October 2018 - Ffmpeg-devel-irc

ffmpeg-devel.log.20181030
by burek 31 Oct '18

31 Oct '18

[03:14:06 CET] <cone-150> ffmpeg 03Michael Niedermayer 07master:0fb83b4c91d5: avcodec/vp56: Add vpX_rac_is_end() to check for the end of input [03:14:07 CET] <cone-150> ffmpeg 03Michael Niedermayer 07master:78862488f852: avcodec/vp9: Check in decode_tiles() if there is data remaining [04:13:58 CET] <cone-150> ffmpeg 03Charles Liu 07master:3d1b7954933b: avformat/hlsenc: fix the duration of m4s segment is unusually smaller than expected. [04:13:59 CET] <cone-150> ffmpeg 03Charles Liu 07master:e9dbd62cb5f1: avformat/hlsenc.c: fix memory leak in fmp4 mode. [04:14:00 CET] <cone-150> ffmpeg 03Charles Liu 07master:2365f47bf51a: avformat/hlsenc.c: remove the useless variable fmp4_init_mode. [04:14:01 CET] <cone-150> ffmpeg 03Charles Liu 07master:76b8e42c1f04: avformat/hlsenc.c: the size of init.mp4 is zero. [04:14:02 CET] <cone-150> ffmpeg 03Charles Liu 07master:1ff4bd59dfce: avformat/hlsenc.c: fix the output's duration smaller than input's in sub-range mode. [05:17:42 CET] <slavanap> Hi! Is there a simple guide for developing image filters for ffmpeg? [06:18:27 CET] <cone-150> ffmpeg 03Jun Zhao 07master:903f2beafc7c: lavc/decode: Fix the error number report if av_image_fill_pointers fail. [06:18:28 CET] <cone-150> ffmpeg 03Jun Zhao 07master:f3bcb9c16a42: lavu/frame: Add error report if av_image_fill_pointers fail. [07:12:02 CET] <cone-150> ffmpeg 03James Zern 07master:32d021cfa652: avcodec/libvpxdec: fix setting auto threads [10:10:33 CET] <superware> any idea why avio_open2(&_output_ctx->pb, "rtsp://127.0.0.1:8854/live.sdp", AVIO_FLAG_WRITE, null, null) fails with -1330794744? [10:12:45 CET] <funman> #define AVERROR_PROTOCOL_NOT_FOUND FFERRTAG(0xF8,'P','R','O') ///< Protocol not found [10:14:57 CET] <superware> I'm using FFmpeg 4.0.1, shouldn't RTSP be supported? I tried calling av_register_all, avdevice_register_all and avformat_network_init (even if deprecated) [10:16:32 CET] <nevcairiel> maybe it wasnt compiled in [10:30:15 CET] <superware> nevcairiel: ok, if I try "rtp://127.0.0.1:1234" instead rtsp then avio_open2 is successful, but then av_interleaved_write_frame(_output_ctx, pkt) fails catastrophically by writing protected memory.. [11:54:22 CET] <atomnuker> BBB: what's with the videolan gitlab being more nazi than a gestapo officer? [11:54:34 CET] <BBB> huh what? [11:54:43 CET] <BBB> thats a little harsh [11:54:57 CET] <BBB> can you be more specific? [11:55:01 CET] <atomnuker> logs me out after 2 days, 2fa required, have to dig my phone out, find out it has no battery, 2fa doesn't work, have to resync time to google servers [11:55:20 CET] <j-b> there are numerous 2fa apps for cli and so on [11:55:24 CET] <BBB> lol [11:55:41 CET] <BBB> ok, sorry, I dont know about that specifically, but I actually think the focus on security is a good thing [11:55:53 CET] <BBB> yes, phone being out of juice is annoying [11:56:21 CET] <j-b> it's just normal TOTP [11:57:02 CET] <durandal_1707> atomnuker: if it were only videolan gitlab.... [11:57:07 CET] <atomnuker> security's fine, just forbid web pushing and make everyone do git pull <repo>; git push using ssh, as a bonus this gets rid of merge commits [11:57:52 CET] <atomnuker> (to master branches anyway) [11:59:33 CET] <tguillem> getting rid of merge commits is not a gitlab bug, we do it on purpose. Linear master branch vs branch with merge commit. We did a choice, both are fine. [11:59:58 CET] <atomnuker> no, merge commits suck is what I meant [12:00:11 CET] <tguillem> I logged once to gitalib, put my .ssh key on it, and don't have to relog to pull/push [12:01:02 CET] <atomnuker> well but you do have to relog to make prs editable by maintainers (which I do very clearly remember doing last time) [12:01:15 CET] <atomnuker> and you have to be logged in to make a pr or post a comment [12:01:33 CET] <j-b> This is a configuration for each project, nothing to do with gitlab [12:01:38 CET] <j-b> Maintainers can push directly [12:02:44 CET] <atomnuker> BBB: "Allow commits from members who can merge to the target branch." <- it was ticked when I opened the webpage [12:08:33 CET] <BBB> atomnuker: huh, weird& I couldnt rebase it [12:09:15 CET] <BBB> Fast-forward merge is not possible. To merge this request, first rebase locally. [12:09:22 CET] <BBB> maybe there is a merge conflict? [12:09:26 CET] <BBB> can you rebase? [12:40:19 CET] <atomnuker> ah, yes, forgot about that commit, rebased [15:22:23 CET] <nevcairiel> whats the latest easiest method to activate the automatic hwaccel decoding? Set a hwdevice context and stuff just magically starts working? [15:22:50 CET] <nevcairiel> docs seem a bit lacking =p [16:10:14 CET] <dualz> So I wanted to be able to change inputs at run-time, so i figured I'd make a filter for it that accepts commands via zmq [16:10:44 CET] <dualz> I copied the movie filter and used that as a base, named my filter inputswap [16:11:47 CET] <dualz> I added the "swap" command to the filter [16:11:50 CET] <dualz> https://gist.github.com/ryanmarin/a4bb8d3d2241b827cff4611e3deca503 [16:12:34 CET] <dualz> now it's supposed to close the current input and open a new one [16:13:00 CET] <dualz> from my log output it looks like it opens the new input fine [16:13:33 CET] <dualz> but I get a segfault [16:15:45 CET] <dualz> https://gist.github.com/ryanmarin/a960a2d36bbb1af744f95969915f5ea6 [16:16:58 CET] <dualz> any insight as to what's going on would be appreciated [16:20:38 CET] <BtbN> I highly doubt the filter graph can be changed at runtime [16:21:09 CET] <BtbN> Also, use the ffmpeg native filter command framework, pulling in another complex library for that won't find very enthusisastic reception [16:21:46 CET] <thardin> changing filters kind of implies nle, for which ffmpeg is woefully unsuited [16:22:04 CET] <thardin> ah, at run-time even. whooh [16:22:21 CET] <thardin> wait this is what OBS is for [16:25:46 CET] <dualz> I've been trying to wrap my brain on how to go about this, basically here's the problem space [16:27:10 CET] <dualz> I have an RTMP server that ingests rtmp://live/cam_1,2,3,4 [16:27:54 CET] <dualz> at the beginning of the stream perhaps only cam 1 is connected [16:28:28 CET] <dualz> and it's not a few moments in, until cam 2 connects [16:28:57 CET] <dualz> but the stream started already with just one cam connected [16:30:29 CET] <dualz> The user has the ability to change scenes, add overlays etc (which I can at runtime with zmq) [16:31:14 CET] <thardin> still sounds like an OBS job [16:31:14 CET] <dualz> if they go to a scene that shows cam 2 [16:32:01 CET] <dualz> i'd like it to be test bars around their input [16:38:49 CET] <dualz> I don't think I'm explaining it right if it sounds like an OBS job [17:37:50 CET] <philipl> BtbN: https://github.com/philipl/FFmpeg/commit/a797f404882cea58a8c039b65aa50a1ac7… [17:37:53 CET] <philipl> A change from nvidia. [17:39:09 CET] <nevcairiel> who even still uses vdpau if you can use nvdec? [17:40:44 CET] <JEEB> some people seem to be holding to it for some reason [17:41:52 CET] <j-b> old distro? [17:42:39 CET] <atomnuker> the worst are people who still use vdpau with the vaapi backend wrapper it has [17:43:43 CET] <atomnuker> those that use vaapi with the vdpau wrapper it has are maybe just as bad [17:56:07 CET] <philipl> I don't understand what nvidia are up to. Clearly people don't talk to each other. [17:56:20 CET] <nevcairiel> its a company, they never do [17:56:26 CET] <philipl> They've got a vdpau maintainer (the guy who wrote this patch) but his remit is clearly just rearranging deckchairs. [17:57:12 CET] <philipl> He's trying to fix the fact that interlacing is built into the GL interop spec, which I guess is nice, but no movement on adding the missing decoder formats or removing the GLX dependency. [17:58:59 CET] <philipl> but anyway, that patch is fair enough and I want to apply it. [17:59:21 CET] <philipl> Poor guy put a bunch of effort into fixing the underlying driver bug, so it's the least I can do :-) [19:36:44 CET] <BtbN> philipl, well, it's something I guess [19:37:05 CET] <BtbN> The patch looks sensible to me [19:37:34 CET] <BtbN> I'd maybe also change the log message, to indicate that it works in recent drivers, but otherwise, simple enough [19:49:29 CET] <philipl> Ok. I will adjust and then push [19:57:42 CET] <BtbN> I'm not aware of being the maintainer for anything vdpau? Not sure who has to "officially" okay that, but I doubt it's an issue. [19:58:15 CET] <BtbN> Also... the commit message says "Have been addressed in Nvidia driver releases 384.59 and 410.66" [19:58:25 CET] <BtbN> But the check only checks 410 [19:58:32 CET] <nevcairiel> it talks about two distinct issues [19:59:14 CET] <nevcairiel> issues (1) and (2) have been fixed in drivers A and B respectively [20:41:57 CET] <atomnuker> well I'm glad intel added vpsllvw to do variable shifts so now you don't have to hack around using pmulhsrw [20:42:16 CET] <atomnuker> but its avx512, and no one has avx512 yet [20:42:42 CET] <durandal_1707> how so? [20:43:37 CET] <atomnuker> well they change chipsets every time they feel like it, new CPUs are much more expensive for not much gain and you usually can't upgrade a laptop's cpu [20:44:59 CET] <durandal_1707> buy new laptop, you have a well paid job [20:46:03 CET] <atomnuker> I'd rather get a big machine, with a new amd gpu [20:58:35 CET] <atomnuker> AAARgh pmulhsrw you betray me [20:59:20 CET] <BtbN> Do any intel CPUs outside of their Xeons have it at all yet? [20:59:25 CET] <atomnuker> cdf probabilities are between (1 << 14) and (1 << 15), and the shifting pmulhsrw trick only works for numbers less than 1 << 14 [20:59:36 CET] <BtbN> Also, AMD GPUs sadly all aren't that great [21:00:17 CET] <atomnuker> I know, they're slower and more expensive, but they do work with wayland [21:01:06 CET] <philipl> BtbN: apparently I am listed as a maintainer. I guess I asked you out of habit :-) [21:01:10 CET] <philipl> It is nvidia related after all. [21:02:00 CET] <philipl> jkqxz: (or anyone else who knows). Has AMD officially gone all in on vaapi vs vdpau? [21:06:38 CET] <BtbN> philipl, I think it depends on the driver in use [21:06:45 CET] <BtbN> iirc they also have a proprietary API? [21:06:56 CET] <BtbN> for decoding I mean [21:07:01 CET] <BtbN> I'm aware of amf [21:07:39 CET] <JEEB> philipl: they have VAAPI support officially [21:07:55 CET] <JEEB> and they even seemed to have what was required for interop (no RAM copying needed) with opengl [21:08:00 CET] <JEEB> added into vaapi [21:08:05 CET] <JEEB> so they definitely are in the vaapi camp [21:08:09 CET] <JEEB> (if you're talking about AMD) [21:08:37 CET] <JEEB> and with windows they support dxva2 and d3d11, which means that their AMF thingamajig then can receive the surfaces received from those hwdecs [22:01:41 CET] <philipl> JEEB: the amf guy was around last week trying to get some traction on reviewing and merging his last set of changes. [22:01:50 CET] <philipl> Seemed a thoroughly thankless task. [22:56:32 CET] <durandal_1707> what you guys think about adding bayer 10/12/14 formats support to swscale?, in similar way how is currently done, also changing 16bit bayer to output 16bit format [23:00:41 CET] <jamrial> if some codec can output that, then sure [23:01:16 CET] <durandal_1707> MLV and TIFF have it [23:04:16 CET] <atomnuker> sounds fine [23:18:22 CET] <cone-547> ffmpeg 03Werner Robitza 07master:ad5ca1fb72fc: doc/hls: fix grammar for HLS options [23:20:42 CET] <durandal_1707> so you are ok that for bayer to any other pixel format - it goes to rgb first and than to that format? [23:22:16 CET] <nevcairiel> sure, bayer is basically gimped rgb [23:51:19 CET] <BBB> atomnuker: feel like addressing the 2 comments on the MR? [23:52:26 CET] <atomnuker> BBB: I did that yesterday, didn't I? [23:52:33 CET] <BBB> theres 2 more [23:52:47 CET] <atomnuker> oh, right [23:57:18 CET] <atomnuker> k done [23:58:08 CET] <atomnuker> the (-(cdf[i] - 32768)) >> rate; thing was because that's what my simd code does, and I thought it'd be easier for compilers [23:58:17 CET] <atomnuker> to vectorize [23:58:26 CET] <atomnuker> though I don't think it made any difference [23:59:15 CET] <atomnuker> after all, its a load (to load 32768 in a reg per iteration) vs a mult (by -1 and using 32768 x 8 in an always loaded reg) [23:59:52 CET] <atomnuker> I gave up on the idea to iterate and just always do all 16 max cdf symbols at a time with 256bit regs [00:00:00 CET] --- Wed Oct 31 2018

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ffmpeg.log.20181030
by burek 31 Oct '18

