December 2018 - Ffmpeg-devel-irc

ffmpeg-devel.log.20181215
by burek 16 Dec '18

16 Dec '18

[00:06:27 CET] <cone-232> ffmpeg 03Paul B Mahol 07release/4.1:a4ddc3c9fc60: avfilter/vf_overlay: fix filtering with negative y [00:56:02 CET] <cone-232> ffmpeg 03Carl Eugen Hoyos 07master:641d52152f6a: lavc/g729dec: Support stereo streams. [00:56:03 CET] <cone-232> ffmpeg 03Carl Eugen Hoyos 07master:011c9112a020: lavc/g729dec: Cosmetics, fix indentation after last commit. [03:12:05 CET] <jamrial> durandal11707: your recent changes seem to have broken a bunch of mxf tests [03:40:35 CET] <philipl> BtbN: Did you see that patch for adding nv12 to transpose_npp? [03:40:46 CET] <philipl> From 2nd of nov. I missed it at the time [10:00:55 CET] <cone-730> ffmpeg 03Paul B Mahol 07master:be60dc2145ba: avutil/avsscanf: do not use long double functions [11:56:29 CET] <BtbN> philipl, yes, I even replied to it. [11:57:07 CET] <BtbN> I intentionally removed that functionality from the original patch, as it would duplicate large amounts of code with scale_npp [11:57:28 CET] <BtbN> And since scale_npp can be used as pure format conversion filter, I don't see any reason why transpose_npp should have the same code [12:24:19 CET] <philipl> Ah. I missed that [19:00:45 CET] <Compn> is there a way to force the codec tag format for wav output in ffmpeg? similar to -fafmttag in mencoder hmmm [19:00:57 CET] <Compn> oh yeah -atag [19:00:59 CET] <Compn> nice [00:00:00 CET] --- Sun Dec 16 2018

