <hiby007> then how do you update that image? [20:19:50 CET] <hiby007> and will ffmpeg use the updated image [20:19:51 CET] <hiby007> ? [20:19:59 CET] <jj15> I have a script that opens the background, and overwrittes the file. It seems to work. [20:20:21 CET] <jj15> It doesnt feel right though [20:20:22 CET] <hiby007> ok. trying that :) [20:21:04 CET] <jj15> the script runs every 5 seconds to get the new reactions from Facebook. [20:25:35 CET] <jj15> Are you trying to do the same thing @hiby007 ? [20:25:40 CET] <hiby007> yes [20:25:53 CET] <hiby007> tyring to see if I am able to maintain the framerate [20:26:09 CET] <llogan> did the comments in the answer not help? [20:26:37 CET] <hiby007> this one? [20:26:43 CET] <hiby007> "If you don't have audio add silence -f lavfi -i anullsrc & -c:a aac (might be BS but some seem to expect or require audio). You also may need to use YUV 4:2:0, so add -pix_fmt yuv420p as an output option, and try using -re with your inputs." [20:26:49 CET] <llogan> yes [20:26:52 CET] <hiby007> what do you mean by -re? [20:27:04 CET] <llogan> did you refer to the documentation? [20:27:07 CET] <hiby007> I am completely new to ffmpeg [20:27:10 CET] <hiby007> now let me do that [20:27:29 CET] <jj15> ffmpeg works fine, its just saving the image thats the problem. I think ffmpeg is stopping when it tries to read the image as its saving. Every time I run the PHP script to get the reactions and update the image it stops the stream. [20:27:29 CET] <llogan> http://ffmpeg.org/ffmpeg.html#Advanced-options [20:27:58 CET] <persina> Is there a flag to loop a video infinetley or is that something that's dependent on the media player? [20:28:16 CET] <jj15> in ffmpeg its -loop 1 I believe. [20:28:18 CET] <llogan> persina: still curious as to where you got ancient ffmpeg [20:28:47 CET] <persina> llogan: I was following some guide online... reinstalled it the right way now and it's working. [20:29:36 CET] <hiby007> llogan got it so without -re flag it's slowing the input, let me try your suggestions. :) Thank you [20:32:06 CET] <Mysoft> you may need a faster preset [20:32:15 CET] <Mysoft> like -preset veryfast/fast [20:32:28 CET] <Mysoft> the medium default can be slow [20:32:34 CET] <Mysoft> depending on the input [20:32:37 CET] <hiby007> phantomjs runner.js | ffmpeg -y -c:v mjpeg \ [20:32:37 CET] <hiby007> -f image2pipe \ [20:32:37 CET] <hiby007> -framerate 30 \ [20:32:37 CET] <hiby007> -f lavfi -i anullsrc \ [20:32:37 CET] <hiby007> -c:a aac \ [20:32:37 CET] <hiby007> -pix_fmt yuv420p [20:32:39 CET] <hiby007> -i - \ [20:32:39 CET] <hiby007> -re \ [20:32:39 CET] <hiby007> -r 30 \ [20:32:39 CET] <hiby007> -c:v libx264 \ [20:32:39 CET] <hiby007> -x264-params keyint=60 \ [20:32:40 CET] <hiby007> -b:v 1000k -minrate 1000k -maxrate 1000k -bufsize 500k \ [20:32:41 CET] <hiby007> -f flv 'rtmp://rtmp-api.facebook.com:80/rtmp/205825919865835?ds=1&s_l=1&a=AaaSkwwzC1QsyBLs' [20:32:41 CET] <hiby007> llogan is this right? [20:33:00 CET] <hiby007> opps sorry didn't knew it will past per line [20:33:05 CET] <jj15> Manages to output stream in PHP now to feed it into ffmpeg. I think your origional solution will work better @Myrsloik [20:33:14 CET] <jj15> @hiby007 pastebin :p [20:33:23 CET] <Mysoft> that doesnt have the -vsync 1 [20:33:59 CET] <Mysoft> if with -vsync 1 it doesnt reach the desired rate [20:34:13 CET] <Mysoft> one of both things... either the upload rate isnt reaching the desidred value [20:34:28 CET] <Mysoft> or the compression is too slow [20:34:38 CET] <hiby007> http://pastebin.com/3kBgLmiB llogan this gives error *Option framerate not found.* [20:34:49 CET] <Mysoft> hiby007 and update your ffmpeg :P [20:35:07 CET] <Mysoft> not having -framerate means it's like 4 years old [20:35:07 CET] <Mysoft> :P [20:35:26 CET] <llogan> hiby007: order of options matters [20:35:35 CET] <hiby007> I am on ubuntu no clue how to update it though :P Mysoft help please [20:35:38 CET] <llogan> you're attempting to apply -framerate to anullsrc [20:35:57 CET] <furq> -f lavfi -i anullsrc goes after -i - [20:36:13 CET] <jj15> @Mysoft So I should be able to use -f image2pipe, remove the image input and pipe the output from the script into ffmpeg? Going to try now and see if it works. [20:36:16 CET] <furq> and -c:a aac goes after all inputs [20:36:31 CET] <hiby007> ok. trying, but love you guys, I am like a five year old baby [20:36:49 CET] <furq> that's a pretty old baby [20:36:54 CET] <cluelessperson> so for the drawtext option I'm getting this error, "Stray % near T" [20:36:56 CET] <cluelessperson> text='\%T': [20:37:00 CET] <llogan> and -re is not an output option [20:37:20 CET] <KDDLB> cluelessperson, you probably need two percent signs [20:37:20 CET] <cluelessperson> llogan, okay. [20:37:31 CET] <furq> who runs fflogger [20:37:37 CET] <KDDLB> or remove the backslash [20:37:56 CET] <llogan> furq: burek who has vanished a few years ago. [20:38:00 CET] <furq> oh [20:38:24 CET] <furq> i would probably use it more if it didn't recommend pastie.org [20:39:05 CET] <llogan> is there something wrong with that site? and the last sentence is confusing to users because then they only provide command. i'll try to contact him. [20:40:06 CET] <hiby007> llogan is it ok to pm? llogan ? did not get you (and -re is not an output option) [20:40:30 CET] <llogan> here is the correct place. no need to pm. [20:40:45 CET] <cluelessperson> http://paste.ubuntu.com/23491992/ [20:40:48 CET] <furq> actually maybe i'm thinking of a different site, or they changed the defaults [20:40:59 CET] <cluelessperson> KDDLB, I still get the error. [20:41:07 CET] <furq> it used to have a tiny viewport and no word wrap, which is obviously bad for ffmpeg logs [20:41:36 CET] <furq> it's still not great but it at least wraps now [20:42:08 CET] <furq> although it won't let me create public pastes for some reason [20:44:55 CET] <hiby007> http://pastebin.com/7jceWFcz llogan, I now applies -re to input still not correct? [20:48:30 CET] <jj15> Yeeees the stream works!! [20:49:05 CET] <hiby007> jj15 what solution worked ? [20:51:10 CET] <jj15> @hiby007 This to create the image stream (unfinished) https://gist.github.com/JamesTheHacker/1159446b05ceaef588ccf4f7603d69b3 and then this for to pipe the stream into ffmpeg & https://gist.github.com/JamesTheHacker/78bf5c1d1b4e57e945de6bf072b7ca92 [20:51:24 CET] <jj15> Its all rough at the moment. Getting it to work then cleaning it up [20:51:37 CET] <hiby007> ok. [20:52:24 CET] <hiby007> jj15 neat you are using image instead of website :) [20:53:51 CET] <jj15> Knowing my luck it will either be a resource hog, or collapse completely ha ha [20:55:41 CET] <hiby007> haha, ffmpeg is already resource hungry I guess [20:56:30 CET] <hiby007> jj15 what is line 66 doing " echo $img->stream();" in https://gist.github.com/JamesTheHacker/1159446b05ceaef588ccf4f7603d69b3 ? [20:56:58 CET] <jj15> printing the raw image to stdout [20:57:18 CET] <jj15> which is then piped into ffmpeg [20:57:35 CET] <hiby007> but your ffmpeg command has this option -i img_0.jpg \ [20:57:53 CET] <hiby007> why is img_0 then? [20:59:36 CET] <jj15> Oh my bad, I think I linked to the old gist. [21:00:35 CET] <hiby007> link to new one? [21:00:55 CET] <jj15> One moment another issue. Let me get it stable and Ill upload the whole lot to github [21:01:14 CET] <hiby007> ok. [21:05:15 CET] <llogan> hiby007: -c:a aac should be an output option. -pix_fmt yuv420p should be an output option. option order matters. http://ffmpeg.org/ffmpeg.html#Synopsis [21:05:39 CET] <cluelessperson> Sorry for being kinda clumsy at first. I'm trying to do a drawtext with the current timestamp, however it doesn't seem to work [21:05:40 CET] <cluelessperson> http://paste.ubuntu.com/23491992/ [21:05:50 CET] <cluelessperson> I get video, but no overlay [21:06:08 CET] <cluelessperson> I think: Parsed_drawtext_0 @ 0x1b18a40] Stray % near '%T':00:00.00 bitrate=N/A is the reason. [21:06:11 CET] <hiby007> ok. llogan checking it. :) I am clueless, but let me try [21:06:20 CET] <cluelessperson> hiby007, clueless. :( [21:06:28 CET] <cluelessperson> don't use me as a negativism. :( [21:07:13 CET] <hiby007> sorry cluelessperson [21:08:26 CET] <furq> where did you get %T from [21:08:51 CET] <cluelessperson> furq, online example. >.