31 Oct '18

[00:51:06 CET] <flying_sausages> ariyasu, turns out that becuase I was using a loop which read nextlines from a multiline variable via stdin, this interfered with ffmpeg. I'm on WSL, is this bug-report worthy to Microsoft or to ffmpeg? [03:24:54 CET] <raytiley> anyone know where to lookup the h264_qsv warning / error codes. Getting this message a lot and not sure what setting to tweak: `Warning during encoding: incompatible video parameters (5)` [03:32:25 CET] <oterrivel> hello [03:34:28 CET] <abi> hello [03:35:28 CET] <oterrivel> some help here with wget? trying to download segment-0.mp4 to Output file Academy-video.mp4 and append to it another exec of wget with input file {segments-1.m4s to segments-690.m4s}. FFMPEG could not find track id 1 [03:44:20 CET] <oterrivel> no? [09:52:43 CET] <Matador> so weird [09:56:19 CET] <Matador> anyone around to deal see why ffmpeg doesnt want to output to flv/rtmp ? [09:56:22 CET] <Matador> just blanks out, so weird [10:08:22 CET] <superware> any idea why avio_open2(&_output_ctx->pb, "rtsp://127.0.0.1:8854/live.sdp", AVIO_FLAG_WRITE, null, null) fails with -1330794744? [10:29:35 CET] <MarcoT> Hello, can you help me with my problem? I am writing a decoder that reads packets from memory (using AVIO context and AVFormat) and decode them. But I have the following errors, and some frames result corrupted in the video player [10:29:45 CET] <MarcoT> [mpeg1video @ 0x562750178f00] ac-tex damaged at 2 9 [mpeg1video @ 0x562750178f00] Warning MVs not available [mpeg1video @ 0x562750178f00] concealing 887 DC, 887 AC, 887 MV errors in I frame [10:30:15 CET] <MarcoT> Sometimes I have also overread error [11:27:27 CET] <MarcoT> Hello, can you help me with my problem? I am writing a decoder that reads packets from memory (using AVIO context and AVFormat) and decode them. But I have the following errors, and some frames result corrupted in the video player [mpeg1video @ 0x562750178f00] ac-tex damaged at 2 9 [mpeg1video @ 0x562750178f00] Warning MVs not available [mpeg1video @ 0x562750178f00] concealing 887 DC, 887 AC, 887 MV errors in I frame [11:27:42 CET] <MarcoT> Sometimes I have also overread error [11:28:18 CET] <durandal_1707> does same happens with ffmpeg cli tool? [13:54:35 CET] <unlord> Sup my video friends [13:54:46 CET] <unlord> I want to crop a 12-bit 444 y4m file I have losslessly [13:54:59 CET] <unlord> does your fine application do this? [13:56:25 CET] <BtbN> If you don't use a lossy codec, there won't be any quality loss [13:59:02 CET] <durandal_1707> unlord: see crop filter [13:59:32 CET] <unlord> BtbN: y4m is lossless [13:59:42 CET] <BtbN> The input doesn't overly matter [13:59:43 CET] <unlord> durandal_1707: I'm pretty new to ffmpeg, where is the crop filter? [14:09:10 CET] <azarus> "where": in ffmpeg [14:09:34 CET] <azarus> there's no clicky-clicky fancy GUI [14:16:40 CET] <durandal_1707> unlord: ffmpeg -h filter=crop also read documentation [14:18:22 CET] <unlord> durandal_1707: I'm already done, but thanks [14:18:31 CET] <unlord> ffmpeg -loglevel debug -i ducks_take_off_1080p50_60f.y4m -vf "crop=640:360:100:100" ducks-crop.y4m [14:33:43 CET] <shroomM> hey guys [14:34:14 CET] <shroomM> i'm having some issues encoding audio to AAC in a fragmented mp4 format and was wondering if anyone encountered anything similiar [14:34:42 CET] <shroomM> it seems to be related to me piping ffmpeg's output to stdout instead of directly to a file [14:35:24 CET] <shroomM> I've got two slightly different CLIs... [14:35:30 CET] <shroomM> ffmpeg -i tos.mp4 -c:a aac -b:a 128000 -r:a 44100 -ac 2 -movflags frag_keyframe+empty_moov -vn -f mp4 -y - >piped.mp4 [14:35:31 CET] <shroomM> and [14:35:46 CET] <shroomM> ffmpeg -i tos.mp4 -c:a aac -b:a 128000 -r:a 44100 -ac 2 -movflags frag_keyframe+empty_moov -vn -f mp4 -y non-piped.mp4 [14:37:07 CET] <shroomM> the piped.mp4 does not play in VLC and/or quicktime, and macOS' finder quicklook [14:37:47 CET] <shroomM> also when running "ffmpeg -i piped.mp4", I notice the following warning in ffmpeg's output... [aac @ 0x7f9dc5011e00] Number of bands (33) exceeds limit (22). [14:38:04 CET] <shroomM> the "non-piped.mp4" works OK and does not produce this warning [14:38:12 CET] <shroomM> any ideas? [14:40:53 CET] <shroomM> examining stderr of the encoding process, there is really no differenc between invocations [14:45:35 CET] <shroomM> initially tested with 4.0.2, but just tried with a static build from zeranoe from git master (g32d021cfa6) [14:45:42 CET] <shroomM> and the same issue persist [15:05:07 CET] <unlord> uhh [15:05:09 CET] <unlord> what? [15:05:11 CET] <unlord> [yuv4mpegpipe @ 0x5630a8ce6700] 'yuv444p10le' is not an official yuv4mpegpipe pixel format. [15:05:43 CET] <unlord> I gave it a y4m with C444p10 [15:05:50 CET] <shroomM> hmm, so the issue seems to be that I'm producing an audio only mp4, but I'm specifying only frag_keyframes. If I remove the frag_keyframe and add -frag_duration 1000000, the resulting file works [15:06:27 CET] <unlord> oh, n/m [16:48:42 CET] <mertyildiran> how can I pipe the output(the video itself) of this command in real time to a video player? "ffmpeg -y -f rawvideo -s 1920x1080 -pix_fmt rgba -r 60 -i - -an -loglevel error -vcodec libx264 -pix_fmt yuv420p ./MyVideo.mp4" [17:08:43 CET] <pzy> instead of saving it as a file, or in addition to? [17:16:56 CET] <kepstin> if you want "instead of" then the obvious thing is to remove ffmpeg completely and pipe the original raw video to the player. [17:42:47 CET] <mertyildiran> video is generated on the fly, careful that the input is also piped in [17:43:03 CET] <mertyildiran> I'm reading this guide https://trac.ffmpeg.org/wiki/StreamingGuide [17:44:43 CET] <mertyildiran> from my understanding I should use mpegts with some sort of configuration [17:44:53 CET] <mertyildiran> like this ffmpeg -y -re -f mpegts -s 1920x1080 -pix_fmt rgba -r 60 -i - -an -loglevel error -vcodec libx264 -pix_fmt yuv420p - | ffplay [17:44:56 CET] <mertyildiran> or this ffmpeg -y -re -f mpegts -s 1920x1080 -pix_fmt rgba -r 60 -i - -an -loglevel error -vcodec libx264 -pix_fmt yuv420p - | ffplay [17:45:25 CET] <mertyildiran> but it gives me "max resync size reached, could not find sync byte" error and I don't know what it means [17:46:11 CET] <mertyildiran> fixing or this ffmpeg -y -re -f mpegts -s 1920x1080 -pix_fmt rgba -r 60 -i - -an -loglevel error -vcodec libx264 -pix_fmt yuv420p tcp://127.0.0.1:2000 [17:55:26 CET] <King_DuckZ> hi, I'm struggling with the frame to pts estimation again :s [17:56:17 CET] <King_DuckZ> my understanding was that pts would advance in steps of more or less rate.num / timebase.den [17:56:56 CET] <King_DuckZ> so for example I might have a pts sequence like this: 0, 33, 66, 100, 133.... [17:57:41 CET] <King_DuckZ> and in fact my test video's rate was 30000/1001 and timebase 1/1000, so 30000/1000 = 30 [17:58:28 CET] <King_DuckZ> but now I have rate 24/1 and timebase 1/24000, so 24/24000... nope! :( [18:04:39 CET] <ChocolateArmpits> King_DuckZ, why not pick a higher common timebase? maybe 90000? [18:05:30 CET] <King_DuckZ> ChocolateArmpits: what do you mean 'pick'? I'm getting those values from ffmpeg [18:05:53 CET] <ChocolateArmpits> oh [18:06:25 CET] <King_DuckZ> basically I want to know how far apart 2 adjacent pts are going to be *more or less* [18:07:12 CET] <King_DuckZ> by looking at the debug messages in my code I think in my second case the answer is 1000 [00:00:00 CET] --- Wed Oct 31 2018

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ffmpeg-devel.log.20181029
by burek 30 Oct '18

30 Oct '18

[02:24:54 CET] <JC_Yang> do anyone strive to make ffmpeg build with -Wall -Werror cleanly? don't you think there might be hidden bugs due to the warnings spreading in every corners? [02:31:47 CET] <qeed> hard bugs are logic bugs and bugs cant be detected from static analysis [02:53:41 CET] <JC_Yang> yes, probably, but why not first make it warnings clean? is this excuse strong enough to support the left-warning-as-is decision? I encounter a weird bug recently, in the matroska muxer. I implement my own filters to just feed valid packets(with valid dts/pts, all of them are guaranteed to be >0 while as input) to the muxer, but, ocassionally the muxer run into a warning saying "Starting new cluster due to timestamp". I modify the codes to also [02:53:41 CET] <JC_Yang> print out the dts/pts and all locally relate timestamp info, I'm shocked to see that the dts/pts are negative numbers!!! I am not familiar to the codebase, I don't know a reliably and easy way to trigger the bug, and considering the -Wsign-compare and alike warnings spreading everywhere, I doubt... [02:55:48 CET] <BBB> not all warnings are useful [02:55:56 CET] <BBB> some are plain wrong [02:56:06 CET] <BBB> and can only be fixed by code obfuscation or things like that [02:56:20 CET] <BBB> ohwell [02:57:19 CET] <BBB> JC_Yang: negative timestamps are common if you literally interpret some media formats [02:57:28 CET] <BBB> if you dont want it, just subtract the first ts from all following [02:57:29 CET] <BBB> simple [02:57:48 CET] <BBB> most people dont care, they just care about video and audio alignign [03:10:22 CET] <kepstin> also, "starting new cluster" in mkv is normal, clusters have a timestamp, then blocks within a cluster use an offset from the cluster timestamp, and any time that offset would overflow it has to start a new cluster. [03:10:27 CET] <kepstin> iirc? [03:11:37 CET] <kepstin> i wish the mkv spec was more readable :/ [09:32:38 CET] <j-b> 'morning [12:05:17 CET] <J_Darnley> Following on from my question last week about a strobe... [12:05:45 CET] <J_Darnley> What filter can number each frame? drawtest? [12:05:47 CET] <J_Darnley> uh [12:05:51 CET] <J_Darnley> drawtext [14:05:34 CET] <January> jamrial: https://0x0.st/s6n0.patch this look ok? Will send to ML [14:49:18 CET] <nevcairiel> January: while you're in there, did you also s ee https://trac.ffmpeg.org/ticket/7521 ? [14:49:49 CET] <nevcairiel> ssize_t is apparently not quite portable [14:50:25 CET] <January> nevcairiel: do we not have a helper for this somewhere? [14:51:45 CET] <nevcairiel> no, ssize_t is practically unused in ffmpeg, outside of a few platform specfic modules which probably just dont get used anywhere else [14:52:39 CET] <nevcairiel> fread by definition returns size_t though [14:52:40 CET] <jkqxz> The return value of fread() is not ssize_t, anyway. The < 0 check is also wrong. [14:52:54 CET] <jkqxz> Yeah, that. [14:55:55 CET] <January> right I was previously just using read(2), this should fix MSVC then https://0x0.st/s65m.patch [14:56:39 CET] <nevcairiel> seems fine [14:56:48 CET] <nevcairiel> (on that aspect) [16:46:47 CET] <cone-772> ffmpeg 03Paul B Mahol 07master:7e1add2c51d7: doc/filters: add small description to geq filter section [17:02:50 CET] <cone-772> ffmpeg 03Michael Niedermayer 07master:88e3807aafd3: avcodec/vp3: Do not initialize unused tables for keyframes in unpack_superblock() [17:02:51 CET] <cone-772> ffmpeg 03Michael Niedermayer 07master:f56318081777: avcodec/vp3: Reuse local variable in unpack_superblocks() [17:02:52 CET] <cone-772> ffmpeg 03Michael Niedermayer 07master:b5e7e437f4c8: avcodec/vp3: Do not recalculate coded_fragment_list for keyframes [17:02:53 CET] <cone-772> ffmpeg 03Michael Niedermayer 07master:4885ff663bec: avcodec/vp3: reindent unpack_superblocks() [20:07:31 CET] <haasn> atomnuker: you should put film grain into ffv2/daala2 and make it good :^) [20:10:17 CET] <durandal_1707> ffv2 is lossless [20:12:07 CET] <durandal_1707> supposed to be? [20:12:41 CET] <gnafu> I don't believe so. I'm pretty sure atomnuker is working on a lossy video codec using technologies from Daala. [20:13:12 CET] <JEEB> it's like the third thing that's been called "ffv2" ever since pengvado's thing [20:13:33 CET] <JEEB> and yes, atomnuker's "presentation" thing mostly was about lossy coding schemes and ranting [20:13:33 CET] <gnafu> https://github.com/atomnuker/FFmpeg/tree/exp_ffv2_daala [20:14:05 CET] <JEEB> also TIL that someone made an independent implementation for FFV1 [20:14:12 CET] <JEEB> for the archivist community [20:14:35 CET] <JEEB> or to verify the spec, I guess [20:14:51 CET] <durandal_1707> nooooooo, ffv2 must be lossless! [20:15:12 CET] <JEEB> durandal_1707: don't worry - it's just his hobby project :P [20:15:14 CET] <atomnuker> it'll be fiiine, its winter again, I have to work on an encoder [20:16:01 CET] <durandal_1707> serious devs can code when it is hot [20:19:12 CET] <atomnuker> best works with simd asm [20:19:23 CET] <atomnuker> this summer was too hot for even that [20:22:27 CET] <durandal_1707> gonna write some simd before it becomes too cold? [20:23:26 CET] <atomnuker> I am, I have some almost finished av1 simd for dav1d [20:24:50 CET] <BBB> 8) [20:25:15 CET] <j-b> I finally watched the FFv2 presentation, calmly. [20:25:19 CET] <j-b> very cool [20:25:53 CET] <durandal_1707> j-b: calmly? something was bad? [20:26:08 CET] <j-b> durandal_1707: during VDD, I'm always in rush :) [20:26:18 CET] <j-b> So I can't enjoy the conf as well as I should [20:26:38 CET] <kierank> j-b: the nttw sessions are nice [20:26:43 CET] <kierank> They are super keen about oss [20:26:49 CET] <JEEB> I mean, it mentions nice techniques etc, but I think the response towards the "I thought this was lossless, why was this mostly about lossy coding techniques?" wasn't too great :) [20:26:59 CET] <JEEB> kierank: yea that kind of came out of the woodworks for me [20:27:03 CET] <JEEB> seemed like a nice event [20:27:07 CET] <kierank> They are not jaded like us [20:27:17 CET] <kierank> They love carl [20:27:23 CET] <kierank> Like durandal_1707 does [20:27:25 CET] <durandal_1707> shit [20:27:42 CET] <kierank> LOL [20:29:17 CET] <BBB> nttw? [20:29:35 CET] <kierank> No time to wait [20:31:12 CET] <JEEB> where was that btw? [20:31:27 CET] <JEEB> it could make my list of "events I'd like to visit" for next year [20:31:50 CET] <JEEB> ah, london [20:34:44 CET] <JEEB> will have to keep in mind next year :) [20:34:51 CET] <JEEB> (on where it's going to be held etc) [20:36:11 CET] <durandal_1707> it could be remote only conference [20:42:30 CET] <cone-403> ffmpeg 03Mark Thompson 07master:c0692cb2bb3b: vaapi_encode_mpeg2: Fix width/height columns/rows confusion [20:43:22 CET] <jkqxz> Not sure how that happened. I really think I tested it with non-square video... [20:44:28 CET] <j-b> kierank: I will watch nttw tehn [20:51:27 CET] <durandal_1707> how to name filter which would add black(or solid color) frames to beginning or end of video, delaying original video or padding it? [20:51:53 CET] <BradleyS> tpad [20:52:00 CET] <BradleyS> temporal pad [20:52:02 CET] <atomnuker> colorpad [20:53:02 CET] <kierank> durandal_1707: you were invited [20:53:07 CET] <kierank> I would have payed [20:53:26 CET] <kierank> Dave rice said you write good filters [20:53:40 CET] <durandal_1707> kierank: i cant leave country - i'm mafia boss [20:56:17 CET] <durandal_1707> besides, i want to avoid to meet voldemort/saruman in person [20:56:51 CET] <kierank> Lol [20:57:55 CET] <BradleyS> the eye sees all [21:09:15 CET] <uartie> jkqxz: thx :) [21:17:03 CET] <jkqxz> uartie: Do you have a CI setup somewhere which is catching that? [21:17:34 CET] <uartie> jkqxz: yup [21:18:03 CET] <uartie> we are currently running ci internally at intel [21:19:22 CET] <uartie> jkqxz: what's your take on #7523 [21:19:38 CET] <uartie> seems odd that [h264_vaapi @ 0x264d240] Slice count rounded up to 45 (from 4) due to driver constraints on slice structure. [21:21:17 CET] <jkqxz> Just looking at that now... [21:22:37 CET] <jkqxz> The i965 driver doesn't offer support for any slice layouts at all in LP, though it does say it can support up to 32 slices. [21:22:58 CET] <uartie> yeh, saw that [21:23:10 CET] <uartie> bug in i965? [21:23:29 CET] <jkqxz> Looks like it. [21:23:43 CET] <jkqxz> Assuming it does actually support slices. [21:24:01 CET] <uartie> yeh, not sure. iHD should support it though [21:24:57 CET] <uartie> but 45 seems odd [21:25:46 CET] <jkqxz> 45 is just the number of rows. Does it support the EQUAL_ROWS mode? (1 slice per row, MPEG-2 style.) [21:26:15 CET] <uartie> hmm, not sure [21:27:52 CET] <uartie> i'll cc iHD maintainer [21:28:41 CET] <jkqxz> Yeah, it supports 12 = EQUAL_ROWS | MAX_SLICE_SIZE. [21:29:44 CET] <jkqxz> Since the only fixed layout it says it supports is single rows, the request for 4 slices got rounded up to 45 because that's the number of rows in your 720 lines. [21:29:58 CET] <uartie> i see [21:30:09 CET] <uartie> ugh, trac won't let me modify cc [21:34:38 CET] <jkqxz> (Relevant caps: <http://ixia.jkqxz.net/~mrt/ffmpeg/skylake_ihd>.) [21:36:10 CET] <uartie> yup, just saw that in iHD code [21:38:18 CET] <jkqxz> Hmm, the second slice header coming out of iHD is very broken. first_mb_in_slice is correct, but then slice_type and everything after is just rubbish. [21:41:23 CET] <uartie> i can transfer the bug to iHD if you'd like. Or did u want to debug more and file it? [21:43:59 CET] <jkqxz> Go ahead. The driver seems to have only used part of (or overwritten some of?) the slice header, so I'd just be looking in the driver code at this point. [21:44:19 CET] <uartie> kk [21:56:23 CET] <durandal_1707> let's see who knows what 'group delay' means: [22:43:12 CET] <durandal_1707> i need to add group delay drawing to aiir filter IR filter response [00:00:00 CET] --- Tue Oct 30 2018