1 0

ffmpeg.log.20181215
by burek 16 Dec '18

16 Dec '18

[00:00:08 CET] <birdboy88> kepstin: it varies. usually unnoticable, but sometimes 0.5 sec ca [00:00:34 CET] <kepstin> main thing i can think of there is that you might be getting accumulated rounding error in the trims, since they can only trim to frame accuracy [00:00:49 CET] <kepstin> so finding a way to work with absolute instead of relative timings could help [00:00:57 CET] <birdboy88> kepstin: yes, that's an interesting thought [00:01:11 CET] <monery> I had that problem with cuda accelerated encoding some time back, solution for me back then was to make lots of smaller videos then splice them together at the end [00:01:54 CET] <birdboy88> monery: you mean when you recorded with a video card? how did you make many small videos from that? Shadowplay has almost no settings... [00:02:08 CET] <monery> i use ffmpeg from the linux terminal [00:02:13 CET] <monery> so I did it via the command line [00:02:38 CET] <kepstin> well, that's a very different use case from live windows game capture... [00:02:47 CET] <monery> ffmpeg website had the infomation on how to extract a specific part of the video out of a large file [00:03:19 CET] <monery> its a file not a stream right? [00:03:25 CET] <monery> arg door [00:03:25 CET] <monery> brb [00:03:28 CET] <birdboy88> yes, its a file [00:03:41 CET] <birdboy88> output of NVIDIA shadowplay, ca 1 hour gameplay [00:03:49 CET] <birdboy88> default shadowplay settings [00:03:58 CET] <kepstin> birdboy88: note that it *is* possible to do all of your work from multiple commands with a single ffmpeg command and complex filter chain, but it would probably be a pain to write. [00:04:35 CET] <kepstin> (and i'm not sure whether or not it would help with a/v sync) [00:04:41 CET] <birdboy88> kepstin: do you think that would resolve the sync issue? [00:04:54 CET] <birdboy88> ok [00:05:58 CET] <monery> not even sure how to script it in Windows, I know there is some syntax errors... [00:06:04 CET] <monery> not errors, differences [00:06:24 CET] <kepstin> birdboy88: first thing i'd try is adding a "fps=30" (or something) filter in the filter chain when extracting the clips, which might workaround issues due to large frame timing gaps when calculating the trims [00:07:10 CET] <kepstin> use a fps value that matches what you want the final rendered video to be [00:07:39 CET] <monery> kepstin, is there anything you don't know about ffmpeg? [00:07:40 CET] <monery> lol [00:08:36 CET] <birdboy88> kepstin: could you please tell me where excately I should put fps=30? [00:09:06 CET] <kepstin> birdboy88: it would have to be added to the -filter_complex, with [0:v] as input [00:09:29 CET] <kepstin> (you'd also want to give the output pad a name and -map that) [00:09:54 CET] <kepstin> although note that this means you can't use -vcodec copy... [00:10:00 CET] <kepstin> oh, wait, you're using -vcodec copy [00:10:04 CET] <kepstin> that explains everything. [00:10:51 CET] <kepstin> when you're using -vcodec copy, the start time (set with -ss) is rounded to the nearest keyframe [00:10:54 CET] <kepstin> it's not exact [00:11:07 CET] <kepstin> depending on the keyframe interval, this will result in possibly quite large shifts [00:11:37 CET] <kepstin> (also, your commands are applying audio filters on commands with -an, which is confusing/contradictory) [00:12:13 CET] <Skandalist> Why do I get inaccurate video bitrate. For example, while using -b:v 1000k, I get 997kbit/s? [00:12:23 CET] <kepstin> Skandalist: looks pretty accurate to me [00:12:29 CET] <birdboy88> so the problem is that the audio temporary clips are not being extracted from the same excat timepoints? [00:13:21 CET] <Skandalist> <kepstin> yes, but in kdenlive it is exact [00:13:47 CET] <kepstin> birdboy88: yeah, your audio is being re-encoded to wav so it's being cut sample-accurate, but the video's not being precisely cut. [00:15:13 CET] <kepstin> Skandalist: you didn't say anything about codecs, but in general using the -b:v option sets a target "average" bitrate, and it's up to the encoder to try to get as close as possible over longer intervals but raising/lowering quality. It's basically impossible to hit right on with a single-pass encode, although most encoders can get pretty close with 2-pass. [00:15:28 CET] <Skandalist> h264 [00:16:21 CET] <birdboy88> kepstin: so I need to use slow seek (?) to extract video accurately? Or somehow extract audio only where there are video keyframes? [00:17:19 CET] <kepstin> birdboy88: i don't know how to extract audio starting at video keyframes with ffmpeg cli. You're already doing slow seek, which doesn't help (you should move the -ss option to before the -i option to speed it up) [00:17:35 CET] <kepstin> if you want accurate video cutting when saving to a file, you have to re-encode the video [00:17:58 CET] <kepstin> (doing this in a single ffmpeg command means you don't have to save to a file, so you can avoid the issue) [00:18:29 CET] Action: kepstin is off for a bit now [00:18:33 CET] <birdboy88> kepstin: [00:18:36 CET] <birdboy88> Ok thanks! [00:19:09 CET] <monery> kepstin, you don't have insight to my issue by chance do ya? [00:19:23 CET] <monery> [h264_nvenc @ 0x55b3c30d5540] Cannot load libnvidia-encode.so.1 [00:20:51 CET] <stephenwithav> monery: do you have a libnvidia-encode.so.1 file? [00:22:32 CET] <monery> I thought I did [00:22:40 CET] <monery> let me search the machine real quick [00:24:13 CET] <stephenwithav> If not, apt-file search it, then install that package. [00:24:27 CET] <monery> slow disk... [00:24:46 CET] <monery> hmm [00:24:48 CET] <monery> wtf [00:24:57 CET] <stephenwithav> Not found? [00:24:57 CET] <monery> oh, I know why I thought it was there... crap [00:25:10 CET] <monery> I think I was in the wrong ssh when I did the search... [00:27:03 CET] <monery> ... [00:27:08 CET] <monery> I am such a spoon [00:27:41 CET] <monery> ... so its encoding now... [00:27:52 CET] <monery> thanks stephenwithav for making me look a 2nd time [00:28:41 CET] <stephenwithav> yw. You were simply on the wrong server the entire time? [00:30:24 CET] <monery> ya [00:30:34 CET] <monery> I built one out of some scrap just to make the work go faster [00:30:44 CET] <monery> so I got two of em currently [00:31:33 CET] <stephenwithav> Been there myself many times. [00:32:17 CET] <monery> ... [00:32:27 CET] <monery> thats what I get for not smoking enough weed [00:33:06 CET] <Skandalist> <kepstin> Thank you [03:01:28 CET] <d3bug> having trouble converting from h.264 to hevc. Command: ffmpeg -hwaccel vaapi -hwaccel_device /dev/dri/renderD128 -hwaccel_output_format vaapi -autorotate 0 -i infile.mkv -vf crop=1920:800:0:140 -vcodec hevc_vaapi -qp 20 -map_chapters 0 -bf 0 -acodec libvorbis outfile.mp4 [03:01:47 CET] <d3bug> result: "Impossible to convert between the formats supported by the filter 'graph 0 input from stream 0:0' and the filter 'auto_scaler_0'" [03:02:02 CET] <d3bug> when I remove the crop parameter it works. [03:02:08 CET] <d3bug> but i want to crop it [03:02:41 CET] <iive> d3bug, you need filter that can work with hardware surfaces, especially the ones used by vaapi [03:02:51 CET] <iive> maybe somebody here will know more [03:03:00 CET] <d3bug> thats what i am hoping [03:04:31 CET] <furq> d3bug: it's not ideal but -vf hwdownload,crop=1920:800:0:140,hwupload [03:05:04 CET] <d3bug> furq: why is that not ideal? does it do something nasty with the result? [03:05:16 CET] <furq> it copies everything from vram to system memory and back again [03:05:47 CET] <Niklink> I'm trying to install libvmaf for use with ffmpeg, but I haven't been able to do so. I can't find any suitable instructions anywhere [03:05:47 CET] <d3bug> aaah [03:05:53 CET] <d3bug> so a performance hit [03:05:55 CET] <furq> there's a scale_vaapi filter that does it on the gpu but i don't see a crop_vaapi [03:06:05 CET] <furq> i've never used vaapi though so shrug [03:07:47 CET] <iive> d3bug, https://trac.ffmpeg.org/wiki/Hardware/VAAPI has some transcoding options, it uses vaapi_scale, instead of hwdownload/upload [03:07:48 CET] <d3bug> furq: I get this "[hwupload @ 0x563b93efc7a0] A hardware device reference is required to upload frames to. [Parsed_hwupload_0 @ 0x563b93efbf40] Query format failed for 'Parsed_hwupload_0': Invalid argument" [03:08:29 CET] <furq> -filter_hw_device [03:09:17 CET] <furq> https://trac.ffmpeg.org/wiki/Hardware/VAAPI#DeviceSelection [03:09:29 CET] <furq> or maybe you can just use -vaapi_device instead of -hwaccel_device [03:14:31 CET] <d3bug> I just tried this command: "ffmpeg -init_hw_device vaapi=gpu:/dev/dri/renderD128 -hwaccel vaapi -hwaccel_device gpu -hwaccel_output_format vaapi -autorotate 0 -i infile.mkv -filter_hw_device gpu -filter_complex hwdownload,crop=1920:800:0:140,hwupload -vcodec hevc_vaapi -qp 20 -map_chapters 0 -bf 0 -acodec libvorbis outfile.mp4" [03:14:49 CET] <d3bug> it seems to have locked up [03:15:29 CET] <d3bug> i had to kill the process [03:17:09 CET] <jafa> ffmpeg cli - what is the preferred way to map 2 audio channels from the input audio track to create a stereo output track. the input track is 4 channel and I want to grab 0.0.0 and 0.0.1... drop the other two channels [03:17:43 CET] <jafa> i have run tests mapping a single channel and that works [03:25:06 CET] <d3bug> I guess nobody has any ideas... lol [03:33:26 CET] <d3bug> anyone here use vaapi and crop together? [03:40:46 CET] <d3bug> such a simple task.... you would think other people have figured this out... [08:56:29 CET] <logicWEB> hey :-) I was just poking through some on video and found a file captured from a PVR over Firewire many years ago, and when I play it back, it looks like all the fields throughout the file are shifted by one, i.e. the first frame's top field is missing, and its bottom field gets interpreted as its top field and combined with the second frame's top field, and then the second frame becomes the second bottom plus the third top, etc. [08:56:39 CET] <logicWEB> I haven't investigated super closely but going through frame by frame that's what it looks like [08:58:36 CET] <logicWEB> is there a filter that can correct this? the closest I saw in the docs was fieldhint, but that would require me to hint _every single frame_, if I'm reading correctly, so for this video it looks like that'd be a file with over 40,000 identical lines in it [09:00:58 CET] <logicWEB> *poking through some _old_ video [09:09:13 CET] <furq> logicWEB: separatefields? [09:12:40 CET] <furq> oh wait there's a filter specifically for this [09:12:43 CET] <furq> https://www.ffmpeg.org/ffmpeg-filters.html#phase [09:13:04 CET] <logicWEB> hmm, that seems like it would get it half-way there -- it turns a frame with two fields into two frames with one field each, so then if the first one were discarded and then they were recombined... but there's no "combinefields" filter?? [09:13:16 CET] <logicWEB> oh neat, checking that out [09:13:16 CET] <furq> well you'd do separatefields and then tinterlace or something [09:13:19 CET] <furq> but yeah phase looks better [09:13:28 CET] <furq> you probably also want to force your player tff/bff [09:13:35 CET] <furq> just to make sure that you actually have the problem you think you do [09:15:06 CET] <logicWEB> sweet, `phase=b` fixes the problem :-D [09:15:27 CET] <logicWEB> single-stepping, the fields are perfectly lined up even under high motion [09:15:35 CET] <logicWEB> so that's exactly what I needed, thanks! :-D [12:11:41 CET] <ed8> hello, I'm working on a project where I have to extract hundred of small clips from a source video [12:12:08 CET] <ed8> to do so I use time from a subtitle file (to get start and end) [12:13:00 CET] <ed8> my extracted files do not seems to strcitly respect the duration [begin,end] time I gave [12:15:06 CET] <ed8> To extract I use -acodec copy -vcodec copy as seen here: https://github.com/parlr/lsf-data/blob/master/scripts/split-laura-video.bas… [12:15:28 CET] <furq> ed8: you can only cut on keyframes if you're stream copying [12:17:01 CET] <ed8> furq: I should apologize as I have little to no knowledge about video editing [12:18:20 CET] <furq> some video frames are keyframes, most (usually) aren't [12:18:26 CET] <furq> and a video has to start on a keyframe [12:18:50 CET] <furq> so when you cut a clip out, if you're not reencoding, ffmpeg will just seek to the nearest keyframe to the timestamp you gave with -ss [12:19:06 CET] <JEEB> there's keyframes and then there's random access points. difference being about the promises made for frames after ;) [12:19:25 CET] <JEEB> if you cut at a keyframe there's no promise of you being able to decode the frames that pop up later [12:19:29 CET] <furq> yeah i strictly mean IDR/IRAP/whatever [12:19:48 CET] <JEEB> yea, the FFmpeg API using the wording "keyframe" for it is unfortunate [12:19:56 CET] <JEEB> and thus leads to me always commenting from the side like this :P [12:20:17 CET] <JEEB> or well, "keyframe" I think was OK to use in the MPEG-2 and early MPEG-4 (Part 2) era? [12:20:21 CET] <furq> anyway the summary is you either have to live with inaccurate cuts or reencode the video [12:20:28 CET] <furq> you should be able to copy the audio [12:20:29 CET] <JEEB> then after MPEG-4 AVC it became less simple [12:20:51 CET] <JEEB> since you got IDR and you got SEI+I for open gop [12:21:24 CET] <ed8> furq: so no way to do it in 2 steps (clip out, then encode) if I want to precisely respect the specified time point ? [12:22:04 CET] <furq> well you could just do that in one step [12:41:45 CET] <ed8> furq: ok, I'm refactoring my tests to update my code [12:42:24 CET] <furq> also you might want to look into the segment muxer [12:42:34 CET] <furq> specifically -segment_times [15:07:20 CET] <pink_mist> so, I'm trying to use ffmpeg to mux an audio stream and a video stream downloaded from youtube with youtube-dl ... youtube-dl failed at making ffmpeg mux them, and I'm not having any better luck ... ffprobe of the audio file says: https://p.0au.de/b66e715 [15:08:32 CET] <pink_mist> is ffmpeg unable to deal with this format? or do I simply need to install an optional library and recompile ffmpeg? [15:11:59 CET] <relaxed> mux into which container? [15:12:27 CET] <durandal_1707> pink_mist: recompile with libaom or libdav1d for decoding that [15:12:43 CET] <pink_mist> ahh, thanks! [15:12:52 CET] <pink_mist> relaxed: mkv [15:13:24 CET] <relaxed> right, I forgot we don't have a native decoder [15:14:12 CET] <pink_mist> seems libdav1d is not available, so libaom it is [15:16:03 CET] <furq> pink_mist: mkvmerge will do it as of about two months ago [15:18:07 CET] <pink_mist> furq: has that version been released? I downloaded the latest windows version of mkvtoolnix just 15 minutes ago, and it did not work either ... perhaps that version is older than 2 months though? [15:19:03 CET] <kepstin> wait, google's putting av1 stuff on youtube now? wow. [15:19:20 CET] <kepstin> i guess they're the only people with enough compute power to actually encode a whole video with it :/ [15:19:33 CET] <pink_mist> it's just the audio [15:20:02 CET] <pink_mist> according to the youtube-dl format list at least [15:20:02 CET] <kepstin> that's the video file in your ffprobe [15:20:22 CET] <kepstin> you might need a newer youtube-dl, it might have the format wrong because av1 is very new [15:20:50 CET] <pink_mist> ah, that would be annoying [15:20:54 CET] <pink_mist> let me try [15:21:40 CET] <kepstin> pink_mist: for reference, https://github.com/rg3/youtube-dl/issues/17506 [15:21:52 CET] <kepstin> fixed nov 8 [15:23:36 CET] <pink_mist> right, my youtube-dl is certainly older than that :P [15:23:57 CET] <pink_mist> thanks, guess I'll need to redownload an actual audio format then =) [15:29:17 CET] <pink_mist> ok, new youtube-dl solved it :P [15:29:26 CET] <pink_mist> thanks kepstin :) [15:29:56 CET] <kepstin> yep, important to keep youtube-dl updated, because sites change all the time :) [15:39:31 CET] <pink_mist> (I got libaom installed anyway, so might as well recompile ffmpeg with it too) [15:59:29 CET] <Niklink> I'm trying to install libvmaf for use with ffmpeg, but I haven't been able to do so. I can't find any suitable instructions anywhere [16:00:51 CET] <JEEB> I'm not sure if the vmaf people broke compilation again. they've been sending patches every now and then when they break their API [16:01:21 CET] <Niklink> I didn't get to the 'compilation' part [16:02:51 CET] <Niklink> or if you know of any other tool to score videos based on perceptual quality, that could also work [16:05:33 CET] <JEEB> even VMAF can't be blindly re-used since it's been 100% trained on netflix's content and use case [16:05:46 CET] <JEEB> of course it's a +1 metric, and that's great [16:06:00 CET] <JEEB> also metrics you always have to remember are not really ever "perceptional" [16:06:14 CET] <Niklink> what if I had a bunch of similar videos [16:41:37 CET] <relaxed> Niklink: if you're using linux my static builds have libvmaf 1.3.9 https://www.johnvansickle.com/ffmpeg/ [16:42:24 CET] <Niklink> thanks, that's helpful [16:42:27 CET] <Niklink> linux is free, right? [20:40:43 CET] <pfloyd> I have a voicemail in .amr format, which I'd like to make into a video so I can upload to youtube. Is there a way to just pack the audio into a video container and put a static image as the video layer? [20:45:29 CET] <ariyasu> maybe just "ffmpeg -i image.jpg -i audio.amr -vcoded libx264 -acodec aac out.mkv" [20:45:43 CET] <ariyasu> it might work or it might display the jpg for 1 frame heh [20:45:46 CET] <ariyasu> worth a try though [21:36:01 CET] <kerio> ariyasu: just put it however you want, youtube will ffmpeg it out into its own format anyway [21:36:32 CET] <kerio> just -c:v copy -c:a copy [21:39:40 CET] <ed8> I'm iterating over a file content to clip out part of a single source to multiple output using bash: https://gist.github.com/edouard-lopez/e928ac0e05c283aca66509488cc1ef4f#file… [21:40:19 CET] <ed8> earlier furq advice to look at ` -segment_times` but I'm not sure it fit my need [22:05:36 CET] <Nosomy> I use ffmpeg to find discontinuities in files and generate hashes of the blocks of these, should I make a more appropriate program for this? [23:27:52 CET] <kingsley> Do you happen to know how to [23:27:56 CET] <kingsley> a.) brighten, [23:28:08 CET] <kingsley> b.) stretch the contrast and [23:28:16 CET] <kingsley> c.) make less grainy [23:28:24 CET] <kingsley> video taken in low light? [23:29:49 CET] <kingsley> I can imagine something like GIMP's white balance and HSV stretch filter combined with something like ffmpeg's minterpolate filter that compares adjacent frames to average their color and reduce graininess. [23:29:55 CET] <kingsley> What do you think? [23:39:13 CET] <kingsley> Bonus points for explaining how with kdenlive. [23:40:05 CET] <durandal_1707> kingsley: minterpolate does not average colors across frames it interpolates motion [23:40:28 CET] <kingsley> durandal_1707: I agree. [23:40:42 CET] <iive> kingsley, check `man ffmpeg-filters` for these colorlevels, curves, eq ... [23:41:25 CET] <durandal_1707> there are numerous types of grainy - so that needs more info [23:42:52 CET] <kingsley> durandal_1707: OK. I'd like to fix the graininess at [23:43:00 CET] <kingsley> http://loaner.com/2018_winter_Taekwondo_belt_ceremony.webm [23:43:35 CET] <kingsley> iive: I'm reading the man page now... [23:46:50 CET] <durandal_1707> that is from bad encoding, nothing much you can do about it, filters are for unprocessed data not transcoding miracles [23:47:50 CET] <kingsley> durandal_1707: Maybe. Or maybe we just don't know how. Yet. [23:48:24 CET] <iive> there is also "histogram" filter, that might be useful... [23:49:08 CET] <iive> hum... if you have the source , you may try "spp" filter. set quant parameter manually. [23:49:10 CET] <Nosomy> VHS cap? [23:49:43 CET] <iive> it looks like mpeg1 encode of VHS [23:49:53 CET] <Nosomy> Yeah [23:50:08 CET] <iive> and low bitrate one on top of that... [23:51:10 CET] <Nosomy> I hope has been extremely cheap. [23:53:31 CET] <kingsley> iive: I searched the ffmpeg-filters man page for the "histogram" filter. It led me to the "histeq" filter, which evidently it can correct degraded or poorly captured source video. [23:54:38 CET] <iive> well, if it works for you :D [23:54:47 CET] <durandal_1707> histeq is for histogram equalization it will not fix those artefacts... [23:55:07 CET] Action: kingsley also notes the "hqdn3d" filter, which is evidently a denoiser. [23:55:22 CET] <Nosomy> Some transcoding miracles can be happens, but only for audio. [23:55:24 CET] <durandal_1707> you can try that one [23:56:12 CET] <durandal_1707> Nosomy: transcoding (lossy) does nothing [23:56:25 CET] <durandal_1707> nothing good to video and audio [23:56:43 CET] <kingsley> iive: I agree that the "spp" filter may be worth trying. Thank you. [23:57:49 CET] <durandal_1707> none of filters are designed for those type of artefacts [23:58:19 CET] <durandal_1707> you can only try nlmeans with some special options and hope details are not lost [00:00:00 CET] --- Sun Dec 16 2018