> [21:09:05 CET] <furq> try %{pts\:hms} [21:10:47 CET] <cluelessperson> furq, do I need to escape the % ? [21:11:10 CET] <furq> text='%{pts\:hms}' [21:11:10 CET] <Daniel__> I'm having trouble having my stream run at 60fps when using only audio as an input (showwaves filter) [21:11:11 CET] <furq> works for me [21:12:11 CET] <cluelessperson> furq, well, I'm getting a timestamp on screen now, but it's not updating at all [21:12:17 CET] <llogan> Daniel__: use the "r"/"rate" option in showwaves [21:12:19 CET] <cluelessperson> furq, So definite improvement. :D [21:12:39 CET] <Daniel__> llogan, yeah the problem is that the stream still runs at 35fps. [21:12:48 CET] <Daniel__> if I remove -re it runs at 80fps [21:13:03 CET] <llogan> r or rate, not -re [21:13:14 CET] <llogan> showwaves=r=60 [21:13:32 CET] <Daniel__> llogan, yes I have that, my problem is that its running at 35fps. [21:13:37 CET] <llogan> oh, you're streaming [21:13:47 CET] <hiby007> llogan I think I am close? here is the log and command http://pastebin.com/Wa2F2twC [21:14:50 CET] <Daniel__> llogan, let my pastebin this, so you can see more clearly what I'm doing. [21:15:27 CET] <hiby007> jj15 any updates brother? [21:15:40 CET] <jj15> @hiby007 not yet just eating tea [21:16:10 CET] <hiby007> ok. :D [21:16:28 CET] <llogan> hiby007: i don't know. probably a scripting problem? [21:17:03 CET] <Daniel__> http://pastebin.com/hZAX7hMT [21:17:09 CET] <hiby007> does ffmpeg options look fine? [21:17:39 CET] <Daniel__> see line 65 for rate issues. [21:18:25 CET] <llogan> hiby007: you don't need -r 30 since -framerate 30 is used, but otherwise looks good i suppose [21:18:40 CET] <cluelessperson> furq, so that displays something, but it doesn't update [21:19:22 CET] <hiby007> ok. thanks llogan :D (Big brother) :) [21:23:39 CET] <Daniel__> llogan, I have a feeling its because audio has no framerate per-se and its breaking the realtime part of ffmpeg. [21:23:54 CET] <llogan> i don't know, but you'll certainly not want to use flv1 as the encoder. add '-c:v libx264'. [21:26:43 CET] <Daniel__> llogan, oh good spotting, take for granted libx264 being default on most containers. [21:29:02 CET] <_aeris_> hello #ffmpeg ! [21:29:12 CET] <_aeris_> i have a trouble to do something with ffmpeg [21:29:29 CET] <_aeris_> i have 2 videos, first with video+audio, second with video only [21:29:58 CET] <_aeris_> i search a way to put videos side-by-side on a file and to output audio on another file [21:30:07 CET] <_aeris_> https://paste.imirhil.fr/?1186e589712a1b99#CBt9ucLPjAJgKowraJDxnT+4KUkLtFE+… [21:30:30 CET] <_aeris_> i do this, video part is ok, but i cant find a valid command to have audio part :'( [21:30:33 CET] <furq> -i vid1 -i vid2 -filter_complex "hstack[out]" -map "[out]" -map 0:a out.mp4 [21:32:15 CET] <furq> the command you pasted has -an which disables audio, which probably explains why that's not working [21:32:47 CET] <_aeris_> thats not working because of Automatic encoder selection failed for output stream #0:0. Default encoder for format wav (codec none) is probably disabled. Please choose an encoder manually. [21:32:59 CET] <furq> -c:a libopus [21:33:22 CET] <_aeris_> i try to have *two* files at the end : one with only output from first video, second with both video (and no audio) [21:34:07 CET] <_aeris_> (audio in wav format, video in webm) [21:34:41 CET] <_aeris_> (I can achieve to do this with 2 ffmpeg commands, but will be better to do this with only one to minimize risk of desync) [21:36:31 CET] <hiby007> llogan this works http://pastebin.com/SBXtNifA but framerate drops after 6 seconds [21:37:41 CET] <_aeris_> furq > dont work better with libopus :'( [21:39:05 CET] <Daniel__> hiby007, -preset veryfast [21:41:42 CET] <Daniel__> hiby007, What quality is that source? and what CPU do you have? [21:43:03 CET] <hiby007> I am using jpeg as source, which comes from phantom js, cpu is Digital Oceans 16gb Ram server [21:44:09 CET] <hiby007> Daniel__ script is https://gist.github.com/phanan/e03f75082e6eb114a35c with framerate set to 30 instead of 24 in script [21:46:36 CET] <jj15> Sorry back. Going to carry on working on this. I should be able to do php live.php > img.jpg and then use img.jpg with ffmpeg. No need for image2pipe (I had troubles with that). [21:47:11 CET] <Daniel__> hiby007, I would run an output of -c:v copy file.ts to see if you can even get realtime working in the first place. [21:49:43 CET] <hiby007> Daniel__ didn't get you, you are saying that, phantom can't output the images at 30 fps? I also think so as phantom is using good amount of cpu [21:50:38 CET] <hiby007> can I just keep one image liek jj15 did earlier and then loop it and then if I update the same image will the updated image be used by ffmpeg? [21:51:28 CET] <Daniel__> hiby007, well you would hope it could, but unless their jpeg encoder is fast, it may not. the other thing also is that libx264 is expensive. [21:52:04 CET] <cluelessperson> Okay, one last problem [21:52:17 CET] <cluelessperson> At low framerates, ffmpeg (recording the screen) does NOT update the drawtext [21:52:27 CET] <cluelessperson> with a framerate of 24, it updates the clock fine [21:55:31 CET] <jj15> Can anyone confirm this stream is playing ok? (no stutters) https://www.facebook.com/OfficialUniversityRooms/videos/399659807031803/ [21:55:43 CET] <hiby007> ok. going to sleep. jj15 how can I be in thouch, since we are working on same problem ? [21:56:04 CET] <hiby007> jj15 stream works :) [21:56:13 CET] <jj15> Yeah sure @hiby007 and is there no stuttering? [21:56:20 CET] <hiby007> nope [21:56:24 CET] <hiby007> like the music [21:56:26 CET] <cluelessperson> jj15, waiting on stream to load [21:56:30 CET] <hiby007> wait stutters a bit [21:56:43 CET] <Daniel__> hiby007, but you best bet is to see how much CPU usage you have spare, a lot of video can be bottlenecked on a single CPU, especially something like a low clock server. [21:56:56 CET] <cluelessperson> jj15, not loading for me [21:57:00 CET] <jj15> Right going to move this code onto my server. Its useless trying to test from my desktop. [21:57:28 CET] <hiby007> spin a $10 digital ocean server, would suffice I think [21:58:33 CET] <jj15> Yeah thats what Im doing. DO is cool for this stuff. My machine is performing so slow. [21:59:05 CET] <Daniel__> NEVER underestimate the processing required for video! [22:04:58 CET] <jj15> Had to reboot. Had so many apps open and scripts running that my computer froze up. [22:09:33 CET] <hiby007> haha Daniel__ , jj15 true my laptop also heated up had to stop using ffmpeg on my mac :P [22:09:54 CET] <jj15> Yeah Im on a mac now. [22:10:13 CET] <Daniel__> Yeah don't run ffpmeg on a laptop haha. [22:14:47 CET] <blue_misfit> hey folks, I'm trying to use ffmpeg to package VP9 in DASH. This works but it takes many many minutes to actually package. I get a warning about my video input: [webm_dash_manifest @ 00000000000da4a0] Could not find codec parameters for stream 0 (Video: vp9, none, 852x480): unspecified pixel format [22:14:48 CET] <blue_misfit> Consider increasing the value for the 'analyzeduration' and 'probesize' options [22:15:12 CET] <blue_misfit> this is weird because ffmpeg -i <input.webm> does show the pixel format: [22:15:29 CET] <blue_misfit> Stream #0:0(eng): Video: vp9 (Profile 0), yuv420p(tv), 852x480 [23:41:50 CET] <jj15> Guys, all works now. If you go to the video, press one of the reactions and wait it will update the count on the live stream https://www.facebook.com/Reactions-Live-359241181078529/ [00:00:00 CET] --- Fri Nov 18 2016