1 0

ffmpeg.log.20181029
by burek 30 Oct '18

30 Oct '18

[01:09:00 CET] <asterismo_l> furq, its a bug in the x264 lib? [09:42:17 CET] <Shibe> Hi, what is wrong with this command ? sudo ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -vcodec h264_vaapi -f pulse -ac 2 -i 1 -vcodec libx264 -crf 0 -preset ultrafast -acodec pcm_s16le output.mkv [09:42:27 CET] <Shibe> Option vf (set video filters) cannot be applied to input url 1 -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to. [09:57:41 CET] <durandal_1707> Shibe: why are you having two -vcodec ? [10:20:07 CET] <MarcoT> Hello [10:20:18 CET] <MarcoT> may you help me with my problem? I am using the example here (https://ffmpeg.org/doxygen/trunk/decoding__encoding_8c_source.html#l00503) to build a decoder. However, in the example they read from file, while I should read from socket, and in the framework I am using, I have already TCP packets as input. What I do now is to fill avcodec_decode with that packets but it results in some warnings (ac-tex damaged, Warning MVs not availabl [10:20:31 CET] <MarcoT> And at a certain point avcodec_decode fails returning a negative number (Error slice too small) [10:21:43 CET] <bencoh> Most codecs need complete encoded frames to be able to decode video stream [10:22:06 CET] <MarcoT> I am using mpeg [10:22:28 CET] <bencoh> meaning you need some code to chunk stream properly before feeding it to decoder [10:23:58 CET] <bencoh> you can either write it (it can get pretty complex with some formats), re-use code from some other project, or somehow manage to feed your data to avformat before handing it to avcodec [10:25:24 CET] <MarcoT> But if I read a video from file, with chunks of size 1448 (length of the packets I receive) it works [10:25:25 CET] <bencoh> (see AVIOContext for instance ... assuming this is still the way to do it nowadays) [10:25:38 CET] <bencoh> it "works" yeah [10:29:42 CET] <MarcoT> Yes actually I looked on other projects but I found nothing interesting to solve my problem, because they use AVInputFormat and AVStream to directly read from a url... [10:32:03 CET] <MarcoT> Anyway thanks for your help @bencoh, I will see if I can solve with AVIOContext ;) [10:32:44 CET] <bencoh> :) [10:33:00 CET] <bencoh> you might want to check first whether this is still the proper way to do it :) [10:33:39 CET] <MarcoT> Hmm what do you mean? [10:34:13 CET] <bencoh> I haven't written ffmpeg-related code for a few years, so I might have missed API changes since then [10:36:04 CET] <MarcoT> Ah ok, but at least you gave me hope :) [10:41:28 CET] <mertyildiran> Hi, I have a video file that keeps being updated by another program (let's say every 10 seconds and it's adding more and more frames at the end of the video). What I want to do here is to play that video file using ffmpeg and continue to play the new frames whenever the file updated by the other program. How can I do that? [10:41:59 CET] <BtbN> What format is the file in? [10:42:13 CET] <mertyildiran> I also wonder if are we able to replay the video whenever the file updated? [10:42:21 CET] <mertyildiran> BtbN: it's .mp4 [10:42:28 CET] <BtbN> Impossible then [10:42:36 CET] <BtbN> mp4 cannot be played until it's finalized [10:42:53 CET] <mertyildiran> it's finalized [10:42:59 CET] <BtbN> You just said the opposite [10:43:45 CET] <mertyildiran> there is a playable file with let's say 4 seconds duration. After 10 seconds delay we add 2 more seconds to the video file so it becomes 6 seconds. [10:43:59 CET] <mertyildiran> what I want to do is continue to play that 2 seconds [10:44:21 CET] <BtbN> It can't be mp4 then. Unless you completely re-write the whole file from scratch every time, which sounds highly impractical and slow. [10:44:41 CET] <BtbN> mp4 has a final header with an index on how to play the whole file. That final header sits at the end of the file [10:45:07 CET] <BtbN> So appending to an mp4 file is impossible, and playing an mp4 file without that header is also impossible. [10:45:23 CET] <mertyildiran> it's .mp4 and yeah it's rewritten every time. I'm not debating whether it's efficient or not [10:45:34 CET] <mertyildiran> just want to continue to play the file [10:45:45 CET] <BtbN> You cannot reasonably "continue" then. As there is no meaningful way to continue. [10:45:46 CET] <mertyildiran> also wondering if it's possible to replay it [10:46:02 CET] <BtbN> You'll have to wait until it's no longer being re-written [10:46:27 CET] <c_14> you can store the current timestamp, reopen the file and seek to the timestamp (assuming the file at that point is complete and indexed) [10:46:53 CET] <BtbN> You'll need to write your own application for that though. Pretty sure nothing out of the box supports such a weird mode [10:47:03 CET] <ritsuka> BtbN: it's possible if you use a fragmented mp4, but the player should support it [10:47:14 CET] <BtbN> ritsuka, this doesn't sound like fragmented mp4 at all [10:47:23 CET] <BtbN> The easy solution would be to not use mp4, but mkv or mpeg-ts. [10:47:27 CET] <BtbN> Those can be appended to [10:47:48 CET] <mertyildiran> OK how can I achieve this if the file is mkv? [10:47:54 CET] <ritsuka> for example QuickTime Player on macOS supports it, you can watch a mp4 exported by Final Cut and it will update as soon as a new part is ready [10:47:56 CET] <mertyildiran> could you give me the command? [10:48:06 CET] <BtbN> Just pipe it into ffmpeg with tail -f [10:48:36 CET] <BtbN> (and make sure tail -f starts at the beginning, not actually just at the tail) [10:49:36 CET] <mertyildiran> I was actually using ffplay to play the file. Is there any option for ffplay? [10:49:51 CET] <c_14> the same thing, but with ffplay instead of ffmpeg [10:49:52 CET] <BtbN> no, but you can use the exact same method [10:50:17 CET] <mertyildiran> if there is none then please provide the complete command for ffmpeg because I did not understand the usage [10:50:26 CET] <BtbN> One problem with this is though, that it will never know the file actually finished, and hang forever at the end [10:50:54 CET] <mertyildiran> "hangover at the end" sure that's the exact behavior I'm looking for [10:50:56 CET] <BtbN> What even is the usecase for that? It sounds as weird as it gets. [10:51:12 CET] <mertyildiran> live-stream teaching solution [10:51:18 CET] <BtbN> Well, at some point, it will _actually_ end. And you have no way to finish cleanly then [10:51:20 CET] <bencoh> uh? [10:51:42 CET] <BtbN> And why is your live stream in an mp4 file that gets constantly fully re-written? [10:51:53 CET] <mertyildiran> https://github.com/3b1b/manim [10:52:14 CET] <mertyildiran> This library generates math animations (using ffmpeg by the way) [10:52:57 CET] <mertyildiran> so I'm investigating methods to turn that video generation into live-stream [10:53:19 CET] <mertyildiran> though I couldn't figure it out directly from the library itself and looking for a workaround [10:53:33 CET] <BtbN> Unless that thing generates a constant and steady flow of frames, any kind of livestream from it will fail [10:54:44 CET] <Kleiner0901> Does anybody know how to input a H264 fragment into FFMpeg? [10:55:16 CET] <BtbN> define h264 fragment. Just a plain .h264 file with raw bitstream? [10:55:17 CET] <Kleiner0901> I have raw H264 frames created both by extracting from a mp4 with ffmpeg and created programmatically [10:55:46 CET] <mertyildiran> it generates animation by animation. Watch one of his videos so you will understand the flow https://www.youtube.com/watch?v=WUvTyaaNkzM [10:57:42 CET] <Kleiner0901> >"define h264 fragment. Just a plain .h264 file with raw bitstream?" - a file beginning 0000 0001 <h264 bitstream> [10:58:11 CET] <BtbN> ffmpeg can take that as normal input. Use one of the various concat methods to combine them. [10:58:56 CET] <Kleiner0901> thanks I'll look for those [11:07:51 CET] <sibok> Hi i'm on Funtoo GNU/Linux and compiling ffmpeg saying smbclient does not exists. Samba is already installed on the system and the needed /usr/include/samba-4.0/libsmbclient.h also exists. This [1] is the full (13k lines) log file. And this [2] is the last 500 lines of the config.log file. Could someone take a look? thx [1]https://drive.google.com/open?id=1f7SAahTjBUJIoXtvOlOExULC5Z8h5KHI [2]https://pastebin.com/Dn3502cu [11:19:29 CET] <BtbN> sibok, there's a bunch of errors in that header, so it can't compile the test program. [11:19:59 CET] <BtbN> try latest ffmpeg master, if it doesn't build there, report a bug [11:27:39 CET] <mertyildiran> BtbN: I changed the output format to ".mkv" So what's the easy solution you were talking about? [11:28:06 CET] <mertyildiran> I'm also able to get .mov format [11:28:25 CET] <BtbN> That depends... does it still re-create the whole file every time? If it just keeps appending, just use tail -f ... | ffmpeg ... [11:28:29 CET] <BtbN> mov is mp4. [11:28:42 CET] <BtbN> at least for all that matters [11:29:24 CET] <mertyildiran> BtbN: I don't understand what do you mean with "-f ... | ffmpeg ..."? [11:29:42 CET] <mertyildiran> "tail -f ... | ffmpeg ..." [11:29:54 CET] <mertyildiran> what are these ....s? [11:30:02 CET] <BtbN> dots, put the right stuff you need there.. [11:30:47 CET] <BtbN> Also, if it still re-writes the whole file every time, that won't work. I don't think there's a proper way to do that at all [12:02:15 CET] <Kleiner0901> BtBN do you know where I should look for converting a raw frame to image? My attempt is so far resulting in failure https://i.imgur.com/BuwwnLX.jpg [12:02:34 CET] <BtbN> raw frame as in raw yuv/rgb? [12:02:59 CET] <Kleiner0901> raw h264 extracted from yuv420p [12:03:36 CET] <BtbN> That looks like you told ffmpeg to interpret the h264 file as raw uncompressed yuv. [12:04:00 CET] <Kleiner0901> ohhh [12:25:28 CET] <Kleiner0901> "looks like you told ffmpeg to interpret the h264 file as raw uncompressed yuv." - thanks got it to work :) [13:31:01 CET] <fling> Is there a fine filter for making a slow motion video? [13:31:12 CET] <fling> The problem is the source has only 25 fps [13:40:08 CET] <durandal_1707> fling: minterpolate... there are other solutions too, but not in lavfi [13:42:19 CET] <fling> durandal_1707: thanks will try it [13:48:16 CET] <fling> durandal_1707: I'm about to make a side-by-side comparison of two clips, wanted to slow them down to make the similarity more obvious to the viewer. [13:49:18 CET] <JEEB> you could use minterpolate and then slow it down (or in the other order of first slowing down and then doing minterpolate) [13:49:58 CET] <fling> I will script this today. [13:50:42 CET] <fling> Both clips are at 25fps but the object on the video is moving with different speeds so I will need to change fps twice on one of them or drop some frames. [14:25:42 CET] <analogical> how do I use FFmpeg to split randomvideo.mp4 into one video track and one audio track ? [14:26:48 CET] <BtbN> -c:v copy -an -sn /// -c:a copy -vn -sn [14:28:22 CET] <analogical> what does the /// do ? [14:29:20 CET] <spaam> its a seperator between the first arguments and the second one . [14:29:50 CET] <analogical> is it really necessary? [14:30:25 CET] <spaam> you need to run ffmpeg twice .. [14:30:55 CET] <BtbN> Technically you can do it in one invocation, but just running it twice is simple enough [14:58:10 CET] <analogical> [NULL @ 0000000002c61080] Unable to find a suitable output format for '///' [14:58:46 CET] <durandal_1707> you are not supposed to put that [15:03:10 CET] <analogical> ffmpeg.exe -i video.mp4 -c:v copy -an -sn doesn't work :/ [15:03:34 CET] <fling> How do I slowdown with minterpolate properly? Should I just set output fps somehow? [15:03:50 CET] <JEEB> I think setpts filter forces the frame rate instead of dropping/adding frames [15:04:02 CET] <JEEB> and then minterpolate that interpolates that lower fps to higher [15:04:43 CET] <fling> Or I can set the input fps [15:06:40 CET] <analogical> please tell me exactly what to type so I can split the audio and video from movie.mp4 into two separate tracks? [15:07:20 CET] <TheSashmo> does anyone know if the 2110-20 has been added to any ffmpeg builds or any other open source solutions? [15:09:34 CET] <JEEB> TheSashmo: there's rtpdec_rfc4175 [15:10:05 CET] <TheSashmo> thanks JEEB [15:10:24 CET] <JEEB> which supposedly is related [15:13:00 CET] <JEEB> seems like its naming was mostly related to teh fact that at the point of merging 2110-20 was not yet published :P [15:45:17 CET] <fling> What does this mean? [ffmpeg/video] h264: Invalid NAL unit 0, skipping. [15:45:30 CET] <fling> it is sometimes unit 7 or unit 8 in mpv output [15:48:35 CET] <dustobub> Would anyone be interested in doing a small custom (paid) patch on libavfilter/vf_pad? [15:49:41 CET] <JEEB> fling: means that the NAL unit (AVC and HEVC packets are called NAL units) type id was invalid [15:49:53 CET] <JEEB> aka "I dunno what type of NAL unit this is, ignoring" [15:50:30 CET] <JEEB> dustobub: I would guess durandal_1707 is one of the filtering people [15:50:49 CET] <fling> JEEB: is this bad? :P [15:51:29 CET] <JEEB> could be anything from "This is a bug" to "the file just has random data in there" [15:51:44 CET] <JEEB> if you have a sample available you could post it on the trac [15:52:02 CET] <dustobub> JEEB: thanks! I'll ping them. [15:52:20 CET] <JEEB> dustobub: also generally I guess you could make a feature request on the trac issue tracker [15:53:18 CET] <fling> It works! [15:53:34 CET] <fling> First I padded the first clip ffmpeg -y -i "concat:$annexb_pad|$annexb_first" -c copy -bsf:a aac_adtstoasc $padded_first [15:53:37 CET] <dustobub> JEEB: yeah, but it's for a very specific use-case and I doubt would be generally useful. I need the padding color to be unique per frame (with around 1000 unique colors/frames possible) [15:53:55 CET] <fling> Then I stacked it with the second clip and it looks great! [15:56:51 CET] <fling> analogical: ffmpeg -i randomvideo.mp4 -c:v copy -an video-track.mp4 [15:57:30 CET] <fling> analogical: ffmpeg -i randomvideo.mp4 -c:a copy -vn audio-track.mp4 [15:57:50 CET] <fling> analogical: might also want -sn [16:20:30 CET] <fling> ffmpeg -r 30 -i $padded_first -r 25 -i $second -filter_complex hstack& [16:20:41 CET] <fling> How do I slow it down properly? ^ [16:20:52 CET] <q3cpma> Hello, does anyone know if the "100 buffers queued in out_0_0, something may be wrong." message is important? [16:20:53 CET] <fling> -filter_complex hstack,setpts=? [16:24:52 CET] <fling> Looks like I'm not doing it properly [16:25:18 CET] <fling> minterpolate should go before hstack/vstack [16:40:01 CET] <durandal_1707> q3cpma: it may be bad if it increases to 1000 buffers, what filtergraph? [16:42:28 CET] <q3cpma> durandal_1707: None, just copying some streams and converting truehd 24bit to flac s16. [16:43:06 CET] <durandal_1707> ah, then it should not increases to 1000 at all [16:43:26 CET] <q3cpma> I suppose, the interweb only gave me filtering problems. [17:10:29 CET] <fling> How do I convert to anaglyph? [17:10:57 CET] <fling> Should I just stack and apply the filter I use for viewing 3d in glasses? [17:13:53 CET] <durandal_1707> fling: anaglyph is old tech [17:15:23 CET] <durandal_1707> if you have separate files for each eye and want to do one of anaglyphs use hstack/vstack & stereo3d filters [17:15:44 CET] <fling> durandal_1707: this is what I'm thinking about. [17:17:18 CET] <durandal_1707> http://trac.ffmpeg.org/wiki/Stereoscopic [17:26:32 CET] <fling> durandal_1707: this worked for me -vf stereo3d=sbsl:arch [18:29:59 CET] <gh0st3d> Hey everyone, looking for some help. I've got 2 commands that create an animated gif with the first 10 seconds of a video. I want to change the 2nd line to also include a png file as an overlay on the gif... Is this even doable in 2 commands? Here's the 2 lines and my attempt at changing the 2nd line: https://hastebin.com/ecayexufec.md [18:31:19 CET] <durandal_1707> that link looks broken to me [18:32:56 CET] <gh0st3d> Hmm, works for me, here's a different one though: https://pastebin.com/tVvY3ehD the ffmpeg.pastebin.com link from the channel info is not working for me so I used these [18:33:28 CET] <gh0st3d> Ugh, also ignore the ffmpeg typo in the first line. That was from me editing out the python code [18:36:35 CET] <durandal_1707> gh0st3d: you need to merge two complex graphs into one [18:37:20 CET] <gh0st3d> Ohh ok. I'll try and figure that out in a few and get back on here if I need more help [18:37:24 CET] <gh0st3d> Thanks for the info! [18:39:00 CET] <durandal_1707> gh0st3d: ffmpeg -t 10 -i "/tmp/videofile.mp4" -i overlay.png -i /tmp/palette.png -filter_complex "[0:v][1:v]overlay=25:25:enable='between(t,0,10)',fps=10,scale=320:-1:flags=lanczos[x];[x][2:v]paletteuse" [18:39:08 CET] <durandal_1707> something like that [18:39:51 CET] <durandal_1707> note that you may need to use palettegen with overlay too, to get best results [18:50:23 CET] <devinheitmueller> Does anyone have any thoughts on how safe it would be to call avfilter_graph_alloc_filter() from within a filter itself? [18:51:14 CET] <devinheitmueller> The use case is a filter which includes a bunch of business logic which decides what combination of scaling/fps/pad to run against video frames. [18:53:16 CET] <devinheitmueller> Doh, meant to say avfilter_graph_create_filter, not avfilter_graph_alloc_filter. [18:53:28 CET] <durandal_1707> you mean separate graph? [18:53:53 CET] <devinheitmueller> No, I want a filter which adds other filters to the current graph. [18:54:35 CET] <devinheitmueller> I dont want to create a huge filter which replicates the functionality in scale/fps/pad, and I would prefer not to jam the logic into ffmpeg.c or avfiltergraph.c, since its specific to one use case. [18:54:51 CET] <durandal_1707> ugh, that can not be possible [18:55:40 CET] <durandal_1707> to be sure, also ask on devel mailing list [18:55:50 CET] <devinheitmueller> Fair enough. Figured I would start here first. [19:11:35 CET] <iive> devinheitmueller, maybe you want to create a separate graph, so your filter takes the input, processes it through the graph and outputs the result of the graph... [19:11:51 CET] <iive> no idea if that is possible at all . [19:11:56 CET] <devinheitmueller> Yeah, I had considered that. [19:12:36 CET] <devinheitmueller> Its not a terrible idea. It obscures to the outside though that the filter references other filters, which could break things if the filter uses fifos. [19:13:13 CET] <devinheitmueller> I do like that today you can dump out the entire filter graph to stderr, and see all the elements in one view. [19:14:36 CET] <devinheitmueller> It feels like a concept that could be really useful (i.e. filters which realize they need to invoke other filters), but I can imagine it causing issues with stuff that is in the process of iterating over the list of graph filters. [19:15:03 CET] <devinheitmueller> We do this today for the scaling filter, but its buried in avfiltergraph.c and isnt very flexible. [19:33:48 CET] <aloo_shu> hey, I think I can describe a long standing and reproducible bug, but would not like to go through filing a proper report myself. Describing it is simple. Anybody willing to listen for a minute? [19:36:22 CET] <durandal_1707> ? [19:37:26 CET] <ChocolateArmpits> aloo_shu, well we can listen [19:37:56 CET] <aloo_shu> the documentation suggests, in two places, to use the image2 filter as an input filter, in order to combine a series of images to a video [19:38:15 CET] <ChocolateArmpits> image2 format * :) [19:38:25 CET] <aloo_shu> like, for instance ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi [19:38:48 CET] <aloo_shu> where WxH needs to be specified [19:39:40 CET] <aloo_shu> what it does, instead, is, as I've repeatedly observed over the years, and now up to v. 4 [19:40:29 CET] <aloo_shu> 1) it *overwrites* the input files with resized versions of themselves [19:41:06 CET] <aloo_shu> 2) it converts exactly one image only to the desired video format [19:43:42 CET] <aloo_shu> either a) image2 needs fixing, or b) combining images needs a different approach, like e.g. concatenate, or c) I've been blatantly missing something [19:44:43 CET] <durandal_1707> how it overwrites when that never worked? [19:45:21 CET] <aloo_shu> try to reproduce, you'll see [19:46:05 CET] <durandal_1707> aloo_shu: what version of ffmpeg you use? [19:46:11 CET] <aloo_shu> seems to be a first pass before converting [19:46:27 CET] <aloo_shu> wait a sec, but various [19:49:07 CET] <furq> aloo_shu: http://vpaste.net/4UrjN [19:49:41 CET] <durandal_1707> i just tried and: 1) did not happen, 2) same [19:54:27 CET] <aloo_shu> ah, getting nearer, thought I had newer versions. got 1.0.10 and a statically compiled one from about the same time, both armv7, the other instance was on ppc debian a while back. that turns it into a different question: any newer version available backported to debian wheezy (the highest that will run on the kernel that android is forcing me to use) for armhf/armv7? [19:59:25 CET] <furq> https://www.johnvansickle.com/ffmpeg/ [19:59:29 CET] <furq> you can try the arm builds on here [20:07:10 CET] <aloo_shu> furq: I think that's where I had my static built from (built 2014), but there seems to be newer. shall manually check wheezy-backports, too, deb-multimedia gives me 1.0.10 only [20:08:34 CET] <aloo_shu> thanks so far, keep up the good work :) (I'd love to see all linux software so well documented) [22:12:55 CET] <flying_sausages> Hey guys, I'm getting a flood of output like the example in the paste even when using the -v quiet option. Anyone know why that is and how can i get rid of it? [22:12:57 CET] <flying_sausages> https://privatebin.net/?028e33cc9cfd269e#PdjX93xID+7HUSLcJc+4+0/u8NJgQ0LNDO… [22:15:55 CET] <durandal_1707> flying_sausages: you use some ffprobe command? [22:18:42 CET] <ariyasu> try add -stats [22:18:56 CET] <ariyasu> ffmpeg -v quiet -stats 'rest of your command here' [22:18:57 CET] <flying_sausages> durandal_1707, not as far as I'm aware, I'm on WSL [23:09:14 CET] <brimestone> Hey guys, im trying to use ffprobe -select_streams v:0 -show_entries stream=avg_frame_rate which works.. but how can I look at the timecode. Its TAG:timecode=23:49:38:00 [00:00:00 CET] --- Tue Oct 30 2018