1 0

ffmpeg-devel.log.20181214
by burek 15 Dec '18

15 Dec '18

[10:20:42 CET] <JEEB> yay, first version of my thing done [10:25:10 CET] <jya> the upgrade from ffvpx from 3.4 to 4.0.2 has introduced a playback regression which can easily be seen when watching https://www.youtube.com/watch?v=lG5LZNhfyGM [10:26:14 CET] <jya> that's the diff https://hg.mozilla.org/mozilla-central/rev/29daa22239e9 of the full change and the windows configuration: https://hg.mozilla.org/mozilla-central/diff/29daa22239e9/media/ffvpx/config… [10:26:22 CET] <jya> is there anything obvious in there? [10:26:41 CET] <jya> the drops of frames doesn't appear to be performance related [10:27:06 CET] <jya> i have a very fast machine, the CPU barely go over 10% oplaying that content at 1080p60 [10:28:41 CET] <nevcairiel> why does your config have all asm disabled, that'll make a crawling slow decoder [10:28:50 CET] <jya> may have to wait until BBB is back online [10:29:25 CET] <jya> nevcairiel: which bit? [10:29:40 CET] <nevcairiel> its not like this is a change in the config [10:29:43 CET] <nevcairiel> but still [10:30:31 CET] <nevcairiel> oh wait i looked at the wrong config, i lost my sc roll position in the diff [10:47:06 CET] <cone-844> ffmpeg 03Paul B Mahol 07master:f2179afb01a3: avformat/nutdec: fix obvious typo [10:47:59 CET] <jya> nevcairiel: the only thing of significance was the removal of the superframe parser, replaced by a bsf. [10:48:21 CET] <jya> Maybe we didn't do that change properly [10:50:22 CET] <JEEB> something like this http://up-cat.net/p/cb86ba34 [10:51:27 CET] <JEEB> although I might as well make the first one "subtitles" [10:51:31 CET] <JEEB> since that's what it is [11:17:27 CET] <JEEB> lol [11:18:04 CET] <JEEB> for some reason if I skip the first subtitle packet ffprobe's show_programs drops the resolution [11:18:16 CET] <JEEB> but it only drops a single packet so at least that seems to work \o/ [11:20:31 CET] <JEEB> http://up-cat.net/p/b33395bf [11:21:14 CET] <JEEB> that fixes start_time etc as well, since we lose the outlier [11:29:20 CET] <cone-844> ffmpeg 03Paul B Mahol 07master:6f058b5cef67: avformat/nut: add support for yuva444/422p12 pixel format [11:36:29 CET] <cone-844> ffmpeg 03Paul B Mahol 07master:e9817636a7e1: avformat/flac_picture: try to guess PNG by actual picture data [11:36:30 CET] <cone-844> ffmpeg 03Paul B Mahol 07master:5e6b93bb4dc4: avformat/id3v2: use png header to get PNG signature [12:05:55 CET] <thardin> doesn't anyone happen to have SMPTE 429-5 on hand? [12:06:05 CET] <thardin> michaelni? [12:10:56 CET] <thardin> oh boy, mxfsplit segfaults on that DCP MXF subtitle file [12:12:21 CET] <durandal_1707> and ffmpeg? [12:13:54 CET] <thardin> mxfdec can't make sense of it, but it doesn't crash at least [12:14:06 CET] <thardin> hah wikileaks has the SMPTE specs [12:14:24 CET] <cone-844> ffmpeg 03Paul B Mahol 07master:d9a91d58a17f: avformat/movenc: treat ALAC same as FLAC and write correct info [12:30:58 CET] <durandal_1707> atomnuker: does aac supports single FL/FR ... channel layouts? [12:32:43 CET] <atomnuker> as in dual mono? [12:32:55 CET] <atomnuker> or as in a non-center mono [12:33:37 CET] <atomnuker> nope, not even with PCEs [12:42:48 CET] <JEEB> umm [12:43:01 CET] <JEEB> dual mono *is* a thing and you can find the AVOption for the .jp mapping of it [12:43:08 CET] <JEEB> seems to be disabled by default [12:43:27 CET] <JEEB> thardin: yes, it also has the DCF or so specs, which was funny (and one of the newer versions of qtff.pdf) [12:48:01 CET] <atomnuker> JEEB: yeah, but doesn't have direct support for it [12:48:47 CET] <JEEB> right [12:58:53 CET] <cone-844> ffmpeg 03Paul B Mahol 07master:ddefd05507fd: doc/metadata: fix error in timebase description [13:22:26 CET] <thardin> JEEB: neat [13:27:34 CET] <JEEB> the sony dump had various random stuff :D [13:38:32 CET] <Compn> new or old dumb [13:38:33 CET] <Compn> dump [13:44:06 CET] <JEEB> at this point it's been a few years [14:03:26 CET] <kierank> av500: did you message me yesterday [14:03:41 CET] <kierank> On some MMS to email gateway [14:03:57 CET] <kierank> Number was from Germany [14:07:06 CET] <durandal_1707> is there any way to drop trailing metadata like ape tags from lavf utils.c? [14:22:27 CET] <cone-844> ffmpeg 03Fan Gang 07master:c6e1966c1a1a: avcodec/ass_split: fix a memory leak defect when realloc fails [17:40:16 CET] <cone-232> ffmpeg 03Paul B Mahol 07master:78e7b70395e9: avformat/ffmetadec: do no limit size of tags to 1024 [18:37:37 CET] <durandal_1707> michaelni: does libnut have relevant EOR code? [18:52:19 CET] <michaelni> durandal_1707, i suspect it does not [18:54:22 CET] <durandal_1707> michaelni: nut demuxer nowhere checks for FLAG_EOR, so this is never implemented [19:10:26 CET] <cone-232> ffmpeg 03Andriy Gelman 07master:5282db5929f2: avcodec/mpeg: Initialize quarter_sample parameter from previous thread. [19:10:27 CET] <cone-232> ffmpeg 03Lauri Kasanen 07master:b4c8c03b0040: swscale/output: VSX-optimize 16-bit yuv2plane1 [19:43:14 CET] <durandal_1707> michaelni: why nut_write_trailer have assert1 for option which is set by user? [19:56:53 CET] <durandal_1707> michaelni: so EOR frames have zero data_size or actual size of frame header? [20:12:21 CET] <durandal_1707> michaelni: also for long 10h nut sample auto generated ffmpeg spits bunch of pts errors after 9:25 when demuxing [20:12:53 CET] <durandal_1707> michaelni: also [20:12:58 CET] <durandal_1707> Last message repeated 957 times [20:13:21 CET] <durandal_1707> [nut @ 0x246ae40] frame size > 2max_distance and no checksum [20:29:48 CET] <michaelni> durandal_1707, the assert should only be reached if its assumtion (about the user option) is true [20:29:57 CET] <michaelni> ill document this better [20:31:54 CET] <michaelni> EOR size refers to size of packet IIRC [20:33:09 CET] <michaelni> about nut bugs, feel free to add me to the CC of the tickets if you like, so i dont forget [20:58:35 CET] <BradleyS> 5282db5929f25ca9566a47ad217794842b364afc can be cherry-picked onto release/4.1 [21:22:37 CET] <durandal_1707> michaelni: somehow pts get fcked up [21:47:15 CET] <durandal_1707> michaelni: found bug and fixed it! [21:54:48 CET] <cone-232> ffmpeg 03Paul B Mahol 07master:6bfc935232fb: avformat/nutdec: fix pts overflow [22:32:30 CET] <jya> nevcairiel: i think I see why in the logs "The deprecated avcodec_decode_* API cannot return all the frames for this decoder. Some frames will be dropped. Update your code to the new decoding API to fix this." [22:32:55 CET] <nevcairiel> ah yeah thats because of the b sf [22:33:59 CET] <jya> is there an easy way to get the old API to work, or must we rework it all? [22:37:19 CET] <jya> that sucks big time [22:49:43 CET] <nevcairiel> jya: probably cant do that with the old api. could do something quick and dirty and call the bsf before the decoder, but thats also code changes, so might as well do it properly. [22:50:20 CET] <jya> nevcairiel: the plan would be to have something small and safe enough to uplift to release [22:50:28 CET] <jya> right now that seems to drop over 10% of the frames [22:50:41 CET] <jya> with that particular 1080p60 from YT [22:50:44 CET] <nevcairiel> it depends how many superframes there are, but yeah [22:51:29 CET] <jya> it's really sad that the old API wasn't fully removed then, what's the point of keeping something that no longer work :( [22:52:23 CET] <nevcairiel> changing to the new api is really not that hard and depending on how your code was structured may not even require much work at all [22:54:29 CET] <jya> it's just a matter of sending a frame / receiving it? [22:54:43 CET] <jya> got a link to an example handy, or some doc? [22:54:58 CET] <nevcairiel> yeah its basically just split, send a packet in, and receive frames [22:55:06 CET] <nevcairiel> in this case send a superframe, and receive more then one frame [22:57:55 CET] <nevcairiel> trying to find an example or doc that seems adequate [23:00:28 CET] <nevcairiel> but the jist of it is really quite simple, i nstead of decode_video2, you just call send(pkt); and then loop over receive(frame) until you get EAGAIN or EOF [23:01:22 CET] <jya> nevcairiel: this is what we used to do with the decode bit, at least for audio [23:01:28 CET] <jya> why not keep that ? [23:02:54 CET] <nevcairiel> mostly because it was undocumented to do that with video, and the old api was terrible for that (having to manually adjust the avpacket etc), and we needed an e ntirely new one for encoding anyway, because m:n input/output was not supported there at all [23:03:08 CET] <nevcairiel> the new api also theoretically supports full async decoding, or more advanced threading [23:03:10 CET] <nevcairiel> if thats ever used [23:04:06 CET] <jya> nevcairiel: from which version of libavcodec is the new API supported? [23:05:03 CET] <nevcairiel> 3.1 apparently [23:05:20 CET] <nevcairiel> ie 2.5 years now [23:08:33 CET] <jya> what's the libavcodec version for those ? [23:09:03 CET] <nevcairiel> 58 [23:09:33 CET] <nevcairiel> wait, thats wrong [23:09:38 CET] <jya> hmm 57 apparently (just checkout release/3.1 [23:09:45 CET] <nevcairiel> yeah 57 [23:09:50 CET] <nevcairiel> but not all 57s have it [23:09:57 CET] <jya> ohhh ffs [23:10:07 CET] <nevcairiel> it was added here: [23:10:07 CET] <nevcairiel> 2016-04-21 - 7fc329e - lavc 57.37.100 [23:10:39 CET] <jya> wm4 I'm not surprised :) [23:11:34 CET] <jya> no codecs supporting that API [23:11:38 CET] <jya> the log states [23:11:43 CET] <jya> they all do now? [23:11:50 CET] <nevcairiel> thats a bit misleading [23:11:57 CET] <nevcairiel> the api works with all codecs that use the old internal api [23:12:06 CET] <nevcairiel> but at that time no codec used the new internal api [23:12:15 CET] <jya> ah ok, so just no codec natively supporting it [23:12:39 CET] <jya> i still think the old API should be made to work, or not be there at all [23:12:49 CET] <nevcairiel> its impossible to make it work [23:12:57 CET] <nevcairiel> without changing its semantics [23:13:11 CET] <nevcairiel> which would still result in you having to change your code [23:13:14 CET] <nevcairiel> so that would be pointless [23:14:02 