1 0

ffmpeg-devel.log.20161116
by burek 17 Nov '16

17 Nov '16

[00:17:58 CET] <nevcairiel> jkqxz: so now vdpau works on amd but vaapi fails? [00:18:25 CET] <nevcairiel> is any of these amd modes not hacky? :d [00:18:37 CET] <nevcairiel> but hey it did improve on intel [00:22:21 CET] <nevcairiel> considering intel is likely the reference for vaapi behavior.. [00:23:50 CET] <jkqxz> Yeah. I think it's definitely a net positive change now, even if some mesa-vaapi does regress. [00:24:41 CET] <nevcairiel> i wonder what one has to do to get vaapi work with field coding [00:25:39 CET] <nevcairiel> the hwaccel hooks for that are mostly designed around how dxva2 wants it to happen, which is basically decoding both fields individually onto the same surface [00:26:05 CET] <nevcairiel> maybe vaapi is just missing the bitstream data though, i see a lot of empty values with "XXX: interlaced" :D [00:29:27 CET] <jkqxz> Yeah, I imagine VAAPI will work like that too. I'm only familiar with H.264, but that indeed does it by decoding onto the same surface twice. [00:30:35 CET] <jamrial> why isn't out software vc1 decoder bitexact anyway? [00:30:56 CET] <nevcairiel> mostly lack of interest from any good devs [00:31:04 CET] <nevcairiel> we lacked field coding support entirely for a long time [00:31:17 CET] <iive> it isn't? [00:31:20 CET] <nevcairiel> i think it was added in gsoc [00:31:35 CET] <nevcairiel> but its not bitexact, obviously [00:31:54 CET] <jamrial> kurosu touched it a couple years ago and yeah, he mentioned he didn't bother much with it since anyone watching vc1 would be using hardware decoding [00:32:44 CET] <nevcairiel> at some point i hacked in field coding hwaccel despite the software decoder not even supporting that yet, just so hw decoding would at least work [00:33:33 CET] <jkqxz> The gstreamer implementation for vaapi has a suspiciously matching set of "XXX: interlaced" comments. [00:33:41 CET] <nevcairiel> heh [00:33:50 CET] <nevcairiel> not very surprising [00:33:57 CET] <nevcairiel> stuff like vc1 doesnt get implemented multiple times [00:34:01 CET] <iive> they are (l)gpl, aren't they? [00:34:45 CET] <nevcairiel> nothing wrong with that, just sad sometimes to not have another independent implementation to bugcheck against [00:35:26 CET] <iive> :) [00:36:11 CET] <Compn> because we are all hoping vc1 dies a painful death [00:36:15 CET] <Compn> and we can just go h264 :P [00:36:51 CET] <nevcairiel> it gets very rarely used these days only, some odd blu-rays use it, but only very rarely, and outside of that its probably dead now [00:36:57 CET] <jkqxz> libyami looks different, possibly more complete. [00:37:00 CET] <nevcairiel> wonder what stuff like windows movie maker encodes into these days [00:37:01 CET] <jamrial> vc1 hasn't been relevant in forever. there are probably more blu-ray discs using mpeg2 than it [00:37:36 CET] <iive> it's like a zombie. it is dead, but it still goes on, because it is part of a standard. [00:38:31 CET] <nevcairiel> a few of the bbc documentary blu-rays used it, in field mode no less, it was quite odd =p [00:38:51 CET] <jkqxz> Compn: But I thought VP8^H9^H^H^HAV1 was the future... [00:39:13 CET] <jamrial> several years ago game a lot of trailers would be released in wmv/vc1. mainly those for x360 games [00:39:14 CET] <nevcairiel> we're undecided what the future is, but h264 is not it, its getting too old for that [00:39:24 CET] <jamrial> but nowadays everything is just dumped to youtube [00:39:25 CET] <iive> so AV1 is vp10? [00:39:36 CET] <nevcairiel> its like a hybrid of vp10, dalaa and t hor [00:39:49 CET] <iive> with wavelets? [00:39:50 CET] <nevcairiel> mostly vp10 with things from the others grafted on [00:40:39 CET] <nevcairiel> there was an article once exploring the techs they imported from dalaa [00:40:58 CET] <jamrial> it's pretty much vp10 with things like PVQ taken from daala and such [00:42:52 CET] <jamrial> i'm glad they are being combined for that matter. we really didn't need three formats competing for the same market (web video) [00:43:44 CET] <nevcairiel> vp9 was already pretty decent, it just lacked an ecosystem, maybe the AOM can fix that with AV1 and catch up to HEVC tech-wise in the same process [00:44:27 CET] <nevcairiel> they just need to put more effort into the encoder beyond the use-cases that google needs it for :/ [00:51:39 CET] <TD-Linux> yeah, there's a lot of stuff in the pipe for webrtc style use cases [02:44:43 CET] <cone-974> ffmpeg 03Thierry Foucu 07master:c512546689eb: Fix -Werror=parentheses error [02:44:43 CET] <cone-974> ffmpeg 03Vittorio Giovara 07master:00c80798160f: fate: Add h264 extradata reload tests [05:50:56 CET] <wpm> HELP: ERROR: gnutls not found using pkg-config when running configure on freebsd (freenas jail) [05:54:35 CET] <wpm> what do I need to do to move past this? [14:14:47 CET] <cone-248> ffmpeg 03Hendrik Leppkes 07master:99218ee30d7c: configure: properly add dxva2 link dependencies [15:06:49 CET] <cone-248> ffmpeg 03Michael Niedermayer 07master:d79d8ef92762: cmdutils: remove duplicate windows.h include [15:06:50 CET] <cone-248> ffmpeg 03kieranjol 07master:605f3084fc85: doc/filters: adds recently added -vf colorspace options [16:12:57 CET] <wm4> michaelni: e5c7229999182ad1cef13b9eca050dba7a5a08da breaks my code [16:13:54 CET] <wm4> it's also inconsistent with the audio path [16:23:23 CET] <michaelni> wm4, do you have a sample with which this results i wrong values being returned ? [16:23:50 CET] <wm4> hm didn't look at the date (2014), so it's unlikely that this caused my regression, odd [16:24:25 CET] <wm4> it's just a decoder wrapper which wants to allocate a frame using a different pixfmt from AVCodecContext [16:24:35 CET] <wm4> (not in git master) [16:39:12 CET] <nevcairiel> why wouldnt you update the avctx format then [16:44:14 CET] <wm4> because it's the format the user sees [17:21:09 CET] <wm4> when does an encoder need to set extradata? [17:27:19 CET] <nevcairiel> init [17:27:35 CET] <wm4> what if I have it only on the first frame? [17:27:45 CET] <nevcairiel> then you are screwed [17:27:53 CET] <nevcairiel> or do what some other do, encode a fake frame [17:28:10 CET] <jamrial> wm4: new extradata as side data? [17:28:13 CET] <wm4> that's terrible [17:28:24 CET] <wm4> jamrial: well it should be muxed properly [17:28:28 CET] <nevcairiel> jamrial: that doesnt work with the majority of containers [17:28:39 CET] <jamrial> containers should be adapted to handle it, then [17:28:43 CET] <jamrial> look at my flac matroskaenc patch [17:28:49 CET] <jamrial> which needs a review btw :p [17:28:49 CET] <nevcairiel> not all cases can, reliably [17:29:28 CET] <nevcairiel> you want streaming to work, so no seeking and writing it back later [17:31:36 CET] <wm4> obviously it could delay writing the first packet and thus everything else) [17:32:09 CET] <nevcairiel> for a generic solution you then end up delaying all streams by x packets in the hope the one stream you want finally sends its first frame [17:33:43 CET] <nevcairiel> what encoder is this about anyway? [17:34:01 CET] <wm4> mediafoundation [17:34:09 CET] <nevcairiel> thats not an encoder [17:35:09 CET] <nevcairiel> thats just an api [17:35:27 CET] <wm4> mf encoders are supposed to have s somewhat uniform behavior, but in this case it's the intel qsv mft [17:35:43 CET] <nevcairiel> why dont you use qsv as is [17:36:13 CET] <wm4> does intel ship libmfx with their normal drivers? [17:36:20 CET] <nevcairiel> the hardware variant yes [17:36:35 CET] <nevcairiel> software only comes with the sdk [17:36:41 CET] <nevcairiel> but who cares about the software decoder [17:37:15 CET] <wm4> anyway, my hope was that it's also going to work on nvidia [17:37:18 CET] <nevcairiel> also for MF, i would think the IMFMediaType would have a chance to contain extradata after init [17:37:36 CET] <wm4> yes, but it doesn't [17:41:03 CET] <wm4> actually it eats no less than 4 packets before it changes the output type [17:42:15 CET] <wm4> this might even be against the MSDN docs (and the MS h264 encoder includes the extradata on init) [17:43:46 CET] <nevcairiel> the funny thing is that QSV actually provides the extradata fter init, so there is no reason a MFT wrapping this couldnt do it [17:44:01 CET] <wm4> yeah [17:44:01 CET] <nevcairiel> but intel isnt very good at software [17:44:02 CET] <nevcairiel> so [17:45:05 CET] <nevcairiel> in any case still not sure what you hope to gain over just using qsvenc and nvenc respectively, after jkqxz recent work qsv at least works ok-ish again [17:47:32 CET] <nevcairiel> i should setup some intel test rig again to see if qsv is worth using now for $work [17:47:58 CET] <wm4> hm was the nvenc license situation resolved? [17:48:17 CET] <nevcairiel> nvenc is no longer nonfree [17:49:13 CET] <nevcairiel> the header was re-licensed as MIT by NVIDIA, and the binary it talks to is part of the driver, and if a driver isnt a system library worthy of the exception, then i dont know what is [17:49:24 CET] <wm4> I also can't find a libmfx.dll on my system [17:50:05 CET] <nevcairiel> it would be called libmfxhw(32|64).dll [17:50:18 CET] <nevcairiel> or sw for the software variant [17:50:52 CET] <wm4> oh right, they exist [17:51:07 CET] <wm4> same for nvenc I suppose (no nvidia windows system here) [17:51:19 CET] <nevcairiel> nvenc is directly part of the driver, yes [17:51:25 CET] <wm4> so can qsv and nvenv return/take d3d surfaces? [17:51:30 CET] <wm4> *nvenc [17:51:40 CET] <nevcairiel> nvenc can, not sure if avcodec exports that yet [17:52:00 CET] <nevcairiel> qsv probably as well, the qsv hwsurface stuff uses d3d surfaces in the background somewhere [17:52:33 CET] <nevcairiel> since d3d is windows specific it may not be exposed in either [17:52:44 CET] <wm4> because I'd like to avoid at least duplicating the filtering parts (the hope was that MF would make this transparent, but didn't test yet) [17:53:09 CET] <nevcairiel> i only ever used MF for software decoding of vc1, never had it deal with d3d [17:53:46 CET] <wm4> decoding to d3d surfaces works fine [17:55:13 CET] <nevcairiel> would be interesting to get a dxva2 zero-copy path from dxva2 decoding to nvenc and/or qsvenc, because dxva2 is just d3d9 surfaces, so it would be very generic [17:55:30 CET] <wm4> indeed [17:56:05 CET] <wm4> btw. the intel mft explicitly doesn't support d3d9 on windows 10 [17:56:33 CET] <nevcairiel> intel likes deprecating stuff fast [17:56:48 CET] <jkqxz> Hardware frame mapping support in libav should already be able to do that? I think it might be missing the explicit d3d->qsv mapping, but the support is all there. [17:56:53 CET] <nevcairiel> i still havent even bothered to play around with d3d11va because i have no reason to [17:57:08 CET] <wm4> what frame mapping [17:57:33 CET] <wm4> nevcairiel: the ability to actually access the nv12 data on the GPU without RGB conversion isn't a reason? [17:57:55 CET] <jkqxz> Also retrieve_data forces it into software in avconv/ffmpeg, so a bit more funny needed there. The equivalent cases in vaapi do work (decode with vaapi and encode with qsv). [17:58:01 CET] <nevcairiel> my main focus is copy back which w orks fine as-is [17:58:38 CET] <jkqxz> wm4: av_hwframe_map() and vf_hwmap [17:59:53 CET] <wm4> nevcairiel: well we want to be able to do it without copyback (later) [18:01:46 CET] <nevcairiel> zero-copy encoding isn't something I'm personally very interested in, just playback [18:02:32 CET] <nevcairiel> adding a full hardware path to the work encoding enging would probably mean i have to rebuild it .. again :p [18:03:28 CET] <wm4> you don't use ffmpeg.c? [18:03:44 CET] <nevcairiel> nah [18:04:13 CET] <wm4> what's your work anyway? [18:04:17 CET] <nevcairiel> that would give me way too many headaches [18:04:32 CET] <nevcairiel> rather have something simple and feature-limited that I know does what I tell it to do :D [19:40:32 CET] <cone-248> ffmpeg 03Andreas Cadhalpun 07master:90ebf3c42835: dds: limit 4 bpp handling to AV_PIX_FMT_PAL8 [19:40:33 CET] <cone-248> ffmpeg 03Andreas Cadhalpun 07master:a86ebbf7f641: libschroedingerdec: don't produce empty frames [19:40:34 CET] <cone-248> ffmpeg 03Andreas Cadhalpun 07master:3c0328d58d98: libschroedingerdec: fix leaking of framewithpts [20:37:04 CET] <rcombs> wm4: nevcairiel: fwiw autobsf delays muxing until the first frame is encoded; it'd probably be easy enough to provide some way to do that delay even when the muxer isn't using autobsf itself [20:39:57 CET] <nevcairiel> doesnt autobsf assume the to-bsf stream is also the first one to send a frame? [20:40:14 CET] <nevcairiel> or does it queue frames from other streams? [20:46:16 CET] <rcombs> it queues frames from everything until all streams have passed the muxer's check [20:47:10 CET] <wm4> and then there's the question how the extradata would even be propagated [21:26:30 CET] <cone-248> ffmpeg 03Michael Niedermayer 07master:2acee08a4a53: avutil/frame: Copy size=0 side data in ff_init_buffer_info() [21:26:31 CET] <cone-248> ffmpeg 03Michael Niedermayer 07master:721c90f0f99c: avutil/frame: fix indention after last commit [21:27:10 CET] <rcombs> wm4: it just waits to call write_header until all streams pass the check, so the encoder just has to put it in avctx (which ffmpeg copies to codecpar iirc) [22:39:29 CET] <cone-248> ffmpeg 03Andreas Cadhalpun 07master:ffdc5d09e498: exr: fix out-of-bounds read [22:39:30 CET] <cone-248> ffmpeg 03Andreas Cadhalpun 07master:ce3147eb1987: exr: reindent after previous commit [23:59:58 CET] <cone-248> ffmpeg 03Martin Vignali 07master:52da3f6f70b1: libavcodec/exr : fix channel size calculation for uint32 channel [00:00:00 CET] --- Thu Nov 17 2016