1 0

ffmpeg-devel.log.20181028
by burek 29 Oct '18

29 Oct '18

[00:25:55 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.0:82e796a4c9cc: avcodec/msrle: Check that the input is large enough to contain a end of picture code [00:25:56 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.0:15296d64ca4f: avutil/integer: Fix integer overflow in av_mul_i() [00:25:57 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.0:527e64d32c34: Changelog: Update [02:18:28 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07n3.0.12:HEAD: avfilter/vf_pixdesctest: Use 32bit read/write [02:57:17 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:7489a527f000: avutil/pixfmt: Document chroma plane size for odd resolutions [02:57:18 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:5cbf4849e376: swresample/swresample: Fix input channel count in resample_first computation [02:57:19 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:9da24737a3b2: avcodec/diracdec: Prevent integer overflow in intermediate in global_mv() [02:57:20 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:330ed0772c6f: avcodec/dirac_dwt_template: Fix several integer overflows in horizontal_compose_daub97i() [02:57:21 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:7068bcf58ad2: avcodec/diracdec: Change frame_number to 64bit as its a 32bit from the bitstream and we also have a -1 special case [02:57:22 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:a594ce26ce38: avcodec/diracdec: Check slice numbers for overflows in relation to picture dimensions [02:57:23 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:9abcade734cb: avcodec/diracdec: Check bytes count in else branch in decode_lowdelay() too [02:57:24 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:7abc4445f911: avcodec/qtrle: Check remaining bytestream in qtrle_decode_XYbpp() [02:57:25 CEST] <cone-511> ffmpeg 03Nikolas Bowe 07release/3.4:c90457a95ee8: lavc/svq3: Fix regression decoding some files. [02:57:26 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:44e878d08674: avformat/flvenc: Check audio packet size [02:57:27 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:b6098dd17fec: avcodec/aacpsdsp_template: Fix integer overflow in ps_stereo_interpolate_c() [02:57:28 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:4df3a367df8e: avcodec/mpegaudio_parser: Initialize poutbuf* [02:57:29 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:7f480bedd069: avcodec/shorten: Check verbatim length [02:57:30 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:ec573bd2eb08: avcodec/shorten: Fix integer overflow in residual/LPC combination [02:57:31 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:e3cc5e81ab00: avcodec/shorten: Fix signed 32bit overflow in shift in shorten_decode_frame() [02:57:32 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:007da8396fed: avcodec/scpr: Check for min > max in decompress_p() [02:57:33 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:17c034797791: avformat/rmdec: Fix EOF check in the stream loop in ivr_read_header() [02:57:34 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:ff5196a98afb: avformat/mlvdec: read_string() received unsigned size, make the argument unsigned [02:57:35 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:0ef49c0818f2: avformat/nsvdec: Do not parse multiple NSVf [02:57:36 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:812f7fae356a: avcodec/snowdec: Fix integer overflow with motion vector residual [02:57:37 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:88afcff2f553: avcodec/vb: Check for end of bytestream before reading blocktype [02:57:38 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:b61b38766ee9: avcodec/hq_hqa: Check remaining input bits in hqa_decode_mb() [02:57:39 CEST] <cone-511> ffmpeg 03Michael Bunk 07release/3.4:462edf5b9435: examples: Fix use of AV_CODEC_FLAG_GLOBAL_HEADER [02:57:40 CEST] <cone-511> ffmpeg 03Dale Curtis 07release/3.4:eab5f6e419ec: avformat/mov: Error on too large stsd entry counts. [02:57:41 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:073a65aefc2e: avcodec/indeo4: Check dimensions in decode_pic_hdr() [02:57:42 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:ee8b4c16d792: avcodec/ra144: Fix undefined integer overflow in add_wav() [02:57:43 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:ab5d93076238: avcodec/h264_refs: Document last if() in ff_h264_execute_ref_pic_marking() [02:57:44 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:1a106752f37c: avcodec/dvdsubdec: Avoid branch in decode_run_8bit() [02:57:45 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:98709a124482: avcodec/shorten: Fix bitstream end check in read_header() [02:57:46 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:80af29f32e54: avcodec/zmbv: Update decomp_len in raw frames [02:57:47 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:127ec77e8cc5: avcodec/zmbv: Check that the decompressed data size is correct [02:57:48 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:f80da843b219: avcodec/mpeg4videodec: Fix undefined shift in get_amv() [02:57:49 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:50aa132f4d34: avcodec/dvdsubdec: Sanity check len in decode_rle() [02:57:50 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:848726afc642: avcodec/gdv: Replace divisions by shifts in rescale() [02:57:51 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:0cbd4fb9955e: avcodec/unary: Improve get_unary() docs [02:57:52 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:e9975d1b5118: avformat/utils: Fix integer overflow in discontinuity check [02:57:53 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:d17d08035cd4: avformat/utils: Never store negative values in last_IP_duration [02:57:54 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:a3ef90a73cd9: avcodec/ra144: Fix integer overflow in add_wav() [02:57:55 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:ced37ef52c24: avcodec/h264_cavlc: Check mb_skip_run [02:57:56 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:6763ff890e41: avcodec/mpeg4videodec: Fix typo in sprite delta check [02:57:57 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:1bcc79db44ed: avcodec/jpeg2000dec: Fix off by 1 error in JPEG2000_PGOD_CPRL handling [02:57:58 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:707ec3cfc00b: avcodec/msrle: Check that the input is large enough to contain a end of picture code [02:57:59 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07release/3.4:0e853b61e65c: avutil/integer: Fix integer overflow in av_mul_i() [02:02:51 CET] <J_Darnley> Oh hey. I can watch an hour more now that I've gone back in time [08:19:37 CET] <cone-726> ffmpeg 03hwren 07master:9c190ad39481: lavc/libxavs2: unified naming style [08:19:37 CET] <cone-726> ffmpeg 03hwren 07master:4c2326281106: lavc/libxavs2: fix intra period meaning conflict [08:19:37 CET] <cone-726> ffmpeg 03hwren 07master:c00ed8d0e7bd: lavc/libxavs2: enable OpenGop [08:19:37 CET] <cone-726> ffmpeg 03Jun Zhao 07master:6885fa749996: lavc/libdavs2: Remove yuv420p10le from fromat list. [08:19:37 CET] <cone-726> ffmpeg 03Jun Zhao 07master:bdfd2e3c79dd: lavc/libxavs2: Remove yuv420p10le from fromat list. [12:46:10 CET] <BtbN> philipl, I'll look at the series tomorrow, or at least this week if I don't find time tomorrow. [14:32:58 CET] <cone-726> ffmpeg 03Paul B Mahol 07master:bb54c0ae719e: avfilter/af_afftdn: switch to activate [15:02:42 CET] <cone-726> ffmpeg 03Paul B Mahol 07master:0c8b5cb369fc: avfilter/af_afftdn: add alias for sample_noise end [15:15:20 CET] <philipl> BtbN: much appreciated. [17:42:57 CET] <cone-726> ffmpeg 03Mark Thompson 07master:7070955d43dd: libaomenc: Add support for tiles [18:02:05 CET] <durandal_1707> what to not do next ... ? [18:03:48 CET] <atomnuker> an ldac decoder? [18:05:57 CET] <durandal_1707> no more sony [18:51:54 CET] <durandal_1707> libavcodec/libaomenc.c:545:9: warning: unused variable 'pict_type' [-Wunused-variable] [19:00:10 CET] <jkqxz> durandal_1707: Upgrade libaom. Is it useful to fix it for the old version? [19:00:56 CET] <durandal_1707> jkqxz: right, ignore it [19:02:33 CET] <jamrial> jkqxz: no, not worth it. as soon as they tag a new release i'll make it the minimum required version anyway [19:03:04 CET] <jamrial> libaom 1.0.0 is missing some essential API, a crapload of optimizations, and a lot of fixes [21:55:40 CET] <durandal_1707> is there a way after idct to reduce all blocks[x] by 1? [21:56:04 CET] <atomnuker> reduce? [21:56:16 CET] <durandal_1707> minus 1 [21:56:22 CET] <atomnuker> as in pixels[] -= 1? [21:56:26 CET] <durandal_1707> yes [21:56:55 CET] <atomnuker> can you alter the DC before transform? [21:57:01 CET] <jkqxz> Subtract N*N from the DC coefficient. [22:04:51 CET] <durandal_1707> hmm, doesnt help, for some pixels i get fade to 0 each next frame - perhaps this decoder needs new idct? [22:06:56 CET] <atomnuker> just write some SIMD, it'll be very quick [22:09:35 CET] <durandal_1707> why would i write SIMD? [22:14:52 CET] <atomnuker> well you can choose not to, the compiler will do it for you, in this case certainly [22:20:40 CET] <durandal_1707> atomnuker: but why for certain quality/qscale/quant_matrix it would give good output and for others it would give wrong one? [22:31:28 CET] <atomnuker> the -1 solution? [22:32:21 CET] <atomnuker> I thought they had that to handle fades, but no, especially not in yuv [22:32:30 CET] <durandal_1707> atomnuker: no, general case, the -1 solution makes some pixels stay as they should be, but others fade [22:33:55 CET] <cone-118> ffmpeg 03Paul B Mahol 07master:e95987f6ca8e: avfilter/af_afftdn: fix memory leaks reported by coverity [22:34:34 CET] <atomnuker> durandal_1707: is it signalled? is there something common for pixels which stay/fade? [22:35:34 CET] <durandal_1707> nothing is signalled, except flag for different idct in reference dll [22:36:24 CET] <durandal_1707> it probably does one line something different, it uses 255 constant - the other idct [22:37:46 CET] <atomnuker> what's the blocksize? [22:44:24 CET] <durandal_1707> 8x8 [22:46:01 CET] <durandal_1707> in both cases [22:46:25 CET] <atomnuker> do the pixels which don't fade have the same transform? [22:49:04 CET] <durandal_1707> everything fades with certain quantization, when its exactly 1 [22:49:21 CET] <durandal_1707> and it fade to 255 [22:50:01 CET] <durandal_1707> changing dc just fixes some pixels, but others fade to other side... [22:50:31 CET] <atomnuker> oh, II see [22:51:09 CET] <durandal_1707> for other, smaller quality if still fades somehow but much slower [22:56:36 CET] <atomnuker> I suppose they're using scalar quantization and in some ways they maybe have some primitive temporal AQ [23:02:31 CET] <jshanab> I am trying to learn the code as I want to consider replacing live555 in a large project but need to add some bits. But I am having a bit of trouble figuring out how the pieces go together. For example in the libswscale directory there is a rgb2rgb.h and .c but then there is also an rgb2rgb.c rgb2rgb_template.c and a rgb_2_rgb.asm. I do see the _template.c is directly included. Is that how... [23:02:32 CET] <jshanab> ...the code interfaces with the C code? [23:03:49 CET] <JEEB> usually SIMD is done so that the functions are defined as asm, and then there's function pointers that are set to the correctly optimized functions according to available instruction sets [23:04:01 CET] <JEEB> that said, my condolences for first of all looking into swscale [23:04:43 CET] <nevcairiel> the template file is used to create various versions of code for different instruction sets [23:05:09 CET] <JEEB> also isn't live555 just the rtsp parts? [23:05:20 CET] <nevcairiel> it probably has all sorts of stuff now [23:05:30 CET] <jshanab> I though it was easier than avcodec. I am trying to figure out the makefiles. [23:06:57 CET] <jshanab> live555 is rtsp.rtp,rtcp and with my changes rtsp-over-https and rtp backchannel. It is server and client with a lot of muxers and demux but NO decoding or file. That is what we use ffmpeg for [23:09:45 CET] <jshanab> So how do the c files of the same name and the assembly fit together. [00:00:00 CET] --- Mon Oct 29 2018