CET] <nevcairiel> not sure when the old api was officially deprecated and when its slated for removal [23:15:04 CET] <nevcairiel> oh in the same commit even [23:15:08 CET] <jya> may as well be removed seeing that it no longer works [23:15:14 CET] <nevcairiel> so they'll probably be removed in 59 then [23:24:30 CET] <BtbN> When did the old API stop working? It just doesn't support multiple output frames per input [23:24:58 CET] <jya> nevcairiel: there's no need to use an AVParser with the new API I gather [23:25:06 CET] <nevcairiel> not for vp9 [23:25:19 CET] <jamrial> i don't think it's been scheduled for removal yet. there are no guards or anything for it [23:25:21 CET] <jya> BtbN: well, more like the parser+old_api no longet work [23:25:25 CET] <nevcairiel> with other codecs your experience may vary [23:25:41 CET] <nevcairiel> depending on what your demuxer outputs etc [23:25:50 CET] <nevcairiel> generally ffmpegs parsers job is considered a demuxers job [23:25:54 CET] <nevcairiel> vp9 was .. special [23:25:57 CET] <nevcairiel> which is why it was changed [23:26:26 CET] <jamrial> vp9 parser was doing packet splitting, which was a bad idea as this resulted in split packets making it to muxers [23:26:43 CET] <jya> well, we went with the idea that a demuxer demux a container, and a decoder decode the bytestream [23:26:47 CET] <jamrial> so the parser was changed to simply mark keyframes, and a bsf was added to handle the splitting internally within the decoder [23:26:51 CET] <nevcairiel> jamrial: well it was deprecated in 2016, so guards or not, users have been warned long enough that there is no blocker to doing it [23:26:54 CET] <jya> so the demuxer shouldn;'t know about what it's demuxing [23:27:54 CET] <nevcairiel> jya: thats all fine for competent containers like mp4 and mkv, but what do you do if there are no clear frame boundaries and someone has to figure out where frames start and end? thats practically what ffmpeg parsers do, generally as part of avformat, ie. demuxing. [23:28:18 CET] <jya> that's fine we only support mp4 and webm :D [23:28:22 CET] <jya> (and ogg) [23:28:53 CET] <jamrial> yeah, most parsers were written to assemble full frames out of raw bitstreams. the vp9 parser however was being unique and splitting frames instead :p [23:29:47 CET] <jya> right, so < 58, use parser and old API, >= 58 don't use parser and new API [23:30:07 CET] <jya> what about vp8, need the parser too ? [23:30:15 CET] <jya> (more an issue with Libav IIRC) [23:30:20 CET] <jamrial> vp8 parser just marks keyframes [23:30:24 CET] <jamrial> so probably not [23:30:35 CET] <nevcairiel> the superframe splitting stuff was only vp9 [23:30:39 CET] <jamrial> if you're using webm/mp4 only, those are correclty marked already [23:32:01 CET] <jya> so the old api only outputs the superframe? [23:32:17 CET] <nevcairiel> whichever is first in that superframe probably [23:32:34 CET] <nevcairiel> actually, thinking about it, should this actually be a problem? [23:32:42 CET] <nevcairiel> arent additional frames in superframes invisible? [23:33:02 CET] <jamrial> nevcairiel: it's only a problem because the frame mt code is not playing nice with the autoinserted split bsf, it seems [23:33:12 CET] <nevcairiel> oh right [23:33:19 CET] <nevcairiel> frame m t [23:33:34 CET] <nevcairiel> because we call it multiple times, it outputs some older frames from before the superframe [23:33:41 CET] <nevcairiel> without threading it should probably work [23:33:47 CET] <jamrial> yeah [23:34:08 CET] <nevcairiel> one of these days we should make frame mt more smart and dynamic [23:34:13 CET] <nevcairiel> probably after the old api is dead =p [23:34:38 CET] <jya> here's the video it's quite noticeable [23:34:39 CET] <jya> https://www.youtube.com/watch?v=lG5LZNhfyGM [23:34:42 CET] <jya> 1080p60 [23:35:00 CET] <nevcairiel> i still use the old api for audio in my code [23:35:04 CET] <nevcairiel> i was being lazy :/ [23:35:12 CET] <nevcairiel> re-did video some time ago [23:35:12 CET] <BtbN> the video looks smooth here [23:35:13 CET] <jya> stream 302 with youtube-dl [23:35:28 CET] <jya> BtbN: not using firefox 64 and later [23:35:41 CET] <BtbN> This is ff65.0b4 [23:35:46 CET] <nevcairiel> BtbN: it would only be a problem if you use a new ffmpeg with a player that uses the old api and threading [23:35:48 CET] <atomnuker> nevcairiel: more smart and dynamic? in what ways? [23:35:52 CET] <jya> must enable vp9 (right click on the video and select stats for nerds) [23:35:53 CET] <nevcairiel> also probably no hardware [23:36:18 CET] <jya> BtbN: which OS are you using? [23:36:29 CET] <BtbN> Windows 10 [23:36:39 CET] <jya> sure you're getting vp9? [23:36:45 CET] <BtbN> I am according to the stats [23:36:46 CET] <jya> oh, what graphic card? [23:36:56 CET] <BtbN> gtx1060 [23:37:00 CET] <nevcairiel> atomnuker: I imagine a more flexible pool of worker threads, so that we can retrieve finished frames without being required to have new data to fill those slots, for example [23:37:02 CET] <jya> there you go, HW decoder [23:37:19 CET] <jya> yeah, that will be butter smooth [23:37:20 CET] <nevcairiel> ie. make frame-mt join the spirit of decoupled send/receive [23:37:36 CET] <BtbN> It stopped being smooth the moment I opened the stats though [23:38:12 CET] <jya> did you enable webrender? [23:38:17 CET] <cone-232> ffmpeg 03Michael Niedermayer 07master:52ba824c6581: avcodec/rasc: Check input space before reading chunk [23:38:18 CET] <cone-232> ffmpeg 03Michael Niedermayer 07master:35a603050d6c: avcodec/dvdsubdec: discard accumulated buffer on error [23:38:19 CET] <cone-232> ffmpeg 03Michael Niedermayer 07master:7aaab127bebb: avcodec/clearvideo: Check remaining input bits in P macro block loop [23:38:19 CET] <BtbN> Not that I'm aware of, let me check [23:38:25 CET] <jya> it's still on the work for nvidia [23:38:43 CET] <jya> webrender will be enabled for nvidia last, we've done amd and intel in 65 already [23:38:58 CET] <BtbN> nothing on about:config for *webrender* is changed from defaults [23:39:08 CET] <jya> about:support [23:39:26 CET] <jya> in graphics -> compositing section [23:39:26 CET] <BtbN> Compositing: WebRender [23:39:32 CET] <jya> interesting [23:39:54 CET] <jya> maybe it's enabled on the beta or you're part of the lucky test sample [23:40:02 CET] <BtbN> It's dev edition [23:40:10 CET] <jya> we enabled it to a small % of the population I believe [23:40:22 CET] <BtbN> 65 is by far the best running FF in a long while [23:40:31 CET] <BtbN> that might explain why [23:40:35 CET] <jya> that's great to hear [23:40:57 CET] <BtbN> And since nvidia finally fixed their crashing driver the other day, I also stopped sending dozens of crash reports about twitch per day [23:41:32 CET] <jya> I had to disable the windows HW decoder for vp9 on nvidia 10xx for 10 bits videos [23:41:39 CET] <jya> was resetting like crazy [23:42:06 CET] <BtbN> Twitch is just normal h264 though. And it was still crashing tabs every once in a while. [23:42:28 CET] <BtbN> The whole tab started glitching out, elements flashing white [23:42:56 CET] <BtbN> Fixed since nvidia driver 417.21 [23:43:35 CET] <jya> really? never experienced this (1080ti here) [23:43:52 CET] <jya> for testing the reset with P010 format, set media.wmf.force.allow-p010-format to true [23:43:52 CET] <BtbN> Only ever happened on Twitch, and only on some streams. [23:44:00 CET] <BtbN> But for those streams it happened every 5 minutes [23:44:16 CET] <jya> and load https://jyavenard.github.io/htmltests/tests/webm-hdr.html [23:44:20 CET] <BtbN> Switching to a transcoded stream fixed it, so it must be something their encoder was doing [23:44:39 CET] <BtbN> As the Source stream comes straight from the Streamer [23:44:52 CET] <jya> to save processing/bandwidth, the original stream is whatever the broadcaster is sending, it's not touched [23:45:15 CET] <BtbN> Yeah, and only those were crashing, if at all [23:45:20 CET] <jya> i can;'t image the hassle it must be to make their transcoded work and have the same kf as the original [23:45:22 CET] <BtbN> any of the Twitch-Transcode always ran fine [23:45:59 CET] <jya> would be interested to no if that last 10 bits videos work for you with your 1060 and the media.wmf.force.allow-p010-format set [23:46:06 CET] <jya> to know [23:46:10 CET] <BtbN> https://crash-stats.mozilla.com/report/index/7b00819d-462c-4e45-b4b0-d3a540… that's one of those crashes btw. [23:46:15 CET] <BtbN> one sec [23:46:22 CET] <jya> in which case can relax the restrictions a bit [23:47:22 CET] <jya> that crash is in the GPU process, should recover smoothly normally [23:47:40 CET] <BtbN> It put the Tab in a "This tab has crashed" state [23:48:02 CET] <BtbN> The video at the very bottom is just white with that flag set [23:48:11 CET] <jya> it's crashed the content process too? [23:48:18 CET] <jya> I'll investigate [23:48:49 CET] <BtbN> It seems fixed since nvidia released the driver that specifically claims to fix video decoding crashes (in Edge though) [23:50:06 CET] <BtbN> Yeah, that 10 bit video is definitely not working. [23:50:57 CET] <jya> i typically get all green, all black and shortly after a driver reset [23:51:10 CET] <BtbN> It's just all white here, audio plays fine, no crashes or anything [23:51:45 CET] <jya> that was the card I used when I added P010 format support, I struggled for 4 days wondering what I did wrong, until I got an intel one and miracle all worked first go [23:52:14 CET] <BtbN> I might have copied the wrong crash report. That one is way too recent, the fixed driver as already installed by then. Must be of something else [23:52:19 CET] <BtbN> One sec, I'll dig for an actual one [23:52:21 CET] <jya> P010 is microsoft name for NV12 in 10 bits [23:52:32 CET] <BtbN> they are unfortunately gone from about:crashes already [23:53:05 CET] <jya> if it got fixed with the new drivers, there wouldn't much we could have done anyway [23:53:20 CET] <BtbN> Yeah, it was pretty clear it was a driver issue [23:53:29 CET] <jya> I'm about to run out of battery, will resume tomorrow. thanks [23:53:30 CET] <jya> good night [23:53:42 CET] <BtbN> night [00:00:00 CET] --- Sat Dec 15 2018