1 0

ffmpeg.log.20161116
by burek 17 Nov '16

17 Nov '16

[00:08:24 CET] <Chloe[m]> OK so it'd be manual. And, I have a lot of DV would h264 lossless + flac be the best way to compress it losslessly? (using veryslow preset) [00:13:24 CET] <furq> if by "compress" you mean "make bigger" then sure [00:14:03 CET] <Chloe[m]> you can't compress DV further? [00:14:17 CET] <furq> dv is already a lossy codec [00:14:30 CET] <furq> you can compress it further by using another lossy codec but you'll obviously lose quality [00:14:42 CET] <Chloe[m]> yes, but modern lossless codecs aren't better than DV? [00:14:51 CET] <furq> i doubt it [00:15:09 CET] <furq> it probably depends on the source so you'd have to test to be sure [00:15:18 CET] <furq> lossless h264 is huge, though [00:15:30 CET] <furq> and it's one of the better lossless codecs [00:17:03 CET] <furq> also if this is ntsc dv then i'm not sure if h264 supports yuv411p [00:17:27 CET] <furq> it's not listed in ffmpeg -h encoder=libx264 [00:17:35 CET] <furq> s/h264/x264/ [00:17:42 CET] <Chloe[m]> PAL DV luckily [00:17:45 CET] <TD-Linux> DV is basically a worse JPEG [00:20:19 CET] <Chloe[m]> yeah, this isn't going to compress more than DV :/ 50000kb/s where the DV is 30000kb/s [00:20:46 CET] <llogan> if you want to filter it such as deinterlacing and denoising, the sure, re-encode. if you have a bunch of disk space then just leave them. [00:20:54 CET] <furq> there are other lossless codecs you can try, but i'd be surprised if any of them beat 30mbps [00:21:55 CET] <Chloe[m]> I no longer have the tapes, so I want to keep the original streams. But it's quite a lot ~250GB, I was just wondering if I could reduce that [00:22:12 CET] <furq> compressing the audio should work fine, but i expect that's small consolation [00:22:21 CET] <Chloe[m]> I probably will reencode lossy with some filtering in the future though [00:23:43 CET] <llogan> i recommend your favorite deinterlacer with perhaps hqdn3d and maybe even drawbox to cover up any head switching noise on the bottom (unless you want to crop or scale or whatever but i prefer to just keep as it) [00:24:26 CET] <llogan> how did you capture the tapes? [00:24:42 CET] <furq> hardcoded black bars make me sad [00:24:56 CET] <furq> even though i know x264 still encodes them but hides them if you have a non-mod16 crop [00:25:09 CET] <llogan> furq: you could use a softsub [00:25:20 CET] <furq> huh [00:25:47 CET] <llogan> the noise annoys some people [00:25:56 CET] <furq> well yeah but i just crop that [00:32:54 CET] <Chloe[m]> llogan: via a custom program I wrote [00:33:16 CET] <Chloe[m]> well, custom ffmpeg device [00:33:20 CET] <Chloe[m]> still need to clean that up and send a patch [00:45:59 CET] <kbarry> I'm wondering if there is a switch for ffprobe to continually probe (i'm looking for changing metdata) [00:47:10 CET] <Chloe[m]> You can use watch for that [00:47:58 CET] <Chloe[m]> kbarry: watch -n1 ffprobe myfile.mp4 [00:52:04 CET] <kbarry> yeah, Thanks for the suggestion. Im using Mac. I found a solution that works: http://stackoverflow.com/questions/9574089/osx-bash-watch-command I should hav googled a little longer. [01:26:28 CET] <kurufu> so does anyone know how to generate the AVCodecParameters::extradata for a muxer reading encoded AAC data? [01:26:57 CET] <kurufu> It looks like its just a raw byte array and im not seeing any way to figure out what it should be but the muxer refuses to open an audiostream without it. [01:29:42 CET] <t4nk388> hello [01:47:09 CET] <kyleogrg> hello [01:48:33 CET] <kyleogrg> I'm trying to concat a few MP3s together. They have the same bitrate, sample rate, etc. The resulting MP3 has (in VLC) a duration that keeps changing. How can I fix this? [01:49:59 CET] <llogan> depends on how you're concatting [01:51:10 CET] <kyleogrg> ffmpeg -i "concat:file1.mp3|file2.mp3" -c:a copy "output.mp3" [01:51:34 CET] <furq> you probably want to use the demuxer [01:51:48 CET] <furq> https://trac.ffmpeg.org/wiki/Concatenate#demuxer [01:52:13 CET] <kyleogrg> Will this have a different effect? [01:52:19 CET] <furq> i assume using the protocol will result in mp3 headers in the middle of the file, which i guess explains the changing duration [01:52:32 CET] <furq> since the protocol is more or less just cat file1.mp3 file2.mp3 > output.mp3 [01:52:55 CET] <kyleogrg> okay [01:53:03 CET] <furq> the demuxer will actually demux the mp3 stream and write it into a new container [01:53:10 CET] <furq> s/stream/streams/ [01:53:28 CET] <furq> inasmuch as you can call mp3 a container [01:54:35 CET] <kyleogrg> okay, i'm trying it [01:55:17 CET] <kyleogrg> no, same problem... [01:57:02 CET] <kyleogrg> i put them into a txt file and then did ffmpeg -f concat -i myfile.txt [02:01:11 CET] <llogan> does it make a difference if you add "-write_xing 0" as output option? [02:03:15 CET] <kyleogrg> it actually helps a lot, but the duration still wavers by 1 sec at the beginning of the file [02:05:59 CET] <kyleogrg> ahh, but the duration is 6 minutes longer than it should be [02:33:27 CET] <kyleogrg> When I concat the MP3s to an MPG (no video), the duration is correct! Then when I copy this MPG to an MP3, the duration is crazy again. Why...? [02:48:07 CET] <kyleogrg> found a solution using a different program [02:48:10 CET] <kyleogrg> see ya [05:56:54 CET] <wpm> is this right channel for help building ffmpeg from source? [06:11:47 CET] <sim590> I have a question which is not related to ffmpeg, but which could most certainly be answered in this channel. I'm trying to use rtmpsrv, as suggested on http://askubuntu.com/questions/262248/record-live-stream, to get the rtmpdump command to use to save a stream. I do execute the iptables command and then run the rtmpsrv. I then, refresh a page with a stream on it (for e.g., twitch.tv), but I never see any [06:11:49 CET] <sim590> output from rtmpsrv. [06:12:10 CET] <sim590> Any idea what I could do to find out how to make it work? [06:19:56 CET] <sim590> or can I directly download a stream from a web page using ffmpeg? [06:23:20 CET] <codehotter> so I have a microphone and I need to record input from it and split it into a bunch of 200ms mp3 files. So 10 seconds would be 50 files. How do I do that? Can I do it realtime? What would be the lag for the next 200ms file to appear in my folder? What settings do I use to minimize that delay? [07:26:44 CET] <Pandela> Has anyone had progress with getting a video stream from ffmpeg into Manycams IP Camera feature? [08:09:09 CET] <fling> [libopus @ 0x562a5275e110] Queue input is backward in time [08:09:18 CET] <fling> Should I use another codec? ^ [09:48:11 CET] <c00lways> anyone know why am i getting Expected int64 for y but found 1 [09:50:49 CET] <durandal_1707> can you explain what are you doing? [09:53:09 CET] <james_> i ran ffmpeg command after compilation, and it returns that error when i tried to convert any video file [09:53:28 CET] <james_> ./ffmpeg -i /opt/webserver/video.h264 -itsoffset -00:00:00.4 -i /opt/webserver/audio.wav -c:v copy -c:a aac -strict -2 -y /opt/webserver/public/videos-source/recording/rpi1.mp4 [09:54:15 CET] <james_> i was using cross compilation on ubuntu for rpi1 [09:56:01 CET] <james_> @duranda1_1707, i can cross compile for rpi on a 64bit ubuntu 14 right? [09:56:39 CET] <james_> could it be due to rpi is 32bit? [09:57:03 CET] <james_> ffplayer did comes out its compilation information before segmentation fault [09:58:32 CET] <james_> btw, i found the static build for armel throws illegal instruction when i tried to encode via libx264 [09:58:51 CET] <james_> that is why iam trying to build it on my own [09:59:19 CET] <james_> and also i think it does not support openmax [10:23:33 CET] <c00lways> anyone have any clue? i saw this parameter has been checked inside width and height [10:24:08 CET] <c00lways> but other version of static build ffmpeg i downloaded doesn't have this issue [11:49:28 CET] <Pandela> I'm having trouble with udp streaming on my network :p [11:49:57 CET] <Pandela> I cant pick up the stream in ffplay on another computer in the network, pretty sure my firewall and stuff is out of the way [11:50:40 CET] <Pandela> Seems to be looping it back to my machine, instead of broadcasting on the network [12:21:47 CET] <kerio> how does udp streaming work? [12:21:54 CET] <kerio> just put the video in a mpegts and send it? [12:56:36 CET] <DHE> kerio: typically yes... multicast is popular as a means of doing so [12:57:09 CET] <kerio> does mpegts support uncompressed video? [14:04:53 CET] <DHE> kerio: I don't think so. but I wouldn't want to steam bulk data over UDP anyway. if you're talking uncompressed I assume that's a lot of bandwidth [18:19:18 CET] <Mysoft> assuming i have raw video for input and... x264 as output using -pix_fmt yuv420p [18:19:36 CET] <Mysoft> having that raw video as yuv420p speed up things a little? [18:22:23 CET] <kerio> probably yeah [18:22:32 CET] <kerio> it's a relatively simple conversion tho [18:23:11 CET] <Mysoft> yeah, but in my case it has a increased gain [18:23:41 CET] <Mysoft> because then i can convert the RGB32 capture to yuv420p before piping to ffmpeg [18:25:09 CET] <Mysoft> also since i have to copy from the 3D surface to a buffer to be able to pipe... doing the conversion during that point is also "extra effective" [19:02:34 CET] <bencoh> anyone checked libde265 vs ffmpeg native hevc decoder recently? [19:02:42 CET] <bencoh> (in term of speed) [19:24:45 CET] <sweLogan> Hi [20:02:33 CET] <sweLogan> Hello? [20:06:52 CET] <c_14> Just ask your question, and if someone can help you they will. [20:11:33 CET] <shincodex> heres my question [20:11:45 CET] <shincodex> send me gcc 6.1.0 binary for i686 [20:11:52 CET] <shincodex> all of them [20:13:01 CET] <c_14> That's neither a question nor ffmpeg related. [20:13:13 CET] <shincodex> ffmpeg doesnt exist without gcc [20:13:25 CET] <shincodex> but as for the question part.... hmmm [20:13:26 CET] <c_14> You can use clang [20:13:31 CET] <shincodex> lol shite [20:13:43 CET] <c_14> And ffmpeg doesn't exist without electricity but you don't see us talking about power generators in here. [20:13:54 CET] <kerio> cool story bro [20:14:11 CET] <shincodex> energy.... gotta stop using it up [20:37:17 CET] <sweLogan> I im trying to stream a webpage to twitch, for example like OBS where you enter your URL for donation alerts [22:06:35 CET] <ksk> hey, I read about "faststart movflag" - however in man ffmpeg on debian there is no mentioning of it. could you please help me out? thanks! [22:07:02 CET] <c_14> -movflags +faststart [22:07:09 CET] <c_14> Is in ffmpeg-formats iirc [22:09:34 CET] <ksk> I was missing the "+". thanks! [23:03:52 CET] <ksk> mhhm, more generally speaking, is it possible to convert $input into x264/mp3->mp4 and play that output as it is converting? added -movflags +faststart but it does not seem to play (vlc eg. says "moov atom not found") [23:04:16 CET] <TD-Linux> not with the mp4 container, no. [23:07:29 CET] <ksk> ah okay. which ones can do that? Is there an "easy" way I could find that out? [23:08:52 CET] <c_14> Pretty much every container that isn't mp4 [23:09:13 CET] <c_14> Though you can try enabling fragmentation (if the player you're using supports that) [23:13:34 CET] <ksk> hah, okay, gonna dig that. many thanks! [23:15:38 CET] <haasn> ./configure --enable-lto && make # fails with ar: libavdevice/alldevices.o: plugin needed to handle lto object etc. [23:16:05 CET] <haasn> Sounds like `ar` is being used by the ffmpeg build process incorrectly? you need to add --plugin <path> to get LTO support into ar [00:00:00 CET] --- Thu Nov 17 2016