1 0

ffmpeg.log.20181028
by burek 29 Oct '18

29 Oct '18

[00:28:29 CEST] <Rathann> hi, can anyone provide a working FFmpeg command line that uses Intel QSV via lu-zero's libmfx to encode a H.264 stream? [00:33:04 CEST] <JEEB> did that need anything special? other than IIRC the intel drivers etc being extra finicky? [00:33:21 CEST] <JEEB> I think vaapi was the more generally working interface [00:33:34 CEST] <JEEB> (for encoding as well if you checked with vainfo [00:37:46 CEST] <Rathann> it works via vaapi [00:38:12 CEST] <JEEB> I'd generally recommend that. I remember the QSV crap needing special drivers etc [00:38:17 CEST] <Rathann> I just wanted to test libmfx code path [00:39:07 CEST] <Rathann> https://paste.fedoraproject.org/paste/9tc81EMYdK-WtdIQQUxnRQ <- here's the command and ffmpeg output [00:39:40 CEST] <c_14> Rathann: did you patch and recompile the kernel? [00:39:42 CEST] <Rathann> searching for the error message on the web reveals a couple of similar posts but no resolution [00:39:45 CEST] <c_14> and reboot of course [00:40:03 CEST] <JEEB> yea, that's what I meant with it being a PITA [00:40:09 CEST] <JEEB> QSV needs a patched kernel [00:40:13 CEST] <Rathann> c_14: why would I do that? which kernel and which patch? [00:40:13 CEST] <JEEB> and all that jazz [00:40:22 CEST] <c_14> Rathann: the qsv patchset from intel [00:40:24 CEST] <c_14> it's a royal pain [00:40:28 CEST] <Rathann> huh [00:40:29 CEST] <jkqxz> Are you using the open-source libmfx or the proprietary media SDK? [00:40:32 CEST] <c_14> especially when the patches don't apply and you have to fix them [00:40:38 CEST] <Rathann> jkqxz: libmfx [00:40:39 CEST] <JEEB> jkqxz: the lu_zero thing [00:40:54 CEST] <jkqxz> Open-source libmfx should work normally on current kernels. [00:41:00 CEST] <JEEB> ok [00:41:18 CEST] <jkqxz> (Without crazy patching, which is required for the proprietary one.) [00:42:44 CEST] <jkqxz> Hmm, "Unsupported". Does you system actually support it? (Is one of the platforms on <https://github.com/intel/media-driver>.) [00:44:53 CEST] <Rathann> jkqxz: I have a Haswell CPU [00:45:12 CEST] <Rathann> why would I look there to decide if it's supported or not? [00:45:45 CEST] <JEEB> that probably was supposed to be a "where" [00:45:49 CEST] <JEEB> and not "why" [00:46:01 CEST] <jkqxz> Haswell is not supported by libmfx at all any more. [00:46:13 CEST] <JEEB> alright, that explains it :) [00:46:25 CEST] <jkqxz> Well, it is on Windows with old drivers. But the open-source one doesn't. [00:51:58 CEST] <Rathann> oh? where does the code say that? [00:56:47 CEST] <jkqxz> "Supported Platforms" on that page. [00:57:42 CEST] <Rathann> I'm guessing this is the line that gives the error: https://github.com/lu-zero/mfx_dispatch/blob/master/src/main.cpp#L216 [00:58:08 CEST] <Rathann> jkqxz: but VAAPI works just fine? I don't understand, libmfx seems to depend on VAAPI [00:58:51 CEST] <JEEB> it IIRC uses other interfaces which might have been already left out or never supported in the standard open source drivers for this hardware [00:59:04 CEST] <JEEB> at least for encoding [00:59:23 CEST] <JEEB> vaapi on the other hand is just VAAPI (dec and enc) [00:59:40 CEST] <JEEB> and clearly intel seems to be putting more interest there than QSV for a lot of things [01:00:10 CEST] <Rathann> well, ok, I'll defer to you guys then [01:00:23 CEST] <Rathann> thanks for trying to help [01:00:33 CEST] <jkqxz> libmfx uses the iHD VAAPI driver rather than the i965 one, which doesn't support older platforms at all. [01:01:35 CEST] <JEEB> looking at the va_glue file all it seems to do is initialize a VA-API display and pass it to the MXFVideoCORE [01:02:26 CEST] <Rathann> ok [01:03:39 CEST] <Rathann> JEEB: indeed, I can see that [01:13:09 CEST] <analogical> how do I download a geo-blocked video from youtube? [01:15:10 CEST] <Rathann> jkqxz: can you point me to some definitive upstream information that it requires the iHD driver? [01:15:27 CEST] <Rathann> analogical: that is rather independent from ffmpeg ;) [01:16:08 CEST] <Rathann> analogical: I'd suggest accessing the video via a proxy in a location that is not blocked [01:16:17 CEST] <analogical> ok thanks [01:17:04 CEST] <Rathann> I used to do that with Netflix US, but they started blocking whole US datacenters where people ran their own proxies [01:17:14 CEST] <Rathann> so I stopped paying them ;) [01:17:59 CEST] <Rathann> local Netflix catalogue is like 10% of the US one [01:21:19 CEST] <jkqxz> Rathann: <https://github.com/Intel-Media-SDK/MediaSDK>, "System requirements". [01:22:05 CEST] <Rathann> jkqxz: got it, thanks a lot [04:26:40 CET] <pzy> I have a streaming input file with 8 stereo audio channels, how can I amerge that into 16 mono PCM tracks? [04:28:16 CET] <pzy> -filter_complex "join=inputs=8:channel_layout=8[aout]" ends up with a "hexadecagonal" layout that doesn't really work haha [04:33:38 CET] <pzy> actually my bad, been using -filter_complex "[0:a]amerge=inputs=8[aout]" [04:37:02 CET] <furq> pzy: -map_channel 0.0.0 out0.wav -map_channel 0.0.1 out1.wav -map_channel 1.0.0 out2.wav -map_channel 1.0.1 out3.wav ... [04:37:20 CET] <pzy> I am outputting them to another stream [04:38:44 CET] <pzy> right now it's something like: ffmpeg -i http://127.0.0.1/bigstream.ts -filter_complex "[0:a]amerge=inputs=8[aout]" -map 0:v -map "[a]" -c:v copy -c:a pcm_s16le -f mpegts - [04:39:09 CET] <pzy> err except that'd be -map "[aout]" [04:43:16 CET] <pzy> ffmpeg -i "http://127.0.0.1/bigstream.ts" -map_channel 0.0.0 [a1] -map_channel 0.0.1 [a2] -map_channel 1.0.0 [a3] -map_channel 1.0.1 [a4] [04:43:20 CET] <pzy> is that the right idea? [04:43:28 CET] <furq> you probably want amerge=inputs=8,channelsplit [04:43:57 CET] <pzy> oh snap [04:44:04 CET] <furq> https://ffmpeg.org/ffmpeg-filters.html#channelsplit [04:44:58 CET] <pzy> hmmmm [04:46:22 CET] <pzy> yeah that wasn't working for me, just gave me one LFE stream [04:54:34 CET] <pzy> -filter_complex "[0:a]channelsplit,amerge=inputs=8[aout]" [04:54:39 CET] <pzy> just splits the first audio stream's channels [05:12:44 CET] <pzy> same thing if I put it after inputs=8 [05:19:49 CET] <tomf> when converting FLAC to ALAC, is there a way to preserve the artwork (I believe its at -map 0:1)? [06:26:19 CET] <Foone> Hello! So I have a tool I'm building that wraps ffmpeg, and it needs to take a movie film and split it into lots of individual smaller files. so like seconds 1-5, 5-10, 10-15, etc. [06:26:31 CET] <Foone> I've got working code that does this in the simple way: seek, only extract X seconds, save [06:26:42 CET] <Foone> but the problem is that the accurate seeking gets slower and slower as you go farther in the film [06:27:09 CET] <Foone> is there a way to do this in one pass (so one movie file into a bunch of sub-files) or should I focus on doing a hybrid keyframe + accurate seek? [06:43:15 CET] <tomf> Foone: look into segment -segment_time -- you can do segment -segment_time 5 to get 5 second chunks, for instance [06:43:35 CET] <tomf> for ten minutes, you could use ffmpeg -i S24E05.mkv -acodec copy -f segment -segment_time 600 -vcodec copy -reset_timestamps 1 OUTPUT%d.mp4 [06:57:23 CET] <Foone> ahh. I need mine at non-uniform times but it looks like I can specify a list with segment_times. Thanks! [06:57:37 CET] <Foone> I'm rapidly going to run into command line length limits but this is a start [16:29:06 CET] <illuminated> lol i just used ffmpeg to create an avi file with an x264 video stream and an ac3 audio stream. I just thought it would be cool to put that in an avi cuz it supports it, and it's uncommon. [17:39:27 CET] <furq> you have a strange definition of cool [18:00:20 CET] <bencoh> :] [18:34:44 CET] <mango_99> Hi -vcodec copy and -acodec copy while slicing MP4s output does not play quite right in most players such as Chromium video element. Clip length/timeline gets messed up and beginning doesn't play. Reencoding works fine. Is there anything else I can play with for this command to give better output or figure out what's going wrong with it? [18:36:45 CET] <mango_99> https://pastebin.com/3BJWiDxW [18:48:27 CET] <mango_99> I tried with latest git [18:51:03 CET] <furq> mango_99: you can't do exact cuts with -c copy [18:51:13 CET] <mango_99> exact cuts? [18:51:19 CET] <furq> the cut section has to start on a keyframe, so it'll just seek to the nearest keyframe [18:51:28 CET] <mango_99> oh [18:51:34 CET] <mango_99> is there anything wrong with the command then? [18:51:39 CET] <furq> no [18:52:07 CET] <mango_99> when I try to play it it says it's a 20 second clip. and the progress bar is all messed up, it starts halfway through video [18:52:46 CET] <furq> try moving -ss before -i [18:53:23 CET] <mango_99> ah [18:53:30 CET] <mango_99> different result, now it's a 30 second clip [18:53:39 CET] <mango_99> better than before [18:53:45 CET] <mango_99> are keyframes that far apart? [18:53:53 CET] <furq> x264 defaults to 250 frames [18:54:01 CET] <furq> so it's probably something around that [18:54:15 CET] <mango_99> cool! [18:54:23 CET] <mango_99> thanks but why does putting -ss before work? [18:54:52 CET] <mango_99> and why does -ss after work as long as it's reencode [18:55:17 CET] <furq> they're just two different seeking methods [18:55:28 CET] <furq> same with -t/-to before/after -i [18:56:32 CET] <furq> i forget exactly how it all works but the issue you were having was that it was cutting the audio at the point you requested but then cutting the video at the next keyframe [18:56:43 CET] <furq> so the video was a few seconds delayed [18:58:17 CET] <mango_99> awesome, really appreciate the quick response thank you [19:00:35 CET] <mango_99> is there any way to tell it to round up/down when looking for nearest keyframe? [19:02:18 CET] <mango_99> seems to round down, 3s starts at 0s keyframe 10s rounds up [19:02:21 CET] <mango_99> looks like [20:44:30 CET] <teratorn> is there any docs for getting NTP timestamps from RTSP stream (RTCP SR report NTP timestamp) and syncing that to RTP packets? [21:43:58 CET] <TheSashmo> does anyone know if the 2110-20 has been added to any ffmpeg builds or any other open source solutions? [22:41:13 CET] <asterismo_l> hi, i'm running a mastodon instance that uses ffmpeg to convert gifs to mp4. After a migration from Debian 8 to Debian 9 that conversion is not working anymore, and i get this error line in the log: ffmpeg: common/cpu.c:251: x264_cpu_detect: Assertion `!(cpu&(X264_CPU_SSSE3|X264_CPU_SSE4))' failed [22:41:55 CET] <asterismo_l> do i have to manually compile ffmpeg? the server runs this CPU: http://paste.debian.net/1049440/ [22:44:55 CET] <durandal_1707> probably [22:54:11 CET] <asterismo_l> is there any tutorial on how to compile ffmpeg in debian 9 stretch from source? [22:54:36 CET] <asterismo_l> and overcome that issue? [23:00:37 CET] <furq> asterismo_l: what are you running that on [23:01:22 CET] <asterismo_l> running what? [23:01:39 CET] <furq> ffmpeg [23:01:44 CET] <furq> what system are you running it on [23:02:27 CET] <furq> or to put it another way, are you launching qemu/kvm yourself or is it just a vps or cloud hosting or something [23:03:20 CET] <asterismo_l> it is a VPS [23:03:30 CET] <asterismo_l> debian 9 stretch [23:03:54 CET] <asterismo_l> it is a vps cloud hosting [23:04:11 CET] <furq> i guess you just need to rebuild and cross your fingers then [23:04:31 CET] <furq> x264's cpu detection is (was?) broken sometimes in qemu if you don't pass the right -cpu flag [23:04:53 CET] <furq> idk if they fixed it but they've definitely updated that code since the version in debian 9 [23:34:49 CET] <asterismo_l> http://paste.debian.net/1049446/ [23:40:33 CET] <asterismo_l> is there any tutorial oh compiling ffmpeg in debian 9? [23:44:30 CET] <jkqxz> To build the ffmpeg utility just run "srcdir/configure" and the defaults are sensible. Most people want to add "--enable-libx264 --enable-gpl", but generally the other options are only for particular use-cases. [23:53:55 CET] <furq> asterismo_l: it's an x264 bug so you'll need to rebuild that and then build ffmpeg with it [23:54:21 CET] <furq> but you can just use the prepackaged debian libs for anything else you want [23:55:52 CET] <furq> https://www.johnvansickle.com/ffmpeg/ [23:55:55 CET] <furq> alternatively you can just use these [00:00:00 CET] --- Mon Oct 29 2018

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ffmpeg-devel.log.20181027
by burek 28 Oct '18