1 0

ffmpeg.log.20181214
by burek 15 Dec '18

15 Dec '18

[01:34:04 CET] <barg> where is the public log for the channel? [01:38:02 CET] <pink_mist> barg: from https://ffmpeg.org/contact.html there's a link to https://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/ [02:50:57 CET] <Keshl> OBS guys claim that OBS doesn't mangle things to adjust for the audio streams. Perplexing. [02:51:20 CET] <Keshl> Not that this means you guys need to do anything now; just an FYI. [06:44:39 CET] <late_in_the_day> Gday, I'm on linux. I'd like to automate video capture from the webcam and audio from several jack channels . I can't specify which jack connections to make in ffmpeg, so i need to use jack_connect. Although I cant write a script because ffmpeg wont background with &. Any ideas? [08:46:08 CET] <Keshl> late_in_the_day: You need to include -nostdin . This is because ffmpeg takes interactive commands, and that flag disables them. [08:46:24 CET] <Keshl> Beyond that, everything else is normal. [08:47:31 CET] <Keshl> Also consider "-loglevel fatal" to make output quieter. -nostats too, if it's still too much.[1] ([1] https://unix.stackexchange.com/questions/222847/how-to-stream-with-ffmpeg-i… ) [09:06:39 CET] <TheAMM> Good citation [10:45:50 CET] <POPEYE> How can I get constant bit rate using ffmpeg (avconv) -b:a (-ab) and -b:v (-vb) doesn't help; avprobe still says that bitrate is variable, while Kdenlive (uses avconv) makes constant bitrate with -ab and -vb in it's profile [10:46:27 CET] <ServiceRobot> hello? [10:49:41 CET] <ServiceRobot> so I'm doing a personal project using ffmpeg, but I'm having a bit of trouble decoding video. I'm not sure if this is a good place to ask for help [10:51:51 CET] <JEEB> yes, for the APIs this is the correct place [10:52:02 CET] <JEEB> together with the libav-user mailing list on ffmpeg.org [10:52:35 CET] <ServiceRobot> I like talking directly with people through chat more though. this place seems good then [10:53:24 CET] <ServiceRobot> so I'm using a library that accesses ffmpeg's C code directly [10:54:02 CET] <ServiceRobot> the problem I'm encountering however is an error directly from ffmpeg itself. I know, because the error code returned is right on their documentation [10:54:28 CET] <ServiceRobot> but, I'm not exactly sure what the cause is. I should go into further details [10:54:59 CET] <ServiceRobot> the error is: invalid data found when processing input (-1094995529) [10:55:49 CET] <ServiceRobot> that negative number translates to INDA in hexidecimal [10:56:01 CET] <ServiceRobot> #define AVERROR_INVALIDDATA FFERRTAG( 'I','N','D','A') ///< Invalid data found when processing input [10:56:19 CET] <ServiceRobot> but I have no idea what is giving me this error [10:58:24 CET] <ServiceRobot> the documentation on this error is quite sparse. I can't find causes of it, how to prevent it, etc [11:06:22 CET] <th3_v0ice> Why would a local UDP connection drop often? [11:07:11 CET] <ServiceRobot> UDP is an unreliable protocol. it just sends data out and hopes it gets there. no checks on the other end [11:07:30 CET] <ServiceRobot> in fact, that's what I'm using to send video and audio packets across the network [11:09:20 CET] <JEEB> look at the doc/examples part [11:09:28 CET] <JEEB> also make sure that whatever you're doing goes fast enough [11:09:38 CET] <JEEB> I recommend you test with AVFormatContext reading from a file first [11:09:51 CET] <ServiceRobot> me, or the other guy? [11:10:01 CET] <JEEB> you're the only API user here active atm ;) [11:10:42 CET] <ServiceRobot> well I'm using a library that uses ffmpeg. what I want to know is, if I send video and audio data across the network, what needs to be known to ffmpeg on the other end to play the video and audio back? [11:11:09 CET] <ServiceRobot> I know height, width, and pixel format are needed for video [11:11:19 CET] <ServiceRobot> but that's not enough apparently [11:11:37 CET] <JEEB> ok, so you have another abstraction level in the middle? [11:12:01 CET] <ServiceRobot> yes. it's written in another language, but it does use ffmpeg's library [11:12:23 CET] <JEEB> that really doesn't matter as long as you're the one poking at the AVFormat/AVCodec contexts [11:12:26 CET] <JEEB> etc [11:12:30 CET] <JEEB> as in, you're in control of that layer [11:12:43 CET] <JEEB> also for what data is needed, just use a container :P [11:12:52 CET] <JEEB> one that sends out headers periodically [11:12:57 CET] <ServiceRobot> that's what I'm worried about. that I'm not. or I'm not setting them up properly. I'm not exactly an expert at C [11:13:03 CET] <JEEB> see doc/examples [11:13:28 CET] <ServiceRobot> wait, you mean just send the entire container? [11:13:29 CET] <JEEB> it has various examples. some are somewhat out-of-date, but they should still work [11:13:48 CET] <JEEB> anyways, $dayjob time :P [11:13:54 CET] <ServiceRobot> that doesn't exactly help since I'm using a different library [11:14:03 CET] <ServiceRobot> well, an abstraction layer [11:14:11 CET] <JEEB> but it's *your* abstraction layer [11:14:23 CET] <ServiceRobot> eeeh, not exactly. it's someone else's [11:14:25 CET] <JEEB> if it's not then there's nothing for me to do [11:14:31 CET] <ServiceRobot> oh [11:15:00 CET] <JEEB> but you could at least yunno check the order of things for various things :P [11:15:13 CET] <JEEB> then check against what the abstraction layer and you are doing [11:15:20 CET] <JEEB> but yea, $dayjob [11:15:35 CET] <ServiceRobot> alright, see ya. I hope someone can enlighten me on this issue more [11:23:56 CET] <th3_v0ice> ServiceRobot, JEEB: FFmpeg also fails at just reading the stream and sending it to /dev/null. That should be doing the work fast enough and still the stream is constantly disconnecting. My app which is using an API is probably not that fast but is also getting disconnects. Can something else be the culprit of the problem? [11:28:23 CET] <ServiceRobot> your issue might be different to mine [11:28:41 CET] <ServiceRobot> I'm getting an error when trying to decode media packets I send across the network [11:29:09 CET] <ServiceRobot> so either I'm incorrectly instantiating the decoders, or the packets are corrupt [11:29:18 CET] <ServiceRobot> or I'm not serializating the packets correctly [11:29:23 CET] <ServiceRobot> no idea [12:07:59 CET] <Skandalist> Can anybody help with constant bitrate? [12:10:22 CET] <sur3> Hi I'm trying to use my amd gpu with openmax and compiled openmax into mesa, when doing configure ffmpeg will show me omx accel and h264_omx but when doing make no h264_omx is compiled, what is the problem? [13:43:28 CET] <d3bug> having an encoding problem with hevc_vaapi... the error encountered is "Impossible to convert between the formats supported by the filter 'graph 0 input from stream 0:0' and the filter 'auto_scaler_0'" which only seems to happen when I try to crop the picture [13:44:46 CET] <d3bug> the actual line I am using is: ffmpeg -hwaccel vaapi -hwaccel_device /dev/dri/renderD128 -hwaccel_output_format vaapi -i inputfile.mkv -vf crop=1920:800:0:140 -vcodec hevc_vaapi -qp 20 -map_chapters 0 -bf 0 -acodec libvorbis outputfile.mp4 [13:45:24 CET] <d3bug> it works without the "crop=" part, but as soon as i try to crop it... boom - error [13:48:04 CET] <d3bug> oh... Lubuntu 18.04 LTS, AMD RX480 GPU [13:48:46 CET] <d3bug> kernel: 4.15.0-42-generic (buildd@lgw01-amd64-023) (gcc version 7.3.0 (Ubuntu 7.3.0-16ubuntu3)) [13:49:11 CET] <d3bug> any ideas what i am doing wrong here? [14:19:55 CET] <FishPencil> I found a dynamic library that exports the following: https://paste.ubuntu.com/p/MG6k7yJ6Cg/ I'd like to get the version information along with the license and configuration, but I don't know if any of these functions provide that. [14:22:42 CET] <FishPencil> I believe I need avcodec_license but I don't see that... [14:29:41 CET] <furq> FishPencil: that stuff is all in libavutil [14:30:31 CET] <furq> https://www.ffmpeg.org/doxygen/trunk/group__lavu__ver.html [14:33:11 CET] <FishPencil> So am I just out of luck if this is all I have? [15:16:18 CET] <th3_v0ice> What is the default timeout for UDP in ffmpeg? [15:39:12 CET] <saml> what's cool project idea? [15:42:51 CET] <hiihiii> hello [15:42:59 CET] <saml> th3_v0ice, what kind of timeout? how long connection can be left open without no packets? [15:43:54 CET] <hiihiii> one can do ffmpeg -i input.mp4 -c copy -f matroska output [15:44:11 CET] <hiihiii> and ffmpeg will know that the output is to be in an mkv container [15:44:29 CET] <hiihiii> I'm trying to do the same for jpeg/png/gif formats [15:44:50 CET] <hiihiii> but I'm not sure what to specify for the -f option [15:45:13 CET] <hiihiii> I checked ffmpeg list of formats but still not sure [15:45:15 CET] <Mavrik> hiihiii: ffmpeg -formats lists all muxers [15:47:43 CET] <saml> hiihiii, are you trying to convert a jpeg to mkv? [15:48:23 CET] <hiihiii> no [15:48:35 CET] <hiihiii> I'm trying to pipe the output [15:48:46 CET] <hiihiii> so I need the format for images [15:48:50 CET] <hiihiii> something like [15:49:00 CET] <Mavrik> png = "apng", jpeg = "singlejpeg", gif = "gif" ? :) [15:49:06 CET] <Mavrik> You can also just leave it out [15:49:11 CET] <hiihiii> ffmpeg -i test.mp4 -c copy -f matroska pipe:1 | gzip - [15:49:12 CET] <Mavrik> And it'll be autodetected by file extension [15:49:16 CET] <hiihiii> but for imges [15:49:29 CET] <saml> image2 ? [15:49:42 CET] <saml> what are you trying to do at large? [15:49:48 CET] <saml> gzip an image? [15:50:40 CET] <hiihiii> no. that would be stupid cuz it's already compressed. I used gzip just as an example [15:50:57 CET] <hiihiii> I'm trying to pipe the output to another program [15:51:05 CET] <saml> what does the program expect? [15:51:14 CET] <saml> bitmap? [15:51:20 CET] <hiihiii> image data [15:51:32 CET] <hiihiii> it can be anything as long as it's an image [15:51:56 CET] <saml> try -f image2 - | yourprogram [15:54:38 CET] <saml> ffmpeg -i a.png -c copy -f image2 - > b.png ; a.png and b.png are identical [15:58:00 CET] <hiihiii> thanks [15:58:13 CET] <hiihiii> it seems to work also with gif and png [15:58:47 CET] <hiihiii> which save me the headache of having to change -f for each format [16:15:02 CET] <HickHoward> so [16:15:16 CET] <saml> yeah HickHoward [16:15:22 CET] <saml> did you finish it? [16:15:30 CET] <HickHoward> finish what [16:16:03 CET] <HickHoward> oh yeah i remember saying something along the lines of "i'll upload more samples soon" [16:21:06 CET] <HickHoward> although i also said i needed