1 0

ffmpeg-devel.log.20161115
by burek 16 Nov '16

16 Nov '16

[00:07:48 CET] <nevcairiel> it appears varying hardware either interprets the slices inside the single slicer buffer, or it doesnt [00:07:55 CET] <nevcairiel> wonder how much I need to poke the code to work [00:08:20 CET] <nevcairiel> lets see if the dxva2 vc1 spec has any insight [00:59:32 CET] <nevcairiel> wee the test passes [00:59:37 CET] <nevcairiel> and its only slightly ugly [01:05:15 CET] <jamrial_> nevcairiel: poke me if you need me to test anything on my hd7xxx [01:05:24 CET] <jamrial_> although i doubt vc1 decoding changed at all compared to your rx480 [01:05:42 CET] <nevcairiel> i dont have any amd gpus ready for testing [01:34:31 CET] <nevcairiel> none of the fate samples test field mode with slices :( [01:42:11 CET] <nevcairiel> jamrial_: jkqxz: if you feel like testing, this works for me on dxva2 on 10091 and 20021 https://github.com/Nevcairiel/FFmpeg/commit/72aec0deece401f7a9bd776396ac3bf… [01:42:20 CET] <nevcairiel> (too tired to split into proper commits) [01:43:03 CET] <jkqxz> sa10091 and sa20021 already pass on AMD with both VDPAU and VAAPI. [01:43:57 CET] <nevcairiel> the main problem I had was that dxva2_vc1 was not designed to handle multiple slices, so that may be true for vdpau or vaapi as well, if it fails after applying the patch [01:45:31 CET] <nevcairiel> i wonder where these samples come from [01:45:33 CET] <jamrial_> nevcairiel: no, that commit broke vc1 dvxa for every test here [01:45:43 CET] <nevcairiel> they smell like reference samples [01:46:02 CET] <nevcairiel> every? lets see if i pushed the right version [01:46:10 CET] <jamrial_> no hangs or crashes, just different crc output [01:47:06 CET] <nevcairiel> how odd, the non-slice tests shouldnt be affected, unless i broke something, which is always possible :D [01:47:31 CET] <jkqxz> For me the only change is that it breaks AMD VDPAU on the slice tests. AMD VAAPI continues to work and Intel VAAPI continues to not work. [01:49:12 CET] <jamrial_> nevcairiel: output is wrong, tons of green squares, at least on vc1_sa00040 [01:50:01 CET] <nevcairiel> guess i'll diff the data structures send to the driver then and see whats different [01:50:16 CET] <nevcairiel> cue my handy debug dxva2.dll wrapper that just dumps everything into a file :D [01:50:19 CET] <nevcairiel> but tomorrow [01:52:23 CET] <nevcairiel> its unfortunate that vdpau breaks in some way, its slice handling code is the most simple so if it would work, i would assume it would work now [01:59:13 CET] <jkqxz> The mesa video stuff is very hacky, so I wouldn't read too much into that. For vdpau it would be better to see it running with the nvidia driver (in some sense the reference implementation) and then work to get mesa to accept that. [02:33:52 CET] <jamrial_> nevcairiel: any way to check if a vc1 sample uses field mode with slices? [02:34:26 CET] <jamrial_> i have a bunch of wmv/vc1 video game trailers from years ago [02:36:18 CET] <kierank> jamrial_: afaik the only one is planet earth blu-ray 1080i59 [02:37:54 CET] <kierank> jamrial_: https://lists.libav.org/pipermail/libav-bugs/2011-December/000608.html [02:38:06 CET] <kierank> 1080i50 it seems [02:39:09 CET] <kierank> nevcairiel: ^ [02:39:15 CET] <kierank> 59 actually [02:39:18 CET] <kierank> interesting [09:34:28 CET] <nevcairiel> kierank: thats field coded but without slices [09:35:06 CET] <nevcairiel> the conformance samples have some some that should work like that, but i dont have the full conformance set, and I couldn't dig it up on the interwebs [09:40:51 CET] <kierank> nevcairiel: I think mike has them [09:40:58 CET] <kierank> And kostya [15:09:01 CET] <cone-922> ffmpeg 03Michael Niedermayer 07master:6ea271576822: avcodec/movtextdec: Fix potential integer overflow [15:09:02 CET] <cone-922> ffmpeg 03Michael Niedermayer 07master:a609905723c0: avcodec/movtextdec: Fix tsmb_size check==0 check [15:09:03 CET] <cone-922> ffmpeg 03Michael Niedermayer 07master:0eb319800567: avcodec/movtextdec: Add error message for tsmb_size check [16:43:24 CET] <wm4> michaelni: this file is detected as mpeg, even though it seems to be a perfectly normal mp3 file, are you possibly interested in looking at it? https://0x0.st/_ij.mp3 [16:47:59 CET] <jamrial> mp3 detected as something else instead of something else detected as mp3, heh [16:56:44 CET] <nevcairiel> gotta keep our life interesting [17:02:44 CET] <cone-922> ffmpeg 03Ronald S. Bultje 07master:83a139e3d85a: vp9: add avx2 iadst16 implementations. [18:28:01 CET] <cone-922> ffmpeg 03Michael Niedermayer 07master:2baf36caed98: avcodec/ituh263dec: Avoid spending a long time in slice sync [21:04:55 CET] <wbs> BBB: care to push the vp9/arm patchset? [21:05:02 CET] <BBB> okiedokie [21:14:05 CET] <BBB> wbs: pushed [21:14:08 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:6409e9b6ccde: vp9dsp: Deduplicate the subpel filters [21:14:09 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:86c5a23ee523: arm: Clear the gp register alias at the end of functions [21:14:10 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:68caef9d48c4: arm: vp9: Add NEON optimizations of VP9 MC functions [21:14:11 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:b4dc7c341eb0: arm: vp9: Add NEON itxfm routines [21:14:12 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:6bec60a683a5: arm: vp9: Add NEON loop filters [21:14:13 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:7fe898dbb949: aarch64: Add an offset parameter to the movrel macro [21:14:14 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:1f7801c2bc93: aarch64: vp9: Add NEON optimizations of VP9 MC functions [21:14:15 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:f43079e11cb4: aarch64: vp9: Add NEON itxfm routines [21:14:16 CET] <cone-922> ffmpeg 03Martin Storsjö 07master:f1212e472b5f: aarch64: vp9: Implement NEON loop filters [21:14:25 CET] <wbs> BBB: \o/ thanks! [21:14:38 CET] <wbs> BBB: I'll do some benchmarks for showoff now [21:14:51 CET] <BBB> very cool [21:15:10 CET] <BBB> lets see whos faster-than-c-er, amd64 or aarch64 [21:15:51 CET] <wbs> I guess amd64, since aarch64 still only has got only 16 byte vectors [21:15:59 CET] <wm4> michaelni: thanks [21:21:21 CET] <BBB> wbs: we shall see, right? :-p [21:21:37 CET] <BBB> and thanks for the patches, thats some amazing work [21:21:49 CET] <BBB> now if only chrome would start using it [21:23:02 CET] <wbs> BBB: the benchmarks are pretty fun, at least in the earlier ones I've done while working on it; in single thread mode, we're like marginally faster, a few percent or so. and then boom, you stop limiting it to a single thread, and it's just ridiculously faster :P [21:25:32 CET] <BBB> you mean compared to libvpx? [21:25:34 CET] <wbs> yeah [21:25:37 CET] <BBB> right [21:25:41 CET] <BBB> it doesnt really have threading [21:26:00 CET] <BBB> I mean, it has tiling and lpf threading, but ... [21:26:20 CET] <wbs> yeah, and when you've got a phone with 8 cores, not being able to use them is a bit of a waste [21:30:25 CET] <BBB> maybe they want to encourage you to get hw support ;) [21:48:14 CET] <gnafu> BBB: Gotta wait for a phone with AV1 support in hardware ;-D. [21:49:12 CET] <gnafu> (Which considering how long I tend to keep a phone, my next one probably will have VP9 and AV1 in hardware.) [21:50:48 CET] Action: llogan still uses free iPhone 4 girlfriend hand-me-down [22:02:07 CET] <cone-922> ffmpeg 03Andreas Cadhalpun 07master:1abcd972c4c0: mlz: limit next_code to data buffer size [23:12:27 CET] <cone-922> ffmpeg 03Michael Niedermayer 07master:1546d487cf12: avcodec/rv40: Test remaining space in loop of get_dimension() [00:00:00 CET] --- Wed Nov 16 2016