28 Oct '18

[00:00:53 CEST] <BBB> oh, and his nick is sometimes inverted (taripe) [00:00:56 CEST] <BBB> that one is online [00:01:02 CEST] <BBB> anyway, good to see it works now [00:01:09 CEST] <BBB> Im gonna say hi to the kids, bbl [00:02:03 CEST] <Taripe> atomnuker hi [00:02:45 CEST] <Taripe> sorry for the nickname confusion [05:16:16 CEST] <cone-962> ffmpeg 03James Almer 07master:59a35fe1f63d: tests/api-h264-slice-test: use the correct function to free the AVHashContext [05:28:00 CEST] <philipl> BtbN: https://github.com/philipl/nv-codec-headers/commit/94c72c5a74a4fae39d60964c… [05:28:17 CEST] <philipl> Allowed me to build and run the filter without SDK deps (beyond nvcc, of course) [09:50:14 CEST] <BtbN> philipl, now manage to build the kernels with clang/llvm, and you freed it. [09:51:26 CEST] <BtbN> No idea how practical it is, but llvm has an nvptx backend [13:38:02 CEST] <cone-384> ffmpeg 03Paul B Mahol 07master:40ac62246028: avfilter/window_func: add bohman window [13:54:19 CEST] <durandal_1707> >> The attacker's Bitcoin address contained the equivalent of only $40, with the last transfer being received way back in April, and suggesting that the colourama package failed to make any money. << [14:34:20 CEST] <atomnuker> BBB: no x86util, vpbroadcastw is an invalid instruction, movd m2, r4d doesn't work unlike in lavc [14:35:15 CEST] <BBB> movd xm2, r4d [14:35:35 CEST] <BBB> and yes, no x86util, not bsd, sorry [14:35:39 CEST] <BBB> Id like to use it but we can't [14:35:48 CEST] <BBB> its not that bad [14:35:53 CEST] <BBB> we dont care about mmx/sse2 support anyway [14:35:56 CEST] <BBB> wihch is half of x86util [14:36:27 CEST] <BBB> vpbroadcastw should work, I use that a lot, just not vpbroadcastw m0, r1d, thats invalid [14:36:39 CEST] <BBB> you need movd followed by vpbroadcastw m, m [14:36:52 CEST] <atomnuker> vpbroadcastw m2, r4w compiles but crashes [14:37:15 CEST] <durandal_1707> you cant use r* [14:37:34 CEST] <atomnuker> vpbroadcastw isn't mm, mm, its mm, gpr [14:59:42 CEST] <iive> i see xmm,xmm and ymm,xmm forms [15:01:23 CEST] <iive> are you writing avx512 ? [15:05:12 CEST] <iive> avx512VL/BL have gpr form. [15:08:02 CEST] <atomnuker> no [17:37:33 CEST] <philipl> BtbN: There's a working compiler but the language feature support is uneven (Doesn't look like texture objects are supported) and when they are supported (texture references), you need the cuda SDK around to use them. [18:37:23 CEST] <durandal_1707> i was thinking about writing filter which would count queued frames in filtergraph per each filter [18:37:43 CEST] <durandal_1707> anybody against it? [18:38:33 CEST] <JEEB> sounds like a general debugging tool [18:55:50 CEST] <durandal_1707> JEEB: yes, it is supposed to debug filtergraph [19:04:06 CEST] <durandal_1707> how should name it ? graphmonitor? [20:41:40 CEST] <cone-511> ffmpeg 03Mark Thompson 07master:2923ed247ee2: vaapi_encode: Support configurable slices [20:41:41 CEST] <cone-511> ffmpeg 03Mark Thompson 07master:29816e278f4f: vaapi_encode_mpeg2: Use common slice sizing code [20:41:42 CEST] <cone-511> ffmpeg 03Mark Thompson 07master:a769e72c750e: vaapi_encode_h264: Enable multiple-slice support [20:41:43 CEST] <cone-511> ffmpeg 03Mark Thompson 07master:a7eda762dce6: vaapi_encode_h265: Enable multiple-slice support [20:41:44 CEST] <cone-511> ffmpeg 03Mark Thompson 07master:fef2162b6ea7: vaapi_encode: Add flag to mark encoders supporting only constant-quality [21:11:03 CEST] <cone-511> ffmpeg 03James Almer 07master:99ef8b8afd99: avcodec/cbs_av1: fix parsing frame_size_with_refs [21:11:04 CEST] <cone-511> ffmpeg 03James Almer 07master:a5d98da4d6f3: avcodec/cbs_vp9: fix parsing sRGB samples [21:12:18 CEST] <pkv> @atomnuker reading page 10 of iso/iec 14496-3:2009/AMD.4:2013 , chan_config = 5 and 6 have been changed to 5.0(side) and 5.1(side) , and chan_config = 7 to 7.1(wide-side), chan_config =12 is 7.1. The interesting bit is chan_config =11 which is 6.1 [21:12:30 CEST] <pkv> http://www.doc88.com/p-7754944656492.html [21:13:23 CEST] <pkv> there are also two chan_configs with top channels: 13 = 22.2 and 14 = 7.1 top [22:48:39 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07master:718044dc1987: avutil/pixdesc: Add av_write_image_line2(), av_read_image_line2() [22:48:40 CEST] <cone-511> ffmpeg 03Michael Niedermayer 07master:cd34c6a57ee6: avfilter/vf_pixdesctest: Use 32bit read/write [00:00:00 CEST] --- Sun Oct 28 2018

1 0

ffmpeg.log.20181027
by burek 28 Oct '18

28 Oct '18

[02:33:40 CEST] <haasn> Before I bash my head against the desk any further, some quick help would be appreciated. I have a yuv422p10 frame inside y4m inside ivf [02:33:56 CEST] <haasn> I want the luma channel from this image (discarding chroma completely) as any lossless image format (e.g. 16-bit png) [02:34:15 CEST] <haasn> I tried using -filter_complex + extractplanes to save the y channel as its own .y4m file, that seemed to work [02:34:29 CEST] <haasn> but now I need to get it from .y4m into something like .png, and trying to do that results in a messed up image (too bright / wrong levels) [02:34:52 CEST] <haasn> (the y4m is gray10le) [02:35:33 CEST] <haasn> It seems like it might be a metadata issue - it displays correctly in `mpv`, but wrongly in `ffplay` [02:35:39 CEST] <haasn> PC/TV range confusion perhaps [02:36:12 CEST] <haasn> In mpv the file metadata attribute shows it's tagged as limited, but ffmpeg seems to treat it as full range [02:36:45 CEST] <haasn> Hmm the same issue happens if I try directly saving the output of extractplanes as png [02:56:55 CEST] <haasn> nvm I'll just mogrify -fx '(u - 16/255) * 255/219' [02:57:00 CEST] <haasn> can't be bothered wasting my time on such a stupid issue [06:24:12 CEST] <worstje> I'm currently trying to save some footage from files that were butchered because of a hard disk crash and make whatever is left into a compliant file. VLC manages to show a couple of seconds, sometimes with audio before stopping. ffmpeg however gets into an eternal loop repeating the same error message. How can I make ffmpeg give up on a particular input like vlc seems to do? [06:49:21 CEST] <akabuddy> hello all [06:50:59 CEST] <furq> worstje: try -xerror -err_detect explode [06:51:07 CEST] <Yagiza> akabuddy, hello! [06:51:54 CEST] <Yagiza> I implemented RTP sessions for VoIP in my XMPP client. [06:53:30 CEST] <Yagiza> For some reason opus and speex codex work all right, but with G722 I have distorted sound (like incorrect sample rate). [06:53:49 CEST] <Yagiza> Is there something special about G722? [06:53:55 CEST] <akabuddy> I am trying to record line in with alsa, i know im supplying the sound externally, but it isnt actually recording any sound [06:59:08 CEST] <furq> Yagiza: are you setting the sample rate [06:59:34 CEST] <Yagiza> furq, well... [07:00:50 CEST] <Yagiza> furq, yes. I check supported sample rates, then select one. [07:10:24 CEST] <akabuddy> nevermind [22:03:25 CEST] <ejr> how can i combine the following two commands into one? ffmpeg -i infile -vcodec copy -af "volume=10db" outfile /// ffmpeg -n -i infile -f mp3 outfile.mp3 (what i want to do is convert a video to mp3 and increase its size. both commands work independently but i want to do it in one process. i tried combining them but got errors...) [22:25:11 CEST] <Hello71> did you read the errors [22:27:34 CEST] <ejr> yeah nvm, just figured it out [00:00:00 CEST] --- Sun Oct 28 2018