some help for some ps2 .int format as used in a few Ubisoft games i have here on my HDD [16:28:47 CET] <th3_v0ice> saml: Yeah, how long until ffmpeg breaks the connection? [16:32:53 CET] <HickHoward> right [16:32:54 CET] <HickHoward> https://www.sendspace.com/file/nek2ig [16:35:11 CET] <HickHoward> one of these files can work quite well with ffmpeg [16:35:50 CET] <saml> th3_v0ice, https://www.ffmpeg.org/ffmpeg-all.html#udp there is timeout option. i don't know the default [16:37:49 CET] <saml> th3_v0ice, https://github.com/FFmpeg/FFmpeg/blob/c6e1966c1a1a8987de0834462d7a76377eaa7… looking at this, maybe timeout is 0 (no timeout) if no option is set [16:54:04 CET] <HickHoward> https://www.sendspace.com/filegroup/hZDPKUYZ8jMzHd9LCR07Rw [16:54:12 CET] <HickHoward> aaaaaaaaaaaaand another one [16:54:30 CET] <HickHoward> if you vave amy questions about these samples just ask [18:26:59 CET] <ariyasu> HickHoward you should upload them to the ftp [18:27:01 CET] <hiihiii> is there a way to format the output filename so that it notes milliseconds or similar [18:27:16 CET] <ariyasu> also ffmpeg doesn't handle ps2 videos very well [18:27:18 CET] <hiihiii> -f image2 -strftime 1 "out_%H:%M:%S.jpg" only gets me up till seconds [18:27:47 CET] <hiihiii> or can I output a random file name? [18:27:54 CET] <ariyasu> you could generate a timestamp prior to running your ffmpeg command [18:27:58 CET] <ariyasu> and store it as a var [18:28:06 CET] <ariyasu> then have your output be that var [18:28:17 CET] <CoreX> .batman [18:28:19 CET] <ariyasu> i do that to get unique filenames [18:28:53 CET] <hiihiii> do you mind if I ask how. I'm a windows dummy [18:28:58 CET] <ariyasu> sure [18:29:29 CET] <mixfix41> would there be a way to pipe ffmpeg screen capture to a local system specifically like from a wayland desktop to x11 [18:29:31 CET] <hiihiii> I did try that other syntax $(date +".%m-%d-%Y-%H:%M:%N.mp4") but it doesn't except it [18:29:58 CET] <mixfix41> or forward i mean [18:30:36 CET] <ariyasu> https://paste.ofcode.org/5aFsVTHZg5ykBPHCgvGFy3 [18:30:56 CET] <ariyasu> i do this to generate a YYYYMMDDhhmmss timestamp in a var called %timestamp% [18:31:10 CET] <ariyasu> then you can just have your output be %timestamp%.mkv or what ever [18:31:22 CET] <ariyasu> i have some other code if you want a unix timestamp [18:32:20 CET] <hiihiii> but [18:32:34 CET] <hiihiii> I'm going to run a script over some images [18:32:46 CET] <hiihiii> obviously it'll run fast [18:32:58 CET] <ariyasu> https://paste.ofcode.org/zS9yZYR6jDTMHCfwhEtBuk [18:33:03 CET] <ariyasu> yeah thats fine [18:33:11 CET] <ariyasu> just save it as timestamp.bat [18:33:26 CET] <ariyasu> then do "call timestamp.bat" before each command [18:33:33 CET] <ariyasu> then it's always updated [18:34:41 CET] <hiihiii> won't two commands run in less than a second? they'd both have the same timestamp, no? [18:35:16 CET] <ariyasu> if you think thats a possibility you can add a line like [18:35:22 CET] <ariyasu> timeout 1 [18:35:32 CET] <ariyasu> and it would wait 1 second before executing the the next command [18:35:44 CET] <hiihiii> ok thank you. [18:35:46 CET] <ariyasu> or if you don't care about the time stamp [18:35:54 CET] <ariyasu> and just want unique name [18:35:59 CET] <ariyasu> you could do [18:36:15 CET] <ariyasu> set /a timestamp=%timestamp% + 1 [18:36:33 CET] <ariyasu> and it would add 1 to the number each time and would be quicker than waiting 1 second [18:38:25 CET] <hiihiii> thanks that will work [20:09:01 CET] <HickHoward> >you should upload them to the ftp [20:09:02 CET] <HickHoward> ...how? [20:17:41 CET] <pink_mist> http://ffmpeg.org/bugreports.html <-- has a "Submitting Sample Media" section [20:18:41 CET] <HickHoward> cool [20:21:49 CET] <HickHoward> does that section assume i have some kind of knowledge about...something or...? [20:31:16 CET] <pink_mist> it assumes you know what ftp is [20:31:26 CET] <pink_mist> and how to use an ftp client [20:31:37 CET] <pink_mist> if you don't, you can google tutorials [20:32:17 CET] <HickHoward> what's that supposed to mean [20:32:36 CET] <pink_mist> exactly what I said [20:33:56 CET] <Foaly> ftp is a protocol for tranfering files basically [20:34:48 CET] <HickHoward> yah i kinda know what an ftp is [20:38:10 CET] <pink_mist> HickHoward: so what do you think is missing? they mention what details to use when connecting, they mention which directory to upload into, they mention the limits on uploading and how to work around them [20:38:26 CET] <HickHoward> yes [20:38:27 CET] <HickHoward> i know [20:39:06 CET] <HickHoward> and i'll try to learn how to ftp when the need arises [20:39:15 CET] <HickHoward> which means "not now" [22:47:08 CET] <monery> test? [22:47:09 CET] <monery> ah [22:47:13 CET] <monery> greetings channel [22:47:19 CET] <monery> forgot about nickserv [22:47:39 CET] <monery> is anyone free to answer some questions about cuda and ffmpeg on debian? [22:49:00 CET] <monery> having some trouble getting ffmpeg running on a box I made from scraps to speed up some encoding stuff [22:51:43 CET] <monery> nvidia-smi reports 384.130 and the ffmpeg is from the debian repo's ffmpeg version 3.2.12-1~deb9u1 [22:52:14 CET] <monery> [h264_nvenc @ 0x55b3c30d5540] Cannot load libnvidia-encode.so.1 [22:52:56 CET] <monery> I am sure I did something stoopid or forgot to install a package.. if anyone knows... if not... I am just screwing around today... I'll keep trying google, but the closest thing I found was from 2015 :( [22:56:58 CET] <monery> hello stephenwithav [22:57:21 CET] <stephenwithav> Hello monery [22:58:01 CET] <monery> do you help with tech issues? I have one involving ffmpeg and cuda that I haven't been able to solve... looking for some insight [22:59:03 CET] <stephenwithav> Not cuda issues, unfortunately. I'm here for assistance, too. :) [22:59:12 CET] <monery> maybe I can help? [22:59:22 CET] <monery> I know my way around linux and ffmpeg [22:59:25 CET] <monery> use it daily [22:59:41 CET] <monery> I'm just a user tho, not part of the channel [22:59:46 CET] <stephenwithav> give me a sec to phrase my question, ty. [22:59:55 CET] <monery> no promises tho :) [23:00:04 CET] <stephenwithav> lol [23:02:49 CET] <stephenwithav> My question is: ffmpeg v4/libavcodec includes ff_libx264rgb_encoder, but wraps it in an #if CONFIG_LIBX264RGB_ENCODER block. How do I enable that? [23:03:42 CET] <kepstin> stephenwithav: that's enabled automatically if the configure script detects that your libx264 supports rgb [23:04:05 CET] <kepstin> it'll only be disabled if your x264 is too old, basically [23:05:15 CET] <kepstin> Note that the rgb support is exposed as a separate encoder, you use "-c:v libx264rgb" on the ffmpeg cli instead of "-c:v libx264" [23:05:16 CET] <stephenwithav> ty, kepstin. Is there a place to go to upgrade my x264 libs for Debian or Ubuntu? [23:07:30 CET] <stephenwithav> I'm trying to use libavcodec directly with RGB data, avoiding the sws_scale call if possible. [23:11:30 CET] <leonardus> Anyone know how this audio effect could have been made? https://leonardus.me/static/output.mp3 [23:11:41 CET] <leonardus> probably involves a low pass filter but not sure what else. [23:13:13 CET] <kepstin> stephenwithav: are you selecting the right encoder? Like i said, you have to use "libx264rgb" [23:13:31 CET] <kepstin> the rgb support's been in libx264 for quite a while [23:14:44 CET] <kepstin> even ubuntu 16.04 has it [23:16:05 CET] <stephenwithav> I'm doing avcodec_find_encoder_by_name("mpeg4"), creating a context, setting ctx->pix_fmt = AV_PIX_FMT_RGB24, then avcodec_open2(ctx, codec, NULL) fails with error -22. [23:17:20 CET] <stephenwithav> I'm running 18.04, so I should be good. [23:28:21 CET] <kepstin> well, that's wrong [23:28:29 CET] <kepstin> mpeg4 is a completely different codec from h264 [23:28:48 CET] <kepstin> and you need to look up a specific encoder "libx264rgb" by name to get the rgb-capable x264 [23:29:12 CET] <leonardus> isn't mpeg4 a container? [23:29:42 CET] <stephenwithav> Ah. ty for educating me. [23:29:56 CET] <kepstin> here, i assume it refers to mpeg 4 asp, i.e. more or less what xvid/divx were [23:30:43 CET] <kepstin> the 'mp4' container is something different. [23:31:01 CET] <kepstin> although both are part of the same overall "MPEG 4" standard [23:31:08 CET] <kepstin> although so is h264, fwiw... [23:31:41 CET] <kepstin> mpeg 4 is more like a collection of related standards than a single thing [23:38:45 CET] <birdboy88> test [23:49:36 CET] <birdboy88> Hi. Hoping someone could help me with this problem I have. I am recording PC games with NVIDIA GeForce ShadowPlay, which produces videos with variable frame rate. I am trying to make a ffmpeg script that extracts subclips and merges them with 0.5 second crossfade. The problem is that the video/audio is sometimes out of sync. I've been reading that it probably has to do with variable frame rate but I dont know how to fix it. My scri [23:49:40 CET] <birdboy88> https://pastebin.com/YkASj9A1 [23:50:15 CET] <birdboy88> Its in windows powershell by the way [23:53:50 CET] <kepstin> birdboy88: in order to keep a/v sync, i'd normally recommend extracting the audio/video together to a single file in a single command [23:54:16 CET] <kepstin> but yeah, getting crossfade in there would be a pain [23:55:21 CET] <birdboy88> yes, I think thats the reason we did it in separate steps. I actually got help for most of the script commands [23:55:48 CET] <birdboy88> I've been reading about -async and -vsync paramenters, but I dont know where to put them [23:56:17 CET] <birdboy88> Do you recommend me to post this on stackoverflow for example? [23:57:43 CET] <BtbN> I'd recommend just not to use that software. It's horrible. [23:58:14 CET] <monery> u can try transcoding to a constant bitrate format [23:58:23 CET] <monery> I know thats not usually ideal [23:58:36 CET] <kepstin> constant bitrate is a different thing from constant framerate [23:59:00 CET] <birdboy88> Its critical to not loose processing speed, so heavy re-encoding must be avoided [23:59:00 CET] <monery> I thought it was audio that was causing the problem, my appolgise [23:59:13 CET] <kepstin> but if you're posting to youtube or something, it might make sense to convert to constant framerate anyways, so you could try doing that as a first step before editing :/ [23:59:29 CET] <kepstin> how far out is the a/v sync getting? [23:59:44 CET] <BtbN> if you do cross-fade, you'll need to re-encode anyway [00:00:00 CET] --- Sat Dec 15 2018