1 0

ffmpeg.log.20161115
by burek 16 Nov '16

16 Nov '16

[00:00:50 CET] <c_14> tried without the width_type ? [00:02:38 CET] <kbarry> much better. [00:03:01 CET] <kbarry> IF I want to grab, say everythign between 100hz and 600 hz, do I just ad two filters? [00:03:21 CET] <kbarry> or is using a widthtype of h, and a width of 600 appropriate? [00:04:24 CET] <c_14> either lowpass+highpass, or bandpass=f=600+100/2:width_type=h:w=500 [00:04:48 CET] <c_14> You can also use the firequalizer [00:05:42 CET] <c_14> Pretty sure you can also do it with the ladspa filter, and presumably another 5 filters I know absolutely nothing about [00:15:43 CET] <kbarry> c_14 I notice that in many of the files I play in ffplay) and look at he spectrogram (?), i noticed that some files seem to have the upper frequencies "muted" so to speak. [00:17:16 CET] <kbarry> If you would, what is this called? I'm trying to highligh this sort of "compression" where frequencies that are less noticible in an audio file are dropped to allow for the audio to be more highly compressed. [00:19:19 CET] <c_14> Lowpass? [00:23:56 CET] <t4c0c4t> @c_14 i found that library, symlinked it into my ldflags and am re-compiling now - sound like a good plan? :D [00:29:56 CET] <c_14> t4c0c4t: It's finding the file (or at least the dlopen call isn't failing) can you run strings .so | grep -i maxsupportedversion on the .so file? [00:35:00 CET] <t4c0c4t> no dice :( [00:35:21 CET] <kbarry> Yeah, i'm trying to use the showspectrumpic to generate the spectrograph. I like that its graduated. But the output is not the same as what i see using ffplay. [00:36:20 CET] <c_14> t4c0c4t: your drivers are firetrucked [00:36:22 CET] <c_14> What distro? [00:36:45 CET] <c_14> kbarry: ffplay uses fancy internal logic and not the actual filter afair [00:55:28 CET] <t4c0c4t> @c_14 amazon linux + nvidia AMI, not super surprised there exists an issue, but was hoping to avoid manual installation as part of my bootstrap process [01:58:49 CET] <kurufu> Is there a function to find all the compatible codecs for a given AVOutputFormat? [03:40:56 CET] <t4nk639> I realize it is possible using ffmpeg api but is it possible to grab video frame pixel data (R,G,B values) just using the binaries? [03:46:19 CET] <evga> Hello, is there anything I can do to speed up mpegts stream parsing in ffmpeg? [03:56:25 CET] <evga> the problem happens when I try to read from /dev/dvb/adapter0/dvr0 [04:03:13 CET] <kurufu> Is av_interleaved_write_frame (or av_write_frame) thread safe? Or should I stick it behind a mutex? [04:08:46 CET] <DHE> I don't think it is. ffmpeg isn't very thread-safe externally to itself [04:10:25 CET] <kurufu> Thats unfortunate. [04:10:28 CET] <kurufu> Thanks [04:12:21 CET] <DHE> submitting video and audio in different threads? [04:13:45 CET] <kurufu> Yea [05:43:58 CET] <t4nk284> Why i can not play 25fps video on 59.9Hz projector ? While BD player are doing it efficiently. [06:16:57 CET] <kurufu> im getting that AVStream::codec is deprecated, how am i supposed to set muxing options like width/height? [11:04:56 CET] <geo_> dvb-t .ts streams freeview recorded and converting to mkv with ffmpeg [11:06:22 CET] <kerio> is that a question [11:07:16 CET] <geo_> NO the subject that i have couple of questions about [11:08:56 CET] <geo_> the recorded .ts streams usually have a lot of errors... no frame!, SPS unavailable in decode_picture_timing, decode_slice header error, non-existing PPS 0 referenced and so on [11:09:17 CET] <geo_> but playback perfectly in VLC and SMplayer [11:10:41 CET] <geo_> ffmpeg spits out lots of errors when i attempt to convert them to mp4 to remove the transport stream errors [11:10:59 CET] <kerio> do you get something watchable at the end tho? [11:11:09 CET] <BtbN> that's what happens with Over-The-Air Streams [11:11:30 CET] <BtbN> Could also just be the first few seconds, until the next IDR-equivalent comes by [11:14:08 CET] <geo_> i can watch the original stream .ts fine on VLC, but when i attempt to convert it with ffmpeg to mp4 thats when i get all the errors [11:15:08 CET] <geo_> so the original broadcast streams contain errors which are invisble/ignored by VLC when played back [11:15:11 CET] <BtbN> well, but does it work fine? [11:15:41 CET] <geo_> the original recordings playback fine but ffmpeg spits out all the errors i listed before [11:15:54 CET] <geo_> when attempting to convert to mp4 [11:15:55 CET] <kerio> does ffmpeg actually stop [11:16:15 CET] <geo_> no just quits to command line prompt [11:16:32 CET] <kerio> is the mp4 viewable [11:17:20 CET] <kerio> also, are you actually transcoding it? [11:17:27 CET] <kerio> or just remuxing? [11:18:05 CET] <geo_> transcoding .ts to mp4 [11:18:20 CET] <BtbN> why transcode? [11:18:21 CET] <kerio> ok the actual output doesn't really matter [11:18:21 CET] <kerio> but [11:18:24 CET] <kerio> is it viewable [11:18:28 CET] <kerio> as in [11:18:35 CET] <kerio> does ffmpeg spit out errors and then continue working until the end [11:18:39 CET] <kerio> producing a valid video file [11:18:43 CET] <kerio> that contains roughly what you expected to see [11:18:58 CET] <kerio> or does ffmpeg stop in the middle at some point because some errors are fatal [11:21:31 CET] <geo_> the mp4 will play but audio sync problems and the navigation ffwd rrwd will freeze VLC or SMPlayer [11:59:13 CET] <CodecDev> Hello all. [12:00:17 CET] <CodecDev> I am writing a simple player using ffmpeg. Is there any API or parameter using which I can get PTS and DTS value in drop/non-drop frame format. Thanks in advance. [12:43:34 CET] <kerio> huh... i can put ffvhuff in rawvideo [12:43:36 CET] <kerio> how do i pull it out [12:43:39 CET] <kerio> can i pull it out [12:44:24 CET] <dsc_> i can pull something out [12:44:40 CET] <dsc_> :-o [12:50:40 CET] <c00lways> hi, I'm facing cross compile issue with ffmpeg, even thou I've all library with lib264 and libvorbis compiled to /home/vagrant/ffmpeg, it still states libvorbis not found [12:51:49 CET] <BtbN> check the config log and see what's missing [12:52:29 CET] <c00lways> okay [12:53:06 CET] <c00lways> i got this fatal error: ogg/ogg.h: No such file or directory [12:53:38 CET] <c00lways> ohh ogg missing [12:57:18 CET] <c00lways> when i configure libvorbis, i received this: Package ogg was not found in the pkg-config search path [13:23:32 CET] <c00lways> ohh i got it solved: export PKG_CONFIG_PATH=/home/vagrant/ffmpeg/lib/pkgconfig [13:40:14 CET] <kerio> ksk: -f containertype - [13:40:20 CET] <kerio> or -f containertype pipe: [14:00:03 CET] <c00lways> anyone encountered this when running ffmpeg? Expected int64 for y but found 1 [14:00:29 CET] <c00lways> is this related to arm32bit trying to run 64bit? [14:00:33 CET] <c00lways> ffmpeg? [14:38:05 CET] <CodecDev> I am writing a simple player using ffmpeg. Is there any API or parameter using which I can get PTS and DTS value in drop/non-drop frame format. Thanks in advance. [14:38:17 CET] <BtbN> in what format? [14:38:45 CET] <CodecDev> DF / NDF . [14:39:12 CET] <BtbN> And that's supposed to be what? [14:39:40 CET] <BtbN> you get the frame pts in stream timebase units. if you need some other base, you have to rescale it [14:40:19 CET] <CodecDev> Yes I want in timebase units only but the values to be as per NDF standard. [14:40:47 CET] <CodecDev> I hope you are aware about drop frame timecode. [14:41:26 CET] <CodecDev> https://documentation.apple.com/en/finalcutpro/usermanual/index.html#chapte… [14:42:02 CET] <BtbN> so it's some apple specialty? ffmpeg gives you exactly the pts I just mentioned. Everything else you have to come up with yourself [14:42:49 CET] <ozette> when segmenting a video (hls) do the segments have the same resolution as the source? [14:43:11 CET] <BtbN> ozette, if all you do is cutting it in parts, of course [14:43:21 CET] <ozette> okay thanks, wasn't sure [14:45:07 CET] <ozette> i don't recall where/how, but i had the idea that when transcoding ffmpeg may output whatever is 'default' [14:45:33 CET] <BtbN> You don't need to transcode to output hls. [14:45:46 CET] <BtbN> If you tell ffmpeg to scale, it can of course do that. [14:46:24 CET] <ozette> so, hls is essentially the same format as the source? [14:46:51 CET] <BtbN> hls is segmented mpegts and a playlist [14:47:00 CET] <ozette> i thought that video to hls was transcoding [14:47:02 CET] <BtbN> what you put in there and how you process is is entirely up to you [14:47:42 CET] <ozette> oh? but hls is a standard [14:47:44 CET] <DHE> HLS is (officially) stringent on the formats. H264 and AAC only. and the original intention was to make multiple variants (low,medium,high quality) available [14:47:52 CET] <DHE> well technically it isn't. it's a draft standard [14:48:15 CET] <DHE> but implementation is straight-forward enough and it's what apple devices officially support [14:48:24 CET] <ozette> i see [14:48:42 CET] <ozette> for different qualities i will first rescale the video to different resolutions [14:48:47 CET] <ozette> and then segment each of those resolutions [14:49:33 CET] <DHE> I've done this. ffmpeg can do all the variants in a single shot. in fact this is probably the best way to do it. [14:50:09 CET] <ozette> you mean i don't have to rescale first? [14:50:10 CET] <BtbN> not really [14:50:24 CET] <BtbN> running multiple parallel encodes in one ffmpeg process is kinda inefficient [14:50:58 CET] <DHE> eg: ffmpeg -i input.mp4 -s 1920x1080 -f hls 1080p.m3u8 -s 1280x720 -f hls 720p.m3u8 .... [14:51:12 CET] <ozette> aha [14:51:17 CET] <DHE> obviously that's very incomplete. you need codec parameters for each one. [14:51:19 CET] <BtbN> it's better to fire up multiple ffmpeg processes in parallel [14:51:31 CET] <BtbN> Otherwise the encoders will slow each other down [14:51:53 CET] <ozette> hmm, i will test this [14:53:35 CET] <ozette> the way i was originally planning to do this was serially [14:56:30 CET] <ozette> anyhow, video to hls is still transcoding right? [14:56:56 CET] <DHE> BtbN: I would argue otherwise. x264 is a parallel and background-capable encoder, so wouldn't a single process decoding and feeding multiple x264 instances be overall better than several ffmpeg instances? [14:57:04 CET] <ozette> video being any format [14:57:14 CET] <BtbN> DHE, no, because ffmpeg.c is strictly single threaded. [14:57:33 CET] <BtbN> It encodes one frame with every libx264 instance it starts, then progresses to the next frame, and so on [14:57:43 CET] <DHE> ozette: if the source is h264 and you're okay with using that as your bitrate, you can use it straight-up most of the time. [14:57:52 CET] <BtbN> So the slowest libx264 will slow down all the others as well [14:58:07 CET] <DHE> BtbN: but doesn't libx264 take the frame and return fairly quickly for background processing? (assuming not using zerolatency mode, etc) [14:58:11 CET] <DHE> I thought it did [14:58:32 CET] <BtbN> It has a delay, but it will eventually start outputting frames. [15:01:47 CET] <ozette> would love to see some metrics showing the difference in execution time until program return [15:03:27 CET] <BtbN> if the slowest encoder runs at 120 fps, it essentialy forces to run the entirety of ffmpeg to run at that framerate [15:03:51 CET] <BtbN> and even slower, as the one encoder runs at that speed, and after it's done, all the other encoders have to do their thing [15:11:33 CET] <kerio> BtbN: you get to decode only once tho [15:12:04 CET] <BtbN> so? Decode once, pipe to encode processes, done. [15:12:36 CET] <kerio> i only ever used multiple outputs in livestreaming situations [15:12:46 CET] <kerio> where i was hitting realtime anyway [15:12:53 CET] <kerio> but ack [15:18:26 CET] <[-T-]> Hey, is there a way in the code, to know if an option was set ? [15:18:39 CET] <[-T-]> For instance, i have "framerate" option which default value is 25 [15:19:01 CET] <[-T-]> i want to know if 25 was actually passed to the cmd line or if it is the default because the param was not set [15:20:18 CET] <kerio> is ultrafast that much worse than veryfast? [15:20:22 CET] <kerio> in libx264 [15:20:55 CET] <BtbN> iirc ultrafast forces it down to constrained baseline [15:21:12 CET] Action: kerio grumbles at the lack of ffmpeg 3 in freebsd ports [15:23:25 CET] <kerio> is -tune zerolatency actually important? [15:26:08 CET] <furq> no [15:26:26 CET] <furq> well yes but you shouldn't use it unless you need the very specific thing it does [15:26:50 CET] <furq> which is not "make my twitch streams appear fastar!!" as most people think [15:29:01 CET] <kerio> what does it do ;o [15:29:31 CET] <BtbN> it turns of reordering [15:29:46 CET] <furq> --bframes 0 --force-cfr --no-mbtree --sync-lookahead 0 --sliced-threads --rc-lookahead 0 [15:30:03 CET] <kerio> BtbN: reordering would be bframes right [15:30:15 CET] <furq> lots of things that you really don't want to do unless you need <1 frame of latency [15:30:45 CET] <kerio> what if i want those sub-10ms encodings [15:30:56 CET] <furq> if you want them then you'll need it [15:31:53 CET] <kerio> and then probably also intra-refresh, right [15:32:03 CET] <kerio> ...do commercial players even support that [15:32:03 CET] <furq> most people who use it because they saw it on a pokemons game streaming wiki are using a streaming service that introduces 5-60 seconds of latency anyway so they're just screwing themselves over for no reason [15:32:37 CET] <furq> no idea about commercial players, but it works in browsers on windows [15:32:56 CET] <furq> so it's not some specialist feature you need a specific decoder for [15:33:17 CET] <kerio> i wonder if it works in videotoolbox [15:38:11 CET] <kerio> it does [15:45:58 CET] <DHE> [-T-]: there is an SEI record in the stream that summarizes the encoding parameters. running 'strings' on your video might find it fairly quickly [15:46:26 CET] <[-T-]> DHE sorry it's not what I meant [15:46:29 CET] <[-T-]> it's simpler than that [15:47:12 CET] <[-T-]> i just need to know, inside the ffmpeg source code for instance in qsvenc.c, if a parameter was set from command line or not [15:47:49 CET] <[-T-]> basically, i want to differentiate between a parameter that is set to it's default value, because it was like this in the command line, or because it was not set [15:47:53 CET] <BtbN> there is no way to know that. Use a default value that can't be set via commandline [15:48:06 CET] <[-T-]> yeah, I used 12345 for the framerate [15:48:09 CET] <kerio> how do you manually set the option to what the consequence of "unset" should be [15:48:12 CET] <[-T-]> but it's a bit hackish ... [15:48:27 CET] <BtbN> usualy something like -1, INT_MAX or the like is used [15:48:53 CET] <[-T-]> Ok I will try these values, it's a better option indeed [15:48:54 CET] <[-T-]> thanks [15:49:01 CET] <kerio> or use another option [15:49:34 CET] <bencoh> zerolatency is "important" for some applications, and it does work when used properly :) [16:03:20 CET] <kerio> zerolatency would be great but i can't play my webcam without like a 2 second delay locally, with avfoundation [16:06:40 CET] <DHE> zerolatency is bad for when you have tight bitrate constraints because x264 has no planning capability [16:08:20 CET] <kerio> well i have 20mbps upload [16:08:23 CET] <kerio> :^) [16:10:26 CET] <DHE> JEALOUSY. [16:10:57 CET] <furq> i could upgrade to 20mbps upload but my isp calls that the GAMER package [16:11:04 CET] <furq> so i'm stuck on 10 until they rename it [16:11:19 CET] <kerio> why's that [16:11:28 CET] <furq> it's a matter of principle [16:12:13 CET] <furq> i'm not sure why they call it the GAMER package. quakeworld works just fine with 128kbps upload [16:13:14 CET] <kerio> oh look at me i'm so old i wouldn't know what these modern "GAMERS" do with lots of upload [16:13:17 CET] <spaam> furq: maybe they think that GAMERS share their gameplay on youtube or twitch :) [16:13:43 CET] <furq> sounds appalling [16:14:37 CET] <kerio> i'm pretty sure that my old 20/1 line was sometimes bottlenecked by the lack of upload [16:14:53 CET] <furq> i'm sure they accompany this gameplay with the "dub-step" i've heard so much about [16:15:22 CET] <kerio> it's all about trap nowadays i think [16:15:34 CET] <spaam> is big room house last year? [16:16:01 CET] <kerio> honestly gaming culture went downhill the moment people stopped using prodigy in their video [16:16:14 CET] <kerio> what is this dab thing [16:16:34 CET] <furq> i had to look up big room house [16:16:36 CET] <furq> Deadmau5 posted a song on SoundCloud, "DROP DA BOMB", satirizing the style of "commercial" house music and big room. [16:16:43 CET] <furq> it's not a good sign if deadmau5 can satirise you [16:16:57 CET] <kerio> furq: https://www.youtube.com/watch?v=97fFYsMbVvE [16:17:29 CET] <furq> is this a dub step [16:17:32 CET] <kerio> no [16:17:47 CET] <furq> oh wait this is trap isn't it [16:17:48 CET] <kerio> ok actually for a better representation of the effect [16:18:15 CET] <furq> or has that split into 17 subgenres that people in thick glasses with no lenses will mock you for being ignorant of [16:19:02 CET] <kerio> furq: https://www.youtube.com/watch?v=geFYnNuuURk&t=1m13s [16:19:35 CET] <furq> is the sound broken on this video [16:19:44 CET] <kerio> absolutely [16:19:55 CET] <kerio> I ABRRRRRRRRRRRRRRRRRRRRRRRRRR [16:20:07 CET] <kerio> hilarious [16:20:23 CET] <furq> sometimes the dash demuxer in firefox craps out and i get a second of fucked audio [16:20:27 CET] <furq> that sounds better than this video does [16:20:52 CET] <kerio> how does that even happen [16:20:56 CET] <kerio> not the bad audio [16:20:57 CET] <JEEB> firefox doesn't have a dash demuxer, it's all JS. it has a ISOBMFF fragment demuxer, though :P [16:20:58 CET] <kerio> that's understandable [16:21:07 CET] <kerio> but why would anyone upload a video with the audio like that [16:21:09 CET] <furq> whatever it is that constantly causes youtube to break [16:21:36 CET] <kerio> wasn't there a way to trick youtube into feeding you a mp4 or something [16:21:45 CET] <kerio> is DASH the only way now [16:22:04 CET] <furq> http only works with certain formats [16:22:13 CET] <furq> 360p and 720p iirc [16:22:23 CET] <kerio> ah yep [16:22:30 CET] <kerio> 22 mp4 1280x720 hd720 , avc1.64001F, mp4a.40.2@192k (best) [16:22:33 CET] <kerio> youtube-dl knows whats up [16:22:54 CET] <furq> firefox didn't support dash for about 18 months after youtube switched that out [16:22:58 CET] <furq> so i couldn't watch anything in 480p [16:23:00 CET] <furq> that was nice [16:23:08 CET] <kerio> why's opus 160k there [16:23:17 CET] <kerio> i thought opus was shite at high bitrates [16:23:22 CET] <kerio> you know, compared [16:23:39 CET] <furq> you thought wrong [16:24:08 CET] <furq> the aim is to be the best at any bitrate, although i've not seen any multiformat listening tests above 96k [16:24:22 CET] <furq> mostly because nobody can hear any difference at >=128k [16:24:25 CET] <kerio> yea lmao [16:24:56 CET] <kerio> Burn the Witch ends up at 1826kbps in ALAC :3 [16:25:09 CET] <furq> is that a dub step [16:25:14 CET] <kerio> ...HOW COULD YOU D; [16:25:16 CET] <kerio> radiohead [16:25:21 CET] <furq> even worse [16:25:25 CET] <kerio> fite me [16:25:50 CET] <kerio> no but seriously the latest album is a masterpiece [16:26:17 CET] <JonG> I have a question unfortunately unrelated to "trap" or a "dub step" ... I have an FFmpeg command that takes an arbitrary number of small (~10s) clips and merges them into a single larger movie. It uses filter_complex with e.g. (a)trim, (a)setpts, (a)fade on each clip, and concat, amix, volume and overlay to combine them. This all happens in a singl [16:26:17 CET] <JonG> e filter_complex. I am experiencing what I think is an out of memory error (exit code 9) for larger numbers of clips (~5-10) on less powerful devices. What is the best way for me to get around this? Is it to split the large filter_complex into multiple smaller ones that save to intermediary files? Presumably I will take a reasonable speed hit becau [16:26:17 CET] <JonG> se of multiple decode / encodings? Is it useful for me to pastebin an example command? It is generated automatically and can have reasonable variation. [16:27:21 CET] <furq> i wish i knew the answer because that opening sentence deserves credit [16:27:35 CET] <JonG> :D [16:27:45 CET] <kerio> hm [16:27:56 CET] <kerio> i wonder if it would be possible to only transcode the fading portions [16:28:02 CET] <kerio> probably not [16:28:36 CET] <c_14> JonG: check the output of dmesg [16:28:40 CET] <c_14> Is it actually being OOM-killed? [16:29:46 CET] <JonG> c_14: I am working on Android. I will have to check if I have access to dmesg. [16:30:01 CET] <c_14> JonG: might be in logcat somewhere [16:31:50 CET] <hamsheet> stuck conversion ;( frame=107146 fps= 93 q=29.0 size= 301770kB time=00:59:31.45 bitrate= 692.2kbits/s speed=3.09x (ffmpeg.exe -y -i "%A" -c:v libx264") on windows [16:32:29 CET] <hamsheet> is there a way to get more info about why it is stuck? [16:33:01 CET] <c_14> JonG: and it should return 137 if it actually got SIGKILLed, anything less than 128 is returned by the application and is not the cause of a signal [16:33:02 CET] <hamsheet> linux versions never do that but I get this kind of freeze once in a while on winx64 [16:34:16 CET] <furq> hamsheet: is %A just a regular file [16:34:28 CET] <hamsheet> yeah it is just part of batch [16:34:34 CET] <furq> nvm then [16:34:45 CET] <furq> i thought it might stand for avisynth script, in which case i'd have a pretty good idea why it froze [16:35:00 CET] <hamsheet> some are mp4 some are mov files, big files though some are [16:35:36 CET] <hamsheet> could it be multithreading? I see ffmpeg is using %100 (I have 12 cores here) [16:36:07 CET] <furq> are these the zeranoe builds [16:36:42 CET] <hamsheet> ffmpeg-20161110-872b358-win64-static [16:36:56 CET] <furq> shouldn't be an issue then [16:37:10 CET] <hamsheet> not sure who bbuild them but i got them from ffmpeg site, whatever is linked or referenced from there [16:37:13 CET] <furq> i vaguely remember some issues with builds which use win32 threads instead of pthreads, but zeranoe uses pthreads [16:37:37 CET] <hamsheet> it converts some of them them but once in a while I get freezes, for instance I got another one yesterday about 50000 frames [16:37:57 CET] <hamsheet> furq: are there other builds? [16:38:07 CET] <furq> https://ffmpeg.zeranoe.com/builds/ [16:38:11 CET] <furq> try the stable version i guess [16:39:08 CET] <hamsheet> furq: yeah maybe I should tyr that [16:39:32 CET] <hamsheet> I just enabled the debug log lets see if i can catch a line when it freezes [16:39:57 CET] <hamsheet> I cant be running out of memory too, i have 32gb here [16:40:17 CET] <hamsheet> it should crash with out of ram anyways [16:41:00 CET] <hamsheet> however I see that sometimes ffmpeg comes to a crawl converting couple frames a sec on linux too, not sure if it is a video format thing [16:44:56 CET] <jonascj> Hi all. I have two videos from a GOPRO camera (Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p, 848x480 [SAR 1:1 DAR 53:30], 30089 kb/s, 240 fps, 240 tbr, 720k tbn, 480 tbc), and I'm trying to align them to display side by side. [16:46:52 CET] <jonascj> by aligning I mean I'm trying to shift them back and forth in time until the same events occur at the same time in the two videos [16:48:10 CET] <jonascj> the two videos are of two monitors playing the same content, so I know almost down to the frame how the two should be aligned / matched. [16:48:26 CET] <kerio> yo dawg i heard you like videos [16:48:53 CET] <kerio> jonascj: open them both in mpv, click on the time display to also display the millisecond, . to advance frame by frame? [16:49:35 CET] <kerio> once you know the exact time offset, ffmpeg -ss time -i video [16:53:55 CET] <jonascj> kerio: so you would do it be cutting the videos to begin at the same time? [17:00:11 CET] <JonG> c_14: adb shell dmesg | grep "company_name": <4>[17178.134274,2] send sigkill to 19188 (k.android.company_name), adj 411, size 24858 [17:00:57 CET] <c_14> JonG: what codec are you using to encode the video output? [17:01:30 CET] <JonG> libx264 [17:05:32 CET] <c_14> That's probably what's using most of the memory, you can try using less threads or a different encoder [17:06:12 CET] <c_14> (most filters only need to keep a single raw frame at a time while the encoder will buffer several for rate-control/motion-estimation etc) [17:08:38 CET] <JonG> Is there an alternative encoder you would recommend?Related to my original suggestion, do you have any thoughts on whether it would help to split the filter_complex into segments and output intermediate files? Or not worth it? [17:09:41 CET] <c_14> any intra-frame-only codec will probably use much less memory (you can also use libx264 with a gop of 1) [17:09:52 CET] <c_14> You can try splitting the filters, but I don't think you'll save much memory there [17:10:45 CET] <kerio> jonascj: i guess so? [17:10:48 CET] <kerio> what were you planning on doing? [17:11:02 CET] <kerio> you