1 0

ffmpeg-devel.log.20181026
by burek 27 Oct '18

27 Oct '18

[00:49:26 CEST] <llogan> michaelni: what's the reason we can't use PayPal for donations? IIRC, you mentioned the reasoning before. This is regarding the "Not a mail listing category suitable" email to devel-owner. [00:54:38 CEST] <michaelni> llogan, i am not sure what you refer to. There may have been some issue with SPI and paypal but i dont remember exactly. But checking i see paypal on https://www.spi-inc.org/projects/ffmpeg/, maybe we need to update our donation page [00:58:59 CEST] <llogan> michaelni: looks like donations page does need an update then. sorry, i meant the email asking about this was sent to ffmpeg-user-owner. i'll reply to it and CC. [01:46:41 CEST] <cone-457> ffmpeg 03James Almer 07master:bf324359be8d: avcodec/vp9_parser: set profile in AVCodecContext [09:27:46 CEST] <akravchenko188> Hello, guys. back to topic about testing and applying patches for amf support (hwcontext_amf, scaler_amf), were sent July [09:28:27 CEST] <akravchenko188> is there anybody who can review it and give some feedback? [09:30:05 CEST] <akravchenko188> is there still issue with missing AMD cards? How such issue solved here? [10:13:35 CEST] <cone-083> ffmpeg 03kjeyapal(a)akamai.com 07master:de43c227fd7d: avformat/dashenc: Support HTTP persistent for init segments as well [12:47:27 CEST] <cone-083> ffmpeg 03Paul B Mahol 07master:4fcfb9c4ebf3: avfilter: add xstack filter [12:49:23 CEST] <kierank> Jaex: [12:49:24 CEST] <kierank> January: [12:49:25 CEST] <kierank> Writing objects: 100% (15/15), 11.71 KiB | 0 bytes/s, done. [12:49:25 CEST] <kierank> Total 15 (delta 11), reused 4 (delta 2) [12:49:25 CEST] <kierank> remote: -Info- Update is fast-forward [12:49:25 CEST] <kierank> remote: -Deny- In e0727068bc6392625a015e47b3399d0c5be4f567: [12:49:25 CEST] <kierank> remote: -Deny- Trailing whitespace found in tests/api/api-h264-slice-test.c. [12:49:25 CEST] <kierank> remote: -Deny- Commit aborted, fix the issue and try again. [12:49:26 CEST] <kierank> remote: error: hook declined to update refs/heads/master [12:49:26 CEST] <kierank> To git@source.ffmpeg.org:ffmpeg [12:49:27 CEST] <kierank> ! [remote rejected] master -> master (hook declined) [12:49:27 CEST] <kierank> error: failed to push some refs to 'git@source.ffmpeg.org:ffmpeg' [12:50:07 CEST] <January> you didn't need to spam with the entire log [12:52:17 CEST] <cone-083> ffmpeg 03Josh de Kock 07master:0a055f463a60: lavc/h264dec: don't error out when receiving multiple IDR slices [12:52:18 CEST] <cone-083> ffmpeg 03Josh de Kock 07master:fb7925ba2fa1: fate: add api-h264-slice test [12:54:58 CEST] <kierank> and now we wait [13:41:57 CEST] <cone-083> ffmpeg 03Cameron Cawley 07master:22238d0b9440: avcodec: Implement Archimedes VIDC encoder/decoder [13:41:58 CEST] <cone-083> ffmpeg 03Cameron Cawley 07master:0e9c01fd87ae: avformat/rpl: Support files containing 8 bit PCM or VIDC audio [14:47:13 CEST] <kierank> January: bah 0a055f463a60af764c083fd0ba3112037557de4b commit message is wrong [14:47:23 CEST] <kierank> you keep conflating "slice threading" with chunks [14:48:43 CEST] <January> it was on the mailing list for a long time, and you pushed it, so not really sure what to say [14:49:53 CEST] <kierank> I'm telling you to be more precise in your wording in the future [14:55:33 CEST] <durandal_1707> oh, some popcorn time finally [14:56:05 CEST] <durandal_1707> rm -rf mediafoundation* [15:00:18 CEST] <January> durandal_1707: you dont want meadiafoundation? [15:02:25 CEST] <durandal_1707> i do not want mediafoundation,video/audio/toolbox, and other corporate crap [15:03:22 CEST] <atomnuker> the mf wrapper doesn't even output hardware frames [15:03:51 CEST] <atomnuker> you can't get good performance with it or good battery life, for that you need the built in decoders we have [15:04:21 CEST] <JEEB> for decoding you don't really need MF unless you need swdec [15:04:29 CEST] <JEEB> for hwdec we have dxva2 and d3d11va for windows [15:05:06 CEST] <atomnuker> the guy states one of the goals is to have a non-gpl h264 encoder and doesn't even mention evading patent fees [15:05:29 CEST] <January> atomnuker: m$ doesnt pay them? [15:07:14 CEST] <nevcairiel> Windows doesnt come with a h264 encoder [15:07:54 CEST] <JEEB> atomnuker: for non-GPL we have openh264 and in theory you can utilize binaries from cisco for that (although I think that requires dload or so since the install has to download it from cisco) [15:07:55 CEST] <nevcairiel> unless that changed recently [15:08:09 CEST] <JEEB> (latter to even evade the MPEG-LA stuff) [15:08:18 CEST] <JEEB> but yea (Ž4@) [15:08:31 CEST] <atomnuker> nevcairiel: the patch says there is: encoders: * video: h264_mf, hevc_mf [15:09:13 CEST] <atomnuker> January: I think they're included in the license for windows but you have to use them and not some other implementation if you don't want to pay patent fees [15:09:57 CEST] <JEEB> the distributor to end users pays so yea, if you distribute another thing that's your problem [15:10:34 CEST] <JEEB> which is why people have loved DShow and MF, you can kind of externalize the problem since you're just generally asking for some format in the framework and "I didn't distribute the thing that gives me this stuff" [15:10:38 CEST] Action: JEEB sighs [15:11:05 CEST] <atomnuker> will there be a CCCP release for mf :) [15:11:26 CEST] <JEEB> MF doesn't have real players etc [15:11:28 CEST] <nevcairiel> noone but MS themselfes really uses MF [15:11:41 CEST] <JEEB> yea, nevcairiel tried to make a player but it was a chick n' egg problem [15:11:50 CEST] <JEEB> nevcairiel: I've seen some people distribute MF filters but yea - it's rare as hell [15:14:05 CEST] <atomnuker> there's already a player, isn't there? windows media player [15:14:25 CEST] <JEEB> that is heavily hard-coded towards MS filters [15:14:28 CEST] <JEEB> even for DShow [15:14:50 CEST] <nevcairiel> not sure how it works with MF, but in DShow you can make it use any system filters [15:14:51 CEST] <JEEB> it seems to be like: hard-coded for MF if MS MF filters support something => fall back to DShow MS filters => fall back on 3rd party DShow filters [15:15:01 CEST] <JEEB> nevcairiel: yea unofficially you can tweak it [15:15:02 CEST] <nevcairiel> the MS filters just often have very high priorities [15:15:24 CEST] <JEEB> I thought there was something in addition to priorities [15:15:29 CEST] <JEEB> actual UUID things [15:15:40 CEST] <JEEB> don't remember, was long enough ago when I last loked into details of that tweak :P [16:21:34 CEST] <philipl> nevcairiel: FYI, I looked at fps cost of my cuda yadif when not doing frame doubling and it's negligible. Looking back I realise that I saw a 50% slowdown with frame doubling because the encoder was working twice as much. Need to test without encoding... silly me [16:37:36 CEST] <J_Darnley> can I generate a strobe video with lavfi easily? A video which alternates between black and white every frame? [16:42:24 CEST] <J_Darnley> maybe just two color sources and an interleave [16:42:32 CEST] <durandal_1707> J_Darnley: ffmpeg -f lavfi -i color=black -f lavfi -i color=white -lavfi hstack,stereo3d=sbsl:al,trim=duration=1 1.y4m [16:43:03 CEST] <JEEB> that's an interesting way... can't you just interleave two inputs? [16:43:09 CEST] <JEEB> frame-wise [16:43:25 CEST] <JEEB> without going through horizontal stacking and then stereo3d doing the interleaving :D [16:46:22 CEST] <J_Darnley> does interlave double the framerate? Two 30fps becomes 60 fps? [16:49:14 CEST] <durandal_1707> J_Darnley: ffmpeg -f lavfi -i color=black -f lavfi -i color=white -lavfi "interleave,settb=1/25,setpts=N,showinfo,trim=end_frame=100" 1.y4m [16:49:55 CEST] <J_Darnley> Thanks for the suggestions but I went with ffplay -f lavfi -i 'color=black:r=30 [a]; color=white:r=30 [b]; [a][b] interleav [16:49:59 CEST] <durandal_1707> without settb and setpts it is memory bomb for some reason [16:51:08 CEST] <durandal_1707> J_Darnley: did you just bombed your machine? [16:51:19 CEST] <J_Darnley> No [16:51:25 CEST] <durandal_1707> good :) [16:52:20 CEST] <J_Darnley> htop says ~7% cpu and 0.7% memory [16:53:39 CEST] <J_Darnley> Well, that didn't help anyway [16:56:10 CEST] <durandal_1707> what are you doing? [16:58:26 CEST] <atomnuker> throwing a rave? [16:58:48 CEST] <J_Darnley> Trying to work out what the heck kind of packing this dumb professional video format is doing [16:58:59 CEST] <philipl> nevcairiel: so I'd say it's ~12% slower to do yadif without rate doubling and then another ~12% to do doubling so ~25% overall [16:59:46 CEST] <durandal_1707> philipl: and comparing to CPU yadif? [17:01:35 CEST] <JEEB> durandal_1707: he only did bobbing in it IIRC, not actual internals of yadif [17:01:38 CEST] <JEEB> also > CUDA [17:02:11 CEST] <nevcairiel> he added the actual yadifing [17:02:18 CEST] <JEEB> ok [17:04:22 CEST] <philipl> I have real yadif. [17:04:51 CEST] <philipl> durandal_1707: it's a bit hard to do apples to apples. Doing CPU decoding + CPU yadif, the yadif makes it 50% slower [17:06:18 CEST] <philipl> Obviously, doing GPU decoding and copying back to do cpu yadif is way way slower. [17:44:47 CEST] <atomnuker> BBB: are packets in dav1d padded? [17:45:26 CEST] <BBB> no [17:45:51 CEST] <atomnuker> turns out AV_INPUT_BUFFER_PADDING_SIZE can be quite useful for entropy coders too :/ [17:46:39 CEST] <atomnuker> also OD_EC_LOTS_OF_BITS is an atrocious badly put out hack with a completely random value [17:47:12 CEST] <atomnuker> basically its normal to overread after the packet ends by calling ec functions, they're guaranteed to return 0 [17:47:28 CEST] <atomnuker> but every time you do that you call the refill function since the count is 0 [17:47:34 CEST] <atomnuker> the refill function will be a noop [17:48:10 CEST] <atomnuker> so if the refill function detects that it has reached the end of the buffer it'll put 0x4000 into the amount of bits and fool the ec into not calling refill [17:48:43 CEST] <atomnuker> why 0x4000? there's a whole chapter of war and peace in the libaom website which tells you "dunno, it was a nice number" [17:49:14 CEST] <atomnuker> it would have been more optimal to pick (1 << (sizeof(bits_counter) << 3)) - 1 [17:49:58 CEST] <BBB> happy to add padding if it helps [17:50:03 CEST] <BBB> you can set the rules if you rewrite it [17:50:05 CEST] <BBB> right? [17:50:07 CEST] <atomnuker> but still what an utterly pointless thing [17:50:31 CEST] <atomnuker> it avoids a 100% predicted branch by adding cryptic code [17:51:03 CEST] <atomnuker> rewrite it? already done, there's no original code left and its around 130 lines shorter than the original file [17:51:41 CEST] <atomnuker> just polishing it, if the buffer was padded it would remove a end of buffer check in the refill function [17:52:02 CEST] <atomnuker> since you never refill more than a few bytes at a time [18:05:14 CEST] <atomnuker> hmm, padding wouldn't work, you'd shift in non-zero bits at the end of each tile [21:50:31 CEST] <atomnuker> BBB: reopened PR, discussion area's a mess but gitlab sucks as much as anything else does except MLs [21:50:53 CEST] <atomnuker> speedup seems to be around 5% overall for the 1080p video I use for testing on my machine [21:51:24 CEST] <BBB> oh sweet [21:51:29 CEST] <BBB> I didnt expect a speedup [21:51:30 CEST] <BBB> but nice [21:52:05 CEST] <atomnuker> me neither, I'm even surprised I managed to get the 64 bit window working with a speedup, but I'm guessing it has to do with the refill function [21:52:30 CEST] <atomnuker> I could have saved on the buf_pos < buf_end check there because I couldn't make it overread ever but better safe than sorry [21:52:49 CEST] <atomnuker> then I believed it would have fit in ryzen's 5 instructions for 1 cycle loop thing [23:10:09 CEST] <BBB> sweet [23:10:37 CEST] <BBB> do you have any interest in experimenting with sse2 versions of update_cdf or read_symbol? [23:10:43 CEST] <BBB> (i.e. do 4 symbol checks at a time) [23:17:30 CEST] <BBB> atomnuker: any preferred reviewer for your patch? [23:17:40 CEST] <BBB> I can review it but maybe someone else knows this stuff better than me [23:18:07 CEST] <atomnuker> pinged unlord, he knows best [23:18:25 CEST] <atomnuker> absolutely, I'll do some simd for the update part first [23:18:52 CEST] <atomnuker> read_symbol is also simdable [23:18:52 CEST] <BBB> inline assembly ftw [23:18:59 CEST] <atomnuker> no need to [23:19:09 CEST] <BBB> oh you think the call overhead is minor? [23:19:21 CEST] <BBB> (I always assumed it was like cabac and itd be suitable for inline asm) [23:19:46 CEST] <atomnuker> nah, update_cdf doesn't use any internal msac stuff, its fully doable in asm [23:20:24 CEST] <atomnuker> as for msac_decode_symbol simd, I'll make it inline along with ctx_norm+ctx_refill [23:21:05 CEST] <atomnuker> then the core cdf part of it can be simd'd whilst still keeping the function calls down to 1 [23:26:29 CEST] <BBB> ok [23:26:32 CEST] <BBB> overall looks very nice [23:26:36 CEST] <BBB> I added 2 minor style comments [23:26:42 CEST] <BBB> but overall I feel it looks good [23:26:53 CEST] <BBB> how etensively has it been tested? do we need to spend some time testing it? [23:28:28 CEST] <BBB> I suppose if its a non-inline, you could also make it avx2 [23:28:39 CEST] <BBB> which allows updating 8 symbols at once [23:28:47 CEST] <BBB> or 16? [23:29:06 CEST] <JEEB> &34 [23:29:21 CEST] <BBB> they are only 2 bytes... [23:30:34 CEST] <atomnuker> BBB: I've tested it with some broken files and I haven't been able to trigger any of the asserts left nor trigger an overrun [23:30:47 CEST] <atomnuker> I think if it passes ubsan it should be fine [23:30:59 CEST] <BBB> and md5 of nonbroken files are identical right? [23:32:18 CEST] <atomnuker> yep [23:33:41 CEST] <BBB> cool [23:33:44 CEST] <BBB> very cool [23:50:40 CEST] <j-b> atomnuker: WoW! [23:52:25 CEST] <atomnuker> BBB: mate...? [23:52:34 CEST] <atomnuker> is _everything_ in x86 templated automatically [23:52:44 CEST] <BBB> no [23:52:47 CEST] <BBB> someone went overboard [23:53:03 CEST] <BBB> theyre just init functions [23:53:27 CEST] <BBB> I think someone hopes well do #if BITDEPTH == 8 & lots of code & # else & lots of code #endif [23:53:55 CEST] <BBB> I was planning to do if (), not #if, but I guess I missed something [23:53:59 CEST] <BBB> ohwell [23:54:01 CEST] <BBB> not so important [23:54:01 CEST] <atomnuker> I'm getting complaints that my init function is defined multiple times because I don't ifdef it for 8 and 10 bits [23:54:04 CEST] <BBB> its just init code [23:54:30 CEST] <BBB> void bitfn(bla_dsp_init_x86)(BlaDSPContext *const c) { .. } [23:54:35 CEST] <atomnuker> msac_init.c:(.text+0x0): multiple definition of `dav1d_msac_init_x86'; src/src@@dav1d_bitdepth_8@sta/x86_msac_init.c.o:msac_init.c:(.text+0x0): first defined here [23:54:37 CEST] <BBB> bitfn() will take care of suffic [23:54:44 CEST] <BBB> oh [23:54:45 CEST] <atomnuker> eh, ok [23:54:46 CEST] <BBB> uh [23:54:47 CEST] <BBB> hm [23:54:50 CEST] <BBB> oh right [23:54:51 CEST] <BBB> for msac [23:54:52 CEST] <BBB> hm... [23:54:57 CEST] <BBB> sorry I misunderstood you [23:55:04 CEST] <BBB> yes you want that non-templated [23:55:07 CEST] <BBB> sorry about that [23:55:18 CEST] <BBB> can you ask epirat? [23:55:21 CEST] <BBB> he knows how to do that [23:56:10 CEST] <atomnuker> no such nick [23:56:17 CEST] <atomnuker> (ePirat too) [23:56:59 CEST] <atomnuker> nvm, I'll do it myself, I see how in meson.build or whatever [23:57:32 CEST] <BBB> in #dav1d [23:57:39 CEST] <BBB> oh, hes asleep [23:57:40 CEST] <BBB> ohwell [23:57:58 CEST] <atomnuker> yep, works now [00:00:00 CEST] --- Sat Oct 27 2018