1 0

ffmpeg-devel.log.20181213
by burek 14 Dec '18

14 Dec '18

[06:11:37 CET] <Compn> how does duckduckgo make its money ? [06:12:03 CET] <Compn> and alternatively, why would anyone pollute google results with other , lesser results [06:16:41 CET] <talih0> Hello, how long does it usually take for a patch to appear on mailing list? I sent mine in ~20 minutes, but it still hasn't been forwarded [06:19:35 CET] <Compn> probably you are stuck in the queue talih0 [06:19:37 CET] <Compn> let me check [06:20:55 CET] <Compn> i dont see it in the queue talih0 [06:21:02 CET] <Compn> where did you send your mail to ? [06:21:52 CET] <talih0> I sent it to ffmpeg-devel(a)ffmpeg.org [06:22:54 CET] <talih0> from email: andriy.gelman(a)gmail.com [06:26:13 CET] <Compn> http://ffmpeg.org/pipermail/ffmpeg-devel/2018-December/237567.html [06:26:15 CET] <Compn> that one ? [06:27:01 CET] <talih0> yes, that's the one [06:27:09 CET] <Compn> it reached the list then :) [06:28:10 CET] <talih0> thanks, Compn :) [06:28:19 CET] <Compn> no problemo [10:23:46 CET] <cone-886> ffmpeg 03Paul B Mahol 07master:f5aad350d369: avformat/wavdec: add support for 'id3 ' chunk [10:23:46 CET] <cone-886> ffmpeg 03Paul B Mahol 07master:102c11745e2d: avformat/utils: add support for reading ID3 tags at start of wav [12:01:43 CET] <durandal_1707> what was function in avformat to read lines from file with no limit on length of line? [14:24:01 CET] <JEEB> so is it valid to make MPEG-TS demuxer ignore non-trustworthy packets until PCR is received? [14:24:19 CET] <JEEB> because for such streams the demuxer seems to adjust the timestamps IFF PCR is available [14:24:28 CET] <JEEB> problem is if you receive such a packet before PCR is received [14:25:30 CET] <JEEB> as in, would anyone here be against making that the default behavior, and then AVOption if people really want to keep such packets that were received before PCR was received [14:25:56 CET] <JEEB> in theory before PCR is received you don't know if any timestamp is valid, so I think VLC f.ex. just keeps reading until PCR is received [14:53:42 CET] <durandal_1707> everybody is against it [14:53:50 CET] <JEEB> yes, thank you [14:55:18 CET] <durandal_1707> write patch, wait for michael to reply with no, start everything again. [14:58:38 CET] <JEEB> well yea, I'm expecting someone to cry out "BUT WHAT ABOUT MY PRECIOUS PACKETS?!" [14:58:59 CET] <JEEB> which is why I would have it have an AVOption if you know your input PTS and DTS are trustable [14:59:30 CET] <JEEB> another option would be to buffer those untrustworthy streams' packets until PCR [14:59:33 CET] <JEEB> buut [14:59:34 CET] <JEEB> not sure I want to go there :P [15:00:02 CET] <JEEB> simpler to discard them if no PCR [15:19:13 CET] <michaelni> durandal_1707, iam not stoping you from commiting your original h264 patch. I just think theres an issue and i pointed that out. I understand quite well that my comment is annoying as this is complex code and finding a different solution that works is not easy. Its certainly not my intend to cause you extra work. So i wont complain if you push it as it was. [15:22:16 CET] <durandal_1707> michaelni: it is not perfect solution, it gives worse output with sample that have single IDR? frame [19:32:32 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:faca28c264be: avcodec: rewrite gif muxing and encoding [19:32:32 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:3f8cdd4ee3fa: avcodec/gif: add support for alpha [19:32:32 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:80f0f8660c8f: avformat/gifdec: export sample aspect ratio [19:32:32 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:bb0984cc29e9: avcodec/gif: enable encoding single gif image per frame [19:32:32 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:280b43246434: avformat/gif: extend description [19:32:32 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:f2664a306fb5: avcodec/codec_desc: extend gif description [19:32:33 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:0aa5a7b2e98e: avformat/gifdec: export duration, nb_frames and comment [19:32:34 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:ec8502f9cb96: avcodec/gif: use avctx->frame_number [19:32:35 CET] <cone-203> ffmpeg 03Paul B Mahol 07master:0ed678a97abd: bump micro after recent gif changes [20:43:34 CET] <durandal_1707> shouldn't there be -sidedata:[v,a] option to drop global stream sidedata? [20:59:55 CET] <durandal_1707> this community is full of clueless people [21:00:18 CET] <JEEB> thank you [21:00:47 CET] <durandal_1707> JEEB: not you, you have not designed this mess like side data [21:01:30 CET] <JEEB> I'm just happy that closed caption side data is going through filter_complex [21:02:08 CET] <JEEB> which means that if you do lavfi filter chains it won't butcher AVFrame side data [21:02:12 CET] <durandal_1707> replaygain side data is extracted from global metadata and stored in stream side data, resulting that in some cases it remain in output even when disabled [21:02:22 CET] <JEEB> ah [21:02:53 CET] <JEEB> yea I bet a lot of use cases haven't fully been implemented [21:03:01 CET] <JEEB> be it stream or global or frame side data [21:11:40 CET] <atomnuker> yeah, I came across that when I tried to remove replaygain side data when remuxing, it was annoying but I had to leave it as-is [21:55:31 CET] <atomnuker> do wave files support cover art? [21:55:55 CET] <atomnuker> I have a file here which mediainfo claims to have embedded coverart [21:56:29 CET] <JEEB> time to check the RIFF parts? [21:56:33 CET] <JEEB> since IIRC WAV was RIFF? [21:57:38 CET] <atomnuker> I'm seeing a JFIF sequence in the file followed by some entropy coded looking data [21:57:48 CET] <durandal_1707> atomnuker: complete uncut ffmpeg console output missing! [22:08:39 CET] <durandal_1707> atomnuker: does wav file have id3/ID3 chunk ? or it starts with ID3 ? [22:11:48 CET] <atomnuker> yep, and your patch from today exports it fine so nevermind [22:28:05 CET] <atomnuker> "wav: Discarding ID3 tags because more suitable tags were found." <- eh, aren't id3 tags more reliable than wav's native? [22:29:35 CET] <durandal_1707> that happens with ID3 at beginning of wavs [22:29:49 CET] <durandal_1707> such files are hack anyway [22:30:06 CET] <durandal_1707> like you could insert ID3 to any file [22:46:53 CET] <atomnuker> yeah, the files seem sketchy as they have incorrectly signalled tag encoding causing everything to appear as ?????? [22:50:07 CET] <durandal_1707> atomnuker: from where you got it? [22:53:08 CET] <atomnuker> baidu -_- [22:56:13 CET] <durandal_1707> michaelni: nut does not take into consideration duration of last frame to return real duration [23:05:24 CET] <durandal_1707> michaelni: so duration is always smaller by small amount [23:05:40 CET] <durandal_1707> michaelni: looks like design failure [23:47:06 CET] <cone-112> ffmpeg 03Paul B Mahol 07master:8c9fff618382: avcodec/xfacedec: fix order of operations [23:53:16 CET] <cone-112> ffmpeg 03Paul B Mahol 07master:e5a0013c4a62: avformat/mxfdec: fix error check in macro [00:00:00 CET] --- Fri Dec 14 2018