could add some black frames at the beginning of the one that begins later [17:22:15 CET] <jonascj> kerio: I was just wondering if you could every achieve alignment on frame level by going through timestamps, maybe I had to explode both videos into individual frames, delete frames until they match, then recombine frames into video [17:50:59 CET] <kepstin> JonG: using '-preset superfast' or '-preset ultrafast' will probably cut down the mem usage quite a bit (that reduces the reference frames and lookahead), and -threads 1 could help too [18:03:26 CET] <JonG> kepstin: OK, thanks, I'll take a look at that too. I was just experimenting with libx264 and a gop of 1. [18:07:42 CET] <ksk> kerio: thanks! [18:08:06 CET] <kerio> wait what did i do [18:09:28 CET] <kerio> furq why didn't you tell me about ffvhuff [18:10:27 CET] <jonascj> Is there a way to extract all frames without fretting about whether some frames gets skipped or duplicated? E.g. I worry that r=240 will miss some frames despite ffmpeg reports the video as 240 fps. [18:10:35 CET] <jonascj> * "-r 240" [18:11:56 CET] <kerio> it's noticeably bigger than ffv1 but not excessively so [18:12:01 CET] <kerio> and it's... mindboggingly fast [18:15:09 CET] <kerio> is there a documentation page for ffvhuff? [18:15:15 CET] <kerio> in the wiki or whatever [18:34:43 CET] <furq> kerio: it's even less compatible than ffv1 [18:35:04 CET] <furq> and it's quite a lot bigger in my experience [18:35:12 CET] <furq> it is the fastest by some distance though [18:35:36 CET] <kerio> it's the fastest by an order of magnitude [18:35:41 CET] <kerio> at least on my gray16 videos [18:35:52 CET] <furq> it was about 1.5x the speed of huffyuv last i checked [18:35:56 CET] <furq> although i didn't test gray16 [18:36:16 CET] <furq> if you only ever need to decode it with ffmpeg then it's a good choice if you just need speed [18:41:08 CET] <kerio> furq: i mean, i'm still getting insane space reductions [18:41:10 CET] <kerio> like, over half [18:41:17 CET] <kerio> my video is gray16 but with very little dynamic range [19:28:03 CET] <durandal_1707> kerio: tried utvideo? [19:31:10 CET] <kerio> durandal_1707: it doesn't appear to be able to do gray16 :\ [20:13:07 CET] <t4c0c4t> are there any quick optimizations that can be made with nvenc (GPU) like -threads on libx264? [20:33:20 CET] <DHE> t4c0c4t: not really. you can change the preset to a number of options. high quality, low quality, and multi-pass are available with obvious performance tradeoffs [21:21:38 CET] <MrMonkey31> yo. does any1 here know, I'm going for a hypothetical 5th pass using libx265 in ffmpeg. but my 4th pass command never ran. what happens then? [21:22:22 CET] <MrMonkey31> would it treat the current command as just a generic "next pass"? would it complain that the previous pass statistic file thingy is missing? or...? [21:22:44 CET] <llogan> you should do 40 passes [21:22:59 CET] <MrMonkey31> I have been illuminated .00004% [21:24:08 CET] <MrMonkey31> my actual question is- would it matter by what mechanism the prior pass was omitted? maybe the command never ran, or maybe it ran and the station lost power, or perhaps it ran but someone pressed q before it finished [21:24:38 CET] <ibisr> Is there a way with -ss and -to args to specify a specific pts rather than seconds or timecode? [21:25:26 CET] <MrMonkey31> [I know that with h264 there's a couple temp files created, but h265 normally I only find one file containing 0 bytes] [21:25:53 CET] <Pandela> Afternoon [21:26:54 CET] <Pandela> Is it possible to output whats playing in ffplay into an avi/mp4 container? [21:27:47 CET] <furq> ibisr: if you're reencoding you can do it with the select filter [21:27:57 CET] <furq> and probably some other filters [21:28:33 CET] <furq> Pandela: no, but you can pipe from ffmpeg into ffplay and save the file at the same time [21:30:06 CET] <Pandela> furq: Damn...I don't think thats exactly what I need because I'm applying different filters to the udp stream, and another filter upon recieving the stream in ffplay. Wanted to render that result [21:30:17 CET] <BtbN> libx265 can do more than 2 pass encoding? oO [21:30:23 CET] <BtbN> what's the gain of that? [21:33:59 CET] <DHE> in theory the 2-pass stats get more "refined" with a second pass. I don't think it matters much unless you need "maximum quality with an exact target bitrate requirement" [21:34:05 CET] <DHE> can't imagine why that's such a big deal though [21:35:10 CET] <MrMonkey31> guys, my question only pertains to what happens if my encoding was interrupted? how do I know for example that the 2nd pass is a real 2nd pass, and not just ffmpeg ignoring the fact that there is no data from the first pass even coming in? [21:36:06 CET] <MrMonkey31> the behavior I have seen with libx264 is that a pair of (stat?) files get made during the 1st pass, then they get renamed when that pass completes [21:36:45 CET] <MrMonkey31> it's all very mysterioso, and I don't care to unravel it. I only care that my final encode is optimal [21:39:05 CET] <DHE> well libx264 will log a warning if the 2nd pass is longer than the first, and error out entirely if there is no logfile for -pass 2 [21:42:00 CET] <MrMonkey31> DHE ah ok that's what I'm looking for but for h265? I also observed that the log file name has "pass-0" in it or something to that effect, but that number never does increase (I had been testing a 4 pass thing when I first learned about h265) [21:42:08 CET] <MrMonkey31> ahem about h264 rather [21:43:19 CET] <MrMonkey31> so I'm at pretty low confidence, and thinking about abandoning 2 pass encodes entirely. hell, it saves our coal right? [21:44:43 CET] <MrMonkey31> h265 has no log file as I have been told, and frankly I defy anyone to learn this answer from searching the web in under 20 solid mins of searching [21:45:19 CET] <DHE> depends on your needs. 2-pass is to get an exact file size target [21:46:08 CET] <MrMonkey31> well - the only thing I haven't tried, because I thought I'd ask around first, is to simply run a command [21:47:18 CET] <MrMonkey31> that would be the 3rd command in the series that pertains to the question: the first two being for a 1st pass and which completed, and the 2nd for a second pass which did not complete. I was afraid the wrong thing would destroy the progress from an 8 hour 1st pass run [21:47:56 CET] <MrMonkey31> maybe running the 2nd command once more will resume as expected and desired, or maybe not. that's the dilemma [21:48:30 CET] <MrMonkey31> so that is what I shall do, and if it garners anything useful I will report [21:48:46 CET] <llogan> you could use -t (and optionally with -ss) to encode a short segment for testing [21:49:14 CET] <ibisr> furq: That sure looks like it might do what I want, thanks! [21:49:16 CET] <DHE> what's the goal? do you have a bitrate constraint? a file size constraint? do you not care and just need it to look good? [21:50:03 CET] <Pandela> Have any of you been able to pick up an ffmpeg stream in say, Manycams IP camera freature? [21:50:21 CET] <Pandela> *feature [21:54:05 CET] <MrMonkey31> DHE, right now the goal is to determine what to do when a 2 pass encode fails during the 2nd pass, when using libx265 [21:54:50 CET] <MrMonkey31> I've run the command for the 2nd pass and got this message: "Error opening file ffmpeg2pass-0.log. / Error opening file ffmpeg2pass-0.log." [21:55:19 CET] <MrMonkey31> then I created an empty file with that name, and tried again. it commenced encodeing without any complaints [21:55:46 CET] <MrMonkey31> (-yet) [21:56:17 CET] <MrMonkey31> so I'm not prepared to wait for this thing to complete right now, and will cancel it with ctrl-c as I had done last time [21:57:48 CET] <MrMonkey31> I believe that with libx264 - which I am not using in this - that quitting the 2nd pass will destroy the data from the 1st pass, by replacing the log files from that pass with the partial logs created during the 2nd pass [21:57:52 CET] <llogan> you didn't really answer his questions. your answers may reveal that you are wasting your time with multiple passes [21:58:33 CET] <MrMonkey31> llogan, if you worry about other folks wasting their time, wouldn't that sorta be... a waste of your time? [22:00:54 CET] <MrMonkey31> I just desire to know that my 1st pass is indeed being used. the presence of temp files would give such knowledge, but libx265 denies us that. were I more capable, I could find out for myself where it tucks the temporary data from the 1st pass, and that would effectively also be of use [22:00:55 CET] <DHE> MrMonkey31: no no. I'm asking this to determine if 2-pass video encoding makes sense in the first place [22:01:03 CET] <DHE> ah, llogan already clarified [22:01:18 CET] <MrMonkey31> I am NOT a troll, goddamit! [22:01:28 CET] Action: MrMonkey31 lurches about comically, with arms bent [22:01:45 CET] <MrMonkey31> well I asked the #x265 folks first [22:01:59 CET] <MrMonkey31> but they are all preoccupied [22:02:18 CET] <MrMonkey31> DHE, I'm sure it makes sufficient sense for my purposes, or I would not seek to use it to begin with. [22:02:48 CET] <unkmar> I have difficulties I'm attempting to solve. How in depth should I go in order to ask my questions? [22:02:58 CET] <DHE> not judging. a lot of people don't know what the options are [22:03:31 CET] <MrMonkey31> I literally seek data. temporary data. which this program has made [22:03:37 CET] <unkmar> Disclainer: Not able to test at the moment. On work PC and Home PC is where my difficulties are. [22:04:26 CET] <llogan> the command and resulting complete console output (provided via a pastebin link), and a general description of the issue is usually sufficient. [22:06:04 CET] <unkmar> In this case, the command and resulting console output may be irrelevant. I may be tackling the problem wrong. [22:06:26 CET] <MrMonkey31> unkmar, how's the weather in Mordor right now? [22:08:11 CET] <unkmar> Overview: I'm a streamer that wishes to have and change scenes without stream interruption, I'd also like to provide dynamic info, Such as latest detected follower. [22:09:35 CET] <llogan> unkmar: zmq may be of use https://ffmpeg.org/ffmpeg-filters.html#zmq_002c-azmq [22:09:40 CET] <unkmar> I already have all the audio and video bitrate and encoding formats figured out. [22:10:47 CET] <unkmar> llogan: Thank you. That likely will help by many magnitudes. [22:11:05 CET] <unkmar> I had been fighting for hours to do things the wrong way. [22:11:06 CET] Action: MrMonkey31 grumbes [22:14:57 CET] <unkmar> MrMonkey31: Smoky overcast with a small chance of showers. [22:21:25 CET] <MrMonkey31> well, grab your lembas, folks, we're going on a hike! [22:31:30 CET] <MrMonkey31> well, ty all for reading, I guess this is rly a question for somebody familiar with the vagaries of libx265 particularly. I'm content to bypass it in future altogether, by using h264 for instance, or just stick to the crf method [22:37:50 CET] <Chloe[m]> can you compile ffmpeg without msys or cygwin on windows? using only mingw [22:42:38 CET] <unkmar> Chloe[m]: I don't usually use Windows or compile. So, I can't help. [23:06:46 CET] <blb> Chloe[m]: it might be weird, but i prefer using the environment setup by the "git for windows sdk" [23:07:05 CET] <blb> afaik, it's mingw/msys [23:07:25 CET] <furq> that's msys [23:08:24 CET] <furq> i imagine it's possible to do it without msys but it's certainly not supported or recommended [23:08:29 CET] <drv> you need some kind of posix shell to run configure, so "no" is the short version of the answer [23:08:44 CET] <furq> you'd need to replicate everything done by configure by hand, and that sounds terrible [23:08:56 CET] <Chloe[m]> ok, thanks :( [23:09:08 CET] <furq> i normally just cross-compile [23:09:38 CET] <blb> i use that to build with msvc, which gives you symbols 'n stuff [23:11:14 CET] <furq> msys2 used to be an absolute nightmare but apparently it's a lot better now [23:11:24 CET] <furq> they claim to have fixed the pacman behaviour that constantly broke everything [23:18:02 CET] <unkmar> pacman behaviour? [23:41:42 CET] <Chloe[m]> can I disable all video codecs categorically? [23:43:32 CET] <furq> --disable-encoders --disable-decoders [23:43:53 CET] <DHE> ./configure --disable-decoders --disable-encoders --enable-decoders=pcm*,aac*,theora,mpeg*,h264 --enable-encoder=libx264,libx265 [23:44:04 CET] <DHE> for a very limited decoder (and encoder) set [00:00:00 CET] --- Wed Nov 16 2016

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