1 0

ffmpeg.log.20181026
by burek 27 Oct '18

27 Oct '18

[00:00:35 CEST] <furq> the input file only contains audio and video [00:00:38 CEST] <furq> so there's nothing to copy [00:40:24 CEST] <steve___> I'm in the weeds a bit. I'm trying to create a number of videos from different images and then concat with them some actual videos. [00:40:47 CEST] <steve___> Here is the output of ffprobe for one of the videos -- https://p.zsw.ca/XT [00:41:10 CEST] <steve___> err [00:41:44 CEST] <steve___> https://p.zsw.ca/SR [00:42:30 CEST] <steve___> Here is the output of ffprobe for the video I created using an image -- https://p.zsw.ca/EZ [00:46:12 CEST] <steve___> after concat'ing those two videos, I get these errors via 'mpv' -- https://p.zsw.ca/7R [00:46:41 CEST] <steve___> playing each video independently works fine [00:47:00 CEST] <steve___> I'm using ffmpeg version 3.3.2 [01:11:38 CEST] <bodqhrohro_> What is the best approach for integrating ffmpeg with custom OpenGL processing module? Maybe frei0r, or something standalone that reads/produces a stream? [02:02:32 CEST] <qwertymodo> @kepstin @iive and anybody else that helped me out yesterday, thanks. I managed to throw together a gimp python script that was able to un-blend the video frames I fed into it. Still needs some TLC to clean up the noise (and GIMP really needs to implement Python bindings for the GEGL functions, but that's another matter...), but I think this is at least a very solid starting point: https://twitter.com/qwertymodo/status/1055608341311811 [02:02:32 CEST] <qwertymodo> 584 [02:05:20 CEST] <iive> looks really good! [02:18:03 CEST] <iive> btw, there are some patterns that repeat and it might not be entirely because of the quantization. [02:18:40 CEST] <iive> some deinterlacers use data from more than 2 fields.e.g. 3 lines from one field picture and 3 from another... [02:19:26 CEST] <iive> that of course makes it a bit harder to revert... [02:19:36 CEST] <iive> anyway, gtg, keep the good work :D [02:20:47 CEST] <iive> and I have feeling some images might be easier to reconstruct by copy/pasting parts of the non-blended frames ;) [02:20:50 CEST] <iive> n8 [07:13:36 CEST] <JC_Yang> do anyone (re-)pickup ffserver(patch & fix) lately? [08:50:45 CEST] <superware> I'm trying to use FFmpeg libav to remux a network stream to a TS file, but when I play the file it doesn't start from the first written frame (av_interleaved_write_frame) but rather starts with a blank period which corresponds to the network stream start offset from that frame... any ideas? [09:43:14 CEST] <JC_Yang> do you have your ts processed correctly? [13:11:01 CEST] <TheWild> hello [13:11:12 CEST] <TheWild> ffmpeg -c copy ... [13:11:23 CEST] <TheWild> why I'm getting: Unknown decoder 'copy' ? [13:15:09 CEST] <th3_v0ice> superware: You can either set out_stream->start_dts = first_packet_dts; out_stream->start_time = first_packet_pts / time_base; or simply deduce first_packet_dts from each packet. [13:19:48 CEST] <TheWild> ah, s**t. Never mind. Order of arguments is important. [13:19:50 CEST] <iive> TheWild, position matters [13:20:05 CEST] <TheWild> ffmpeg -i input -c copy ... [13:21:46 CEST] <TheWild> iive: I ninja'd, but thanks anyway [13:22:10 CEST] <iive> yeh, i was like a second slower :P [13:54:41 CEST] <Harzilein> hmm [13:56:03 CEST] <Harzilein> i used to know more about this... but say i have an audio player that is relatively simplistic about seek points, it'll happily skip mp3 decoding until it can decode again, but display the time offset from what you entered as seek position [13:57:28 CEST] <Harzilein> i now want a list of seek offsets i can safely jump to without getting the wrong time (in seconds) displayed... do i need ffprope -show_packets? how do i find out if it's basically an "audio i-frame"? [13:57:52 CEST] <Harzilein> (maybe the right intuition is to use -show_frames? but that is for video, right?) [13:57:52 CEST] <JEEB> most audio players quickly parse through the whole file [13:58:11 CEST] <JEEB> so in lavf terms they just open a demuxer and get the offsets where to begin for packets [13:58:14 CEST] <JEEB> and the duration [13:58:22 CEST] <JEEB> this is for audio without containers, like mp3 [13:58:50 CEST] <JEEB> both video and audio output frames, just that there's generally a whole lot more audio frames than video frames in general :P [13:59:06 CEST] <JEEB> since a typical audio frame could contain 960 or 1024 samples, out of 48000 for example [14:03:00 CEST] <Harzilein> i was desperate so i started manually trying if something is a start position or an overshoot position. player is xmms2 0.8, file is http://rcveeder.net/clash/cotti30.mp3, i can tell it to seek to $(( (6 * 60) + 51 )) and while that might not be a correct position, $(( (6 * 60) + 52 )) will be an overshoot. how can i determine that fact from ffprobe output? [14:04:14 CEST] <Harzilein> it tells me key_frame for each of the frames :/ [14:04:26 CEST] <JEEB> in mp3 they all are [14:04:34 CEST] <JEEB> your problem is that the seek you're trying to do is done through guesstimation [14:04:52 CEST] <JEEB> which is why I say that your player has to use libavformat to first parse through the file [14:04:59 CEST] <JEEB> "index" it if you may [14:05:15 CEST] <JEEB> then you know the byte-wise indices [14:05:23 CEST] <JEEB> and you can use byte-wise seek [14:05:28 CEST] <JEEB> and get where you want [14:05:44 CEST] <Harzilein> really? a music player that specializes in music playing guesstimates the positions based on cbr? i'll have to dig into this, that can't be true :D [14:06:30 CEST] <JEEB> no, I mean mp3 doesn't have a container or index [14:06:35 CEST] <JEEB> it's raw mpeg-1 layer 3 [14:06:41 CEST] <JEEB> the only way to accurately seek in it is to index the file [14:06:45 CEST] <JEEB> to know where the audio packets are [14:07:09 CEST] <JEEB> and since in case of mp3 all packets are "you can start decoding here", you just need to know how many packets there are and where each of them begins [14:07:33 CEST] <JEEB> that way you can better estimate the length as well as do seeking [14:07:47 CEST] <JEEB> if you are using libavformat without first doing this sort of indexing and seeking based on duration [14:07:51 CEST] <JEEB> that is *guessing* [14:07:57 CEST] <JEEB> because that's all that libavformat at that point can do [14:08:07 CEST] <JEEB> I hope that's clear enough [14:08:09 CEST] <Harzilein> yeah, i get it. and i understand that xmms2 wants to unify streaming and file urls and may only do deferred indexing, but this seems wrong... [14:08:43 CEST] <JEEB> basically seeking without indexing an mp3 file can only work if the encode packets are all the same size [14:08:46 CEST] <JEEB> aka CBR [14:08:50 CEST] <JEEB> but a whole crapload of files are not [14:09:04 CEST] <JEEB> all the players including foobar2000 seem to just index the stuff :P [14:09:09 CEST] <JEEB> the FFmpeg APIs let you do that too [14:11:08 CEST] <JEEB> and that is why I'm happy that no-container audio didn't really take off outside of mp3 :P [14:11:20 CEST] <JEEB> it's a mess and herp derp [14:13:33 CEST] <Harzilein> so basically i'd need to extract xmms2 (mad-based? not sure) seeking formula in order to predict if it will seek correctly... but the assumption is that it's unlikely that it will converge to the vbr positions later anyway (again, the goal here is to get second precision, so i have _some_ hope) [14:14:02 CEST] <JEEB> well you'd have to dive into the code of that player and make sure it indexes non-containerized formats [14:14:08 CEST] <JEEB> otherwise the seeking will be all over the place :P [14:14:52 CEST] <JEEB> well not all over the place, but not necessarily where you wanted it to be [14:16:08 CEST] <Harzilein> JEEB: i'd be somewhat fine with that if i can have "segments" where at the start, the guessed offset happens to match the correct offset with second precision [14:16:44 CEST] <JEEB> that might or might not be possible, you'll have to see how the player you're talking about is actually doing it then :P [14:18:12 CEST] <Harzilein> JEEB: basically i want to insert timestamps into the text adventure transcript of an episode where they read that podcast. going from a transcript position to an mp3 position is "nice to have" but it'd not be crippling the concept if only a small subset of those points would be marked for that [14:18:34 CEST] <Harzilein> s/read that podcast/read that text adventure/ [14:19:26 CEST] <Harzilein> it'd be overkill to include ffmpeg in the if interpreter :D [14:20:26 CEST] <Harzilein> yeah, i'll look at that. maybe they are conservative about wether they want to do the indexing because they assume people use smb shares etc. [14:20:33 CEST] <Harzilein> but still have the functionality [14:21:14 CEST] <Harzilein> but once i have a mapping from second to guessed position, getting my "is it correct-ish" answer from ffprobe output should be easy, thanks. [14:23:31 CEST] <Harzilein> looking at xmms_mad_seek, now i remember, vbr working was about those xing headers... ffprobe can tell me if those are present, right? [14:23:55 CEST] <Harzilein> vbr seeking* [14:24:20 CEST] <Harzilein> bytes = xmms_xing_get_toc (data->xing, i) * (xmms_xing_get_bytes (data->xing) / 256); [14:24:23 CEST] <Harzilein> bytes = (guint)(((gdouble)samples) * data->bitrate / data->samplerate) / 8; [14:24:24 CEST] <Harzilein> :D [14:24:38 CEST] <Harzilein> +/* vs. */ [14:27:32 CEST] <JEEB> not sure what the header contains, as I prefer general solutions, and just parsing (not even decoding) mp3 is generally quick :) [14:30:22 CEST] <Harzilein> there's bound to be a xing header example in the samples collection *digs* [15:02:40 CEST] <TheAMM> Ello minus [15:03:53 CEST] <minus> well yes hello [15:26:25 CEST] <feedbackmonitor> Hi, I have a folder full of MTS files that I need converted to MOV, without quality loss, can anyone provide me with the command to do this? [15:28:25 CEST] <JEEB> remuxing in that case [15:28:32 CEST] <JEEB> keeping the original video/audio tracks [15:34:27 CEST] <feedbackmonitor> JEEB, So ffmpeg is not the tool fo rthis? [15:39:02 CEST] <durandal_1707> feedbackmonitor: do you know what remuxing means? [15:39:15 CEST] <feedbackmonitor> durandal_1707, Swapping containers? [15:40:18 CEST] <JEEB> feedbackmonitor: FFmpeg can do it just fine, just noted what you have to do for it keyword-wise :P [15:41:20 CEST] <feedbackmonitor> Can you perhaps provide the command for this, to convert a folder full of MTS to MOV format files? [15:42:07 CEST] <feedbackmonitor> Though maybe this is not a good idea, the MTS format has different frame rates... [15:45:18 CEST] <feedbackmonitor> I guess there should be a frame rate adjustment too? [15:48:46 CEST] <durandal_1707> why? [16:00:06 CEST] <Harzilein> JEEB: heh, turns out the file is actually cbr, but it has a xing header present which makes the seeking _more_ coarse :D [16:01:36 CEST] <Harzilein> JEEB: so i have a marginal/arbitrary need to use ffprobe to determine the length in seconds, then divide that by 100 and i have the duration of each segment i can seek to [16:01:59 CEST] <JEEB> I don't know why you're just using ffprobe instead of fixing teh application you're using? [16:02:05 CEST] <JEEB> since you clearly are not using the FFmpeg APIs [16:09:43 CEST] <Harzilein> JEEB: i'd use ffprobe for the data but i'm fine with using an inferior player for the implementation if that means i can be lazier with the code [16:12:23 CEST] <JEEB> I really don't get it :P [16:18:45 CEST] <feedbackmonitor> Sorry, my computer locked up and culd not get any response, if there was any [16:20:42 CEST] <durandal_1707> feedbackmonitor: try: ffmpeg -i input -c copy out.mov [16:22:44 CEST] <lowin> Hello. Is there a filter to average every couple of video frames into a single frame? I think it's called long exposure. but I can't find any info on it that doesn't involve splitting the video into frames and then joining them in multiple steps [16:23:20 CEST] <durandal_1707> lowin: tmix [16:23:37 CEST] <lowin> durandal_1707, oh awesome [16:27:08 CEST] <feedbackmonitor> durandal_1707, There are two things; 1) I have a folder full of MTS files, 2) it is my understanding that ffmpeg needs to be instructed about the quality of output, so I want the results to be decent, if possible. [16:27:35 CEST] <feedbackmonitor> So for the first part, can I tell ffmepg to recursively go through the folder and convert? [16:28:17 CEST] <feedbackmonitor> The second, can I tell ffmpeg to output in decent quality, nothing too crazy? [16:28:33 CEST] <relaxed> 1) stream copies are lossless 2) script ffmpeg to do it using a shell [16:28:49 CEST] <relaxed> mixed those up :) [16:30:39 CEST] <feedbackmonitor> I guess the simpler solution is to find a video editing program that can handle MTS formats [16:30:46 CEST] <feedbackmonitor> ; - b [16:31:56 CEST] <relaxed> ffmpeg can handle remuxing to mov if you have the space [16:32:24 CEST] <feedbackmonitor> Yeah, it's the scripting part I am not up on [16:32:39 CEST] <feedbackmonitor> I need to take a few classes in scripting before I can do this project [16:32:47 CEST] <feedbackmonitor> : - D [16:33:22 CEST] <relaxed> is this linux, mac, or windows? [16:33:39 CEST] <feedbackmonitor> relaxed, linux [16:34:27 CEST] <relaxed> in the directory conatining the videos run, for i in *ts; do ffmpeg -i "$i" -c copy "${i%.*}".mov; done [16:38:31 CEST] <steve___> I have two videos that I'd like to join. One of the two videos I created from a jpg using this command: ffmpeg -y -loglevel -8 -framerate 1 -i stills/wte01-%02d.jpg -c:v libx264 -r 30 -pix_fmt yuv420p 02-wte01.m4v [16:39:04 CEST] <steve___> How do I go about concat'ing the two videos. I don't mind reencoding. [16:39:57 CEST] <relaxed> steve___: you'll probably need to use the concat filter: https://trac.ffmpeg.org/wiki/Concatenate [16:44:14 CEST] <steve___> relaxed: yeah i was trying that as well as -filter_complex but I was running ashore [16:44:46 CEST] <relaxed> pastebin.com your complete command and output for help [16:44:51 CEST] <steve___> relaxed: At first I tried 'ffmpeg -y -f concat -safe 0 -i ./meta -c copy' It concat the two video but there is a pause between video1 and video2 [16:45:38 CEST] <steve___> here is the output of that command -- https://p.zsw.ca/6D [16:47:11 CEST] <steve___> ffprobe of video1 -- https://p.zsw.ca/TG video2 -- https://p.zsw.ca/RU [16:50:24 CEST] <relaxed> they're different framerates [16:52:38 CEST] <steve___> relaxed: what is my best bet to get around that? [16:53:52 CEST] <steve___> relaxed: Just so I'm with you, you're talking about 29.97 fps vs 30 fps, right? [16:54:42 CEST] <relaxed> yes, try the concat filter [16:54:48 CEST] <microcolonel> Is there (mature) support for producing Apple CAF files? [16:55:03 CEST] <microcolonel> I want to play Opus streams on Safari and for some reason they haven't implemented standard containers for this [16:55:22 CEST] <microcolonel> -f caff should work? [16:55:46 CEST] <JEEB> yes, opus is there it seems [16:55:53 CEST] <JEEB> in libavformat/cafenc.c that is [16:55:54 CEST] <steve___> relaxed: for video2, I set "-r" to 29.97 and it seem to work. Is this wise? [16:56:10 CEST] <feedbackmonitor> relaxed, I ran that command, but no results came from that but "No such file or directory", oh well. [16:56:23 CEST] <JEEB> I don't know what you ran [16:56:48 CEST] <JEEB> ffmpeg -i welp.opus -c copy out.caf [16:56:52 CEST] <JEEB> should "just work" in theory [16:57:16 CEST] <steve___> feedbackmonitor: cd to the dir with the ts files and then run the command [16:57:18 CEST] <JEEB> forcing the muxer only is required when there's multiple things with the same extension [16:58:51 CEST] <feedbackmonitor> steve___, I did indeed do that, and they are MTS files [16:59:40 CEST] <steve___> then maybe -- for i in *TS; do ffmpeg -i "$i" -c copy "${i%.*}".mov; done [17:01:05 CEST] <feedbackmonitor> steve___, https://pastebin.com/ZZYDLW3u [17:01:53 CEST] <feedbackmonitor> steve___, wait, https://pastebin.com/HXjddiuE [17:01:55 CEST] <feedbackmonitor> there [17:03:20 CEST] <steve___> feedbackmonitor: ah, you have to paste the entire command. [17:04:20 CEST] <feedbackmonitor> steve___, I suspect everything after the word "do" is the command [17:04:45 CEST] <steve___> everything after my "--" is the command [17:05:35 CEST] <microcolonel> hmm [17:05:51 CEST] <microcolonel> JEEB: I guess I'll find out when I get a Safari user to confirm [17:06:07 CEST] <microcolonel> thx [17:07:29 CEST] <feedbackmonitor> steve___, Ahh, I see. Thanks. But the results are highly pixelated. [17:07:35 CEST] <feedbackmonitor> : - ( [17:08:28 CEST] <feedbackmonitor> Then I suspect that FFMPEG really needs to convert these files proper [17:09:29 CEST] <durandal_1707> feedbackmonitor: pixelated? -c copy does not change video/audio [17:10:21 CEST] <feedbackmonitor> durandal_1707, That's the results I have, even the output sizes are smaller [17:11:49 CEST] <durandal_1707> you are doing something wrong [17:12:12 CEST] <feedbackmonitor> durandal_1707, Sure [17:12:54 CEST] <feedbackmonitor> I just copy and pasted the commands, but if there is something better, I am open to pasting that too [17:13:06 CEST] <JEEB> microcolonel: I think someone tested it back when the patch was posted [17:13:09 CEST] <JEEB> so in theory it should work [17:17:03 CEST] <sakrecoer> hi! is it possible to use ffmpeg to inject the necessary metadata for webservices to detect that the video is a 360 video with equirectangular projection ? [17:18:28 CEST] <sakrecoer> my websearch fu is failing me miserably... i know exiftool can add some xmp stuff, but it seems most webservices rquire the information on the codec level (? bare with me, i'm not very good at all this) [17:19:18 CEST] <durandal_1707> no [17:19:38 CEST] <durandal_1707> but there are scripts that do it [17:19:40 CEST] <feedbackmonitor> durandal_1707, Here is the output of what happened: https://pastebin.com/EFmKyFNJ [17:21:04 CEST] <sakrecoer> thanks durandal_1707 :) i'm aware of googles spatial thing, but i can't find anything that runs on GNU+Linux. any links to such scripts, please? [17:22:31 CEST] <durandal_1707> feedbackmonitor: make sure to compare with input file [17:23:40 CEST] <feedbackmonitor> durandal_1707, What do you mean? [17:24:06 CEST] <feedbackmonitor> Are you saying that the input file is pixelated therefore the output file is pixelated? [17:24:47 CEST] <durandal_1707> feedbackmonitor: yes, compare visually both input and output [17:25:20 CEST] <feedbackmonitor> durandal_1707, The input is very nice, the output is very shitty. [17:26:04 CEST] <durandal_1707> feedbackmonitor: take screenshot of both of them? [17:26:43 CEST] <feedbackmonitor> durandal_1707, Sure [17:34:36 CEST] <sakrecoer> ok if anyone else wonders, this helped me imensly :) https://github.com/google/spatial-media/tree/master/spatialmedia [17:35:00 CEST] <sakrecoer> thanks for you hint durandal_1707 it made me find that thing :) [17:48:57 CEST] <feedbackmonitor> durandal_1707, You are correct and I am not correct. I double checked and I had an issue copying my originals to another drive for conversion. The MTS copies were corrupted so I copied the originals over again and this time it looks okay. [17:49:04 CEST] <feedbackmonitor> durandal_1707, Thanks for your patience. [17:49:46 CEST] <durandal_1707> feedbackmonitor: just dont overwrite originals [17:50:17 CEST] <feedbackmonitor> durandal_1707, I have the originals backed up on another drive. : - ) [17:51:48 CEST] <feedbackmonitor> When i filmed at the time, I did not fully understand my camera and recorded in the MTS format, now I know my camera a little better. It records best at MOV, which is less shitty [17:53:51 CEST] <feedbackmonitor> So I have all this footage that should have been shot at MOV and I cannot re-shoot so am stuck with the MTS [18:01:15 CEST] <furq> feedbackmonitor: if the quality of the actual streams is the same then it's probably better to use ts [18:01:38 CEST] <furq> mp4/mov etc will be unplayable if the recording gets interrupted and the camera can't write the header [18:02:23 CEST] <feedbackmonitor> furq, The MOV has a higher quality out put. The file sizes are MUCH LARGER. [18:02:44 CEST] <furq> that sounds fun [18:02:51 CEST] <furq> i guess just don't drop the camera then [18:03:11 CEST] <feedbackmonitor> furq, Whereas an interview segment in MTS is under a gig, in MOV that same segment is several gigs., A half hour interview is 32 gigs in MOV whereas it may be two gigs in MTS [18:04:04 CEST] <feedbackmonitor> Oddly, even though MTS is smaller, it looks really good [18:04:30 CEST] <furq> yeah the output you pasted says it's 1080p h264 [18:04:44 CEST] <furq> so i don't know what the mov would be recording that's so much bigger or better quality [18:04:47 CEST] <furq> unless it's prores or something [18:04:58 CEST] <feedbackmonitor> furq, There are various h264 apparently [18:05:15 CEST] <feedbackmonitor> All the MTS and MOV are h264 [18:05:23 CEST] <furq> weird [18:05:24 CEST] <feedbackmonitor> with different levels of quality [18:05:39 CEST] <feedbackmonitor> furq, I have a Panasonic GH3 [18:43:21 CEST] <fsphil> any handy media files on the net with variable frame rates, or other unusual but valid setups I could use to test with? [18:44:03 CEST] <furq> https://samples.ffmpeg.org/archive/extension/mp4/mov+mpeg4+aac++vfr.mp4 [18:44:06 CEST] <furq> and samples.ffmpeg.org in general [18:44:25 CEST] <fsphil> ah brilliant. thanks [21:20:22 CEST] <alimiracle> hi [21:20:35 CEST] <alimiracle> its this concatenation method performs a re-encode or not? [21:20:44 CEST] <alimiracle> ffmpeg -f concat -i concat.txt -c:v copy merged.mp4 -y [21:21:41 CEST] <JEEB> not re-encoding but IIRC that can go wrong in o9k ways if you happen to hit it with input it wasn't meant to handle by the original author [21:21:45 CEST] <JEEB> or wait [21:21:51 CEST] <JEEB> you are only specifying copy for video [21:21:58 CEST] <JEEB> so everything that is not video is getting default re-encoded [21:22:16 CEST] <JEEB> (audio and subtitles) [21:24:12 CEST] <alimiracle> <JEEB> I nede dont re-encoded everything [21:27:00 CEST] Last message repeated 1 time(s). [21:28:35 CEST] <JEEB> come back when you can make that into an English sentence, thanks [22:55:22 CEST] <alimiracle> hi [22:55:25 CEST] <alimiracle> I'm concatenate two mp4 files using ffmpeg [22:55:30 CEST] <alimiracle> I'm use this [22:55:37 CEST] <alimiracle> ffmpeg -f concat -i mylist.txt -c copy output [22:55:44 CEST] <alimiracle> its this method dont performs a re-encode to all data in file?? [22:58:35 CEST] <alimiracle> if this method performs a re-encode to sum data in file?? [22:59:12 CEST] <alimiracle> can you gave me the tru method [00:00:00 CEST] --- Sat Oct 27 2018

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