1 0

ffmpeg.log.20181213
by burek 14 Dec '18

14 Dec '18

[01:57:21 CET] <cirdan> evening all [01:58:01 CET] <cirdan> if I have a multi-part show each part having subs, how can I combine the parts and subs so it all works? [03:16:57 CET] <quint> I am playing with the HLS format currently, and I am able to successfully stream a video file via my web server using this command: [03:17:48 CET] <quint> ffmpeg -re -i file.mp4 -map 0:0 -map 0:1 -c:v copy -c:a aac -f hls -hls_flags delete_segments -hls_segment_filename 'chunk%04d.ts' stream.m3u8 [03:18:31 CET] <quint> If I close the stream playing from ffplay, shortly after waiting I can no longer view the video stream (ffmpeg is still happily encoding) [03:20:06 CET] <quint> these are the errors I start receiving with ffplay: https://paste.linux.community/view/raw/b0f1667b [03:20:48 CET] <quint> after exiting, then waiting a moment to re play the path of the m3u8 over http [03:24:07 CET] <quint> Has anyone any idea what I've done wrong? I'm trying to play a video file live as an "event" [03:25:57 CET] <quint> The video file is H.264 [07:42:40 CET] <leonardus> can the input file be overwritten by the output file with ffmpeg? [07:46:09 CET] <c_14> ffmpeg doesn't ensure it doesn't do that, but you if you do you'll corrupt your file [08:40:37 CET] <realies> i suspect ffmpeg died mid transcoding and left a semi-corrupted file that's missing some tags, the node app that spawned it failed the call with "spawn ENOMEM" while the system never had less than 800mb free memory, is it possible that ffmpeg really wanted more memory than the available at the sudden? [09:31:34 CET] <Keshl> I just wanna run this by someone else because it's 3:30 AM and I might be missing something obvious.. I have "Stream #0:0: Video: h264 (High), yuv420p(tv, bt709/unknown/unknown, progressive), 1280x720 [SAR 1:1 DAR 16:9], 1k fps, 60 tbr, 1k tbn, 120 tbc (default)". -- Is that stream *REALLY* encoded at 1,000 frames per second, or does "fps" mean something else in this context? [09:35:00 CET] <furq> Keshl: that's 1,000 fps [09:35:23 CET] <furq> you can doublecheck it with something like mediainfo [09:36:50 CET] <Keshl> Okay, then something's wonky somewhere. I used ffmpeg to encode this, and told it -r 60. [09:37:25 CET] <furq> yeah those timebases don't look right for 1k fps [09:37:33 CET] <furq> i guess it's possible that ffmpeg is misdetecting it [09:37:58 CET] <Keshl> Well, to be honest, I /assume/ I told it -r 60. It comes from OBS, which uses ffmpeg to do almost all of the heavy lifting. There's a guano-ton of oddities with the files that've been kinda evolving over time. [09:38:52 CET] <Keshl> For example, if I stream to Twitch and record to disk at the same time, after telling OBS to "use stream encoder" for the disk recording, /I don't get identical output/. [09:39:17 CET] <Keshl> I don't mean "I redownloaded the video and it's different" -- Of course Twitch is going to encode that. It's more dramatic. [09:39:57 CET] <Keshl> So, on the live stream, everything's totally fine. No issues anywhere. Disk, once per second, there's like a solid ten dropped frames. [09:40:20 CET] <furq> that sounds like fun [09:40:39 CET] <Keshl> It's not "Oh, my player can't keep up" -- If I do ffmpeg -i video.mkv frame_%04d.png, seriously, frame_0001.png through Frmae_0010.png are dupicated. [09:41:00 CET] <Keshl> (And doing the same to the downloaded video, it's totally fine.) [09:41:22 CET] <furq> ironically the ffmpeg tee muxer will do exactly that and it seems to work fine [09:41:29 CET] <furq> so presumably they've duplicated that functionality and broken it [09:41:33 CET] <Keshl> HEH. [09:41:54 CET] <furq> i assume this is a cfr video [09:42:15 CET] <Keshl> ...Come to think of it, this broke around the time I switched to CBR. [09:42:21 CET] <furq> cfr, not cbr [09:42:25 CET] <furq> constant framerate [09:42:38 CET] <Keshl> I was using CRF before, but that kept doing different, but equally weird, things. [09:42:45 CET] <Keshl> Oh. "Sort of". [09:42:56 CET] <furq> i mean you've not told obs to record vfr [09:43:08 CET] <furq> if you have then i guess that might explain the framerate misdetection [09:43:13 CET] <Keshl> The video output to the stream and to disk is expected to be a constant frame rate of 60 FPS, yes. The input may not always by 60 FPS, but if it goes under, I expect duplicate frames, and if it goes over, dropped frames. [09:43:19 CET] <furq> yeah, that's cfr [09:43:30 CET] <furq> vfr would just delay the next actual frame instead of duplicating one [09:43:43 CET] <Keshl> ...You can /do/ that? o_o? [09:43:54 CET] <furq> for a given definition of do [09:44:02 CET] <furq> a lot of things really don't like it [09:44:11 CET] <Keshl> Okay, that makes sense. [09:44:36 CET] <furq> you can look at the actual frame timestamps and whatnot with ffprobe -show_frames [09:44:56 CET] <furq> probably piped into head or less or something if you don't want a million lines spammed into your terminal [09:44:56 CET] Action: Keshl does that on a 12 second video, not his 15 hour video.. [09:45:04 CET] <furq> or that [09:45:45 CET] <Keshl> ...That's a lot of info. [09:46:33 CET] <Keshl> Anyway. Sorry for the tangents. Lemme just ask my actual pressing question.. [09:48:11 CET] <Keshl> So, I've obviously recoreded an utter poopton of video. We're at like 400 hours now. 400 hours of 720p60 video is *big* when you record at 4,500 bits per second. I'm running out of storage space, clearly, and would like to not have this problem anymore. I've been re-encoding my videos with x265 and a relitavely low quality, and it's helping massively, but I still want things to, 'yaknow, /play/, in the future. It all plays fine /right [09:48:12 CET] <Keshl> now/, but is appearing to have 1,000 FPS going to break anything down the line with future video players? [09:48:44 CET] <furq> if it plays now it'll probably be fine [09:48:48 CET] <Keshl> I still have the originals right now, but I can't for much longer, so I kinda need to sort out weirdo encoding issues before I start deleting the originals. [09:48:50 CET] <Keshl> Okay, thank'ye. [09:49:34 CET] <furq> i'll leave it to other people go off about the importance of archiving the originals [09:49:38 CET] <furq> to [09:50:01 CET] <Keshl> I'm aware, and normally I do, but this is a special case where it just doesn't matter. I actually considered downscaling to 360p on top of all this. [09:51:44 CET] <Keshl> I'm not planning to actually upload these anywhere, or share them with anyone outside of a select few people, and I'm only really holding on to them because it just feels a little weird not to. I'm doing a full run at default resource rates in a notoriously difficult survival game, and I mostly just want some vague evidence that I actually did it, if in 20 years from now I'm like "was that all a dream?". [09:54:35 CET] <Keshl> Said game's also notoriously poorly optimized, and I'm on a Sabertooth X79 motherboard, which only supports up to PCIe2. I've had to get a second GPU to handle any sort of recording since my first GPU's busy trying to run the game, and the physics calculations are actually thrown off rather dramtically if I encode on my CPU. PCIe2 is also causing issues; I actually legitinimately do not have the bandwidth to encode 60 FPS above 720p, and [09:54:37 CET] <Keshl> I have to keep the bitrate under a certain threshold, else I get tons of frames that just go utterly missing. None of this is ffmpeg's fault; it's just all circumstances that mean I already have somewhat crummy video to start with, and I actually don't notice much of a decrease in quality when I re-encode since everything's already kinda weird-looking in the original. [09:59:30 CET] <Keshl> I actually would be recording at 360p right off the bat if I wasn't streaming online, but people don't seem to appreicate 360p video that much for some reason, and when I tried to downscale the video in real-time in the recording without doing that over the stream, apparently OBS actually made two instances of ffmpeg and genuinely encoded each frame twice from scratch. That was obviously far, far too much of a load for my system, so I [09:59:32 CET] <Keshl> had to just copy the network's encoder (even though it's clearly not actually a copy, somehow) and then encode it again later. My choices were downscale to 360p or try x265, and they actually give pretty comprable results in terms of filesize, but x265 looks a heck of a lot better, so I've gone with that and no downscaling. I figure if I need to save even more space later, I can downscale that and I ain't going to care that I encoded [09:59:33 CET] <Keshl> three times, considereing the amount of downscaling it'd be. [10:00:57 CET] <Keshl> Anyway, thank'ye for the help. I wish I could return the favor, but I only have about half an idea of what I'm doing with ffmpeg. x.x [10:52:56 CET] <Keshl> Not urgent. Just curious: Why does ffprobe mis-detect things like this, but everything's still okay? I figure there has to be a good and probably interesting reason; obviously with such a good project and so many contributors, it seems silly that there'd be a problem as critical as "ffmpeg can't figure out what a frameerate is and just makes absurdely big numbers to compensate", so I assume that that isn't an actual problem and there's [10:52:58 CET] <Keshl> something deeper to it. I'd like a general idea of what that might be. Now and then, I try to learn more about how modern videos work, but it tends to go right over my head. This seems like something I might be able to grasp. [10:55:26 CET] <Mavrik> The reason is probably the fact that static framerate isn't something that's a thing in modern video formats :) [11:16:05 CET] <Keshl> Thank'ye. That is interesting. o_o [11:42:04 CET] <myhau> hi, I have a question: should I expect dash demuxer in standard ffmpeg distribution for macos ? [11:43:30 CET] <myhau> I'm asking, because web documentation lists (https://www.ffmpeg.org/ffmpeg-formats.html#dash-1) it as a provided demuxer, but on my machine ffmpeg -demuxers does not contain it [11:43:49 CET] <myhau> (ffmpeg version 4.1) [11:55:24 CET] <chinasaur> myhau: I don't think so [11:55:37 CET] <chinasaur> https://evermeet.cx/ffmpeg/ lists the compile flags [11:55:58 CET] <chinasaur> enable-libxml2 is not present which is required for dash support [12:25:49 CET] <azaki> is ffmpeg going to use dav1d or are the devs working on their own decoder? [12:26:28 CET] <azaki> for av1 decoding [12:27:55 CET] <furq> azaki: a bunch of the dav1d contributors are also ffmpeg contributors [12:28:01 CET] <furq> so it'll probably just get folded in at some point [12:28:26 CET] <azaki> furq, ah, nice. [14:15:16 CET] <sur3> I'm trying to use vaapi for encoding but ffmpeg gives me "libva info: va_getDriverName() returns -1" even though when invoking vainfo directly it finds my amd driver [14:20:47 CET] <sur3> ok seems i needed: LIBVA_DRIVER_NAME=radeonsi [14:20:57 CET] <sur3> but now there is still a problem.. [14:21:07 CET] <sur3> [h264_vaapi @ 0x55d3d114b720] B frames are not supported (0x1). [14:21:17 CET] <sur3> how to solve that? [14:28:00 CET] <sur3> ok found option -profile:v 578 to solve the b frame problem, but what does it do? [15:31:06 CET] <relaxed> sur3: I believe it sets constrained_baseline, look at ffmpeg -h encoder=h264_vaapi [15:31:30 CET] <relaxed> -profile:v constrained_baseline would work too [15:34:03 CET] <Mavrik> That will look awful tho O.o [16:33:46 CET] <toli> guys, I need help on converting video files without audio in them [16:34:33 CET] <toli> so, the goal is to convert 720p to 1080p, but I want to conserve the bitrate, which changes from 720p 10000kbps to 498kbps [16:34:49 CET] <toli> the video sources are in h264 [16:35:21 CET] <toli> even if I use -minrate 12000k -maxrate 12000k -bufsize 12000k it keeps the output file in 480 bps [16:48:49 CET] <Blacker47> <toli> so, the goal is to convert 720p to 1080p <-- the pulled apart pixels are crying. [16:50:49 CET] <toli> Blacker47, the resolution will stay the same at the end I know, just need this as we are using a software which is limiting the media files, it checks the details of the file before displaying it [16:51:21 CET] <toli> it will not show you the files in 720p, but in 1080p :) [16:52:17 CET] <toli> now we are making all the medias in 1080p, but we still use files since 10 years ago, where we only made 720p, and we don't have the sources to encode them in 1080 [17:04:30 CET] <sur3> I'm just trying to compile ffmpeg with amf now, but configure says "ERROR: amf requested but not found" what exactly is it missing? [17:05:11 CET] <sur3> when trying to use omx it at least says "ERROR: OMX_Core.h not found" [17:08:32 CET] <BtbN> check config.log [17:10:35 CET] <kepstin> toli: please share your complete command line. [17:11:39 CET] <kepstin> toli: main thing to note is that unless you specify a target bitrate (-b:v), x264 uses crf mode by default, and I'm not sure that -minrate really does anything in that case. [17:29:55 CET] <toli> kepstin, This is my command ..\bin\ffmpeg.exe -i 201811281410349938-EPC_PROMO_12_decembre2018_photoderm_dperarls.mp4 -vcodec libx264 -b 750000 -vf scale=1920:1080 out\test.mp4 [17:30:32 CET] <toli> this is the only way that the exported file is 4000-5000 pbs [17:30:36 CET] <toli> bps [17:33:45 CET] <kepstin> toli: you're setting -b 750000 (750kbps) and no other options there. [17:34:43 CET] <toli> and it says max bitrate [17:34:56 CET] <kepstin> you didn't set a max bitrate in that command line [17:35:38 CET] <kepstin> the -b option just sets a target average bitrate, and it's possible x264 might undershoot that on non-complex video (or overshoot it on short video) [17:35:40 CET] <toli> how to set the cpb? [17:36:00 CET] <kepstin> sorry, what's cpb? [17:36:23 CET] <toli> cpb: bitrate max/min/avg: 0/0/76800000 buffer size: 0 vbv_delay: -1 [17:37:15 CET] <kepstin> hmm. adding the -minrate, -maxrate, and -bufsize options should tell x264 to fill in those values, I think... [17:37:54 CET] <kepstin> (you probably need to choose values that make sense, i'd expect x264 will behave strangely if -b is lower than -minrate for example) [17:38:27 CET] <kepstin> also, I don't think x264 does padding? (or if it does, it needs additional options), so you might still see undershoots. [17:47:26 CET] <q3cpma> Hello, does anyone know how to remove the side data from mp3 files? I'm using the volume=replaygain audio filter so I don't want to keep replaygain data. [17:50:37 CET] <kepstin> q3cpma: the volume filter should be dropping the side data as it applies the gain [17:51:12 CET] <durandal_1707> -af asidedata=mode=delete:... [17:51:51 CET] <q3cpma> kepstin: Well, it doesn't [17:52:26 CET] <kepstin> https://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavfilter/af_volume.c;h=b… it's right there in the code :/ [17:52:32 CET] <q3cpma> durandal_1707: Didn't work. Tried "-af volume=replaygain=drop,asidedata=mode=delete" [17:52:51 CET] <kepstin> note that that removes the side data, but it's possible that you have some replaygain *metadata* tags which ffmpeg isn't removing [17:53:55 CET] <q3cpma> It looks like the side data I'm talking about is carried from flac to mp3 [17:54:26 CET] <kepstin> q3cpma: it's probably metadata rather than side data. can you pastebin the complete ffmpeg output? [17:57:55 CET] <q3cpma> Here: https://dpaste.de/URaE [17:59:02 CET] <kepstin> can you pastebin the complete ffmpeg with the *default* log level? [17:59:08 CET] <kepstin> that removed all the good stuff :/ [17:59:15 CET] <q3cpma> Oh yeah, whoops [18:04:23 CET] <q3cpma> Here, sorry for the slowness https://dpaste.de/sEjW [18:06:15 CET] <kepstin> huh, that's really weird [18:06:25 CET] <kepstin> also weird, your ffmpeg isn't showing the version banner. what version is it? [18:06:38 CET] <q3cpma> Yeah, I aliased it for this [18:06:52 CET] <q3cpma> It's 4.0.3 on gentoo [18:08:04 CET] <q3cpma> So what's strange is that's it's removed when writing to flac, but not mp3 [18:08:09 CET] <kepstin> hmm, i'm pretty there's no changes to the volume filter since 4.0, really weird that the side data is still showing up on the output [18:08:56 CET] <q3cpma> Forgot to include `ffprobe test2.flac` but the side data is indeed no here for this one [18:09:24 CET] <kepstin> at this point, all I can say is you might want to file a bug report, and if you do so please actually include the complete output (no aliases that hide output or changes to loglevel), and consider trying a build from git (it's pretty quick/easy to do that). [18:09:51 CET] <q3cpma> I'll do. [18:10:48 CET] <kepstin> does this happen on all your mp3 files or just some of them? [18:10:54 CET] <kepstin> er, flac files* [18:11:24 CET] <kepstin> that said, the metadata looks fairly normal (the reference level stuff is a bit unusual, but those are ignored/removed) [18:11:45 CET] <q3cpma> All the files, here [18:12:19 CET] <q3cpma> Looks like it's only for libmp3lame, libvorbis/libopus have no problem [18:12:37 CET] <kepstin> lets see if I can reproduce this. [18:15:45 CET] <kepstin> weird, it's happening for me with all output formats i've tested with 4.1 [18:17:48 CET] <q3cpma> I thinks I'll use flac -> asidedata -> flac -> mp3 for now [18:19:01 CET] <kepstin> the weird thing is that in my 4.1 build, it doesn't remove the replaygain side data with flac either [18:19:20 CET] <q3cpma> Wait no, asidedata doesn't work too [18:19:32 CET] <q3cpma> I'll emerge 4.1 and try again the whole stuff [18:20:01 CET] <kepstin> the workaround is to use an audio file tag editing software on the mp3 result to remove the replaygain tags :/ [18:20:38 CET] <q3cpma> What kind of software? Since this isn't at the container level, right? [18:21:03 CET] <kepstin> the replaygain info is stored in id3v2 tags (TXXX:REPLAYGAIN_ALBUM_GAIN, etc.) [18:21:12 CET] <kepstin> container level, yeah [18:21:13 CET] <q3cpma> Well, but not the side data, right? [18:21:32 CET] <kepstin> the side data is an ffmpeg internal concept, has nothing to do with how it's stored in the file [18:21:57 CET] <kepstin> in most formats, ffmpeg reads from the metadata tags to fill in sidedata, then saves sidedata by writing to metadata tags. [18:21:58 CET] <q3cpma> Oh, I see [18:23:06 CET] <q3cpma> I updated to 4.1, and encoding to flac still strips the replaygain data [18:26:22 CET] <q3cpma> Strange, but the software I used didn't find any tag [18:27:39 CET] <kepstin> hmm. I might not have time to look at it until later, but if you could encode a sample mp3 (use -t 1 so it's just 1 second long or something) i can check what's stored in there. [18:28:21 CET] <kepstin> although since I can reproduce the issue here that's probably unneccessary actually [18:28:52 CET] <q3cpma> Yeah, but strange that id3tool and id3ted can't find anything [18:29:01 CET] <q3cpma> As I said, it's probably not store in the container [18:29:50 CET] <kepstin> there's no where else it could be stored other than extra container headers to the mp3 data [18:30:05 CET] <q3cpma> Well, it could be store in the actual frames, like for flac [18:30:26 CET] <kepstin> that's not how mp3 works :) [18:30:39 CET] <q3cpma> Well, I don't see any other explanantion [18:30:47 CET] <kepstin> hmm. looks like ffmpeg is writing a xing header, i wonder if it's filling that in. [18:30:48 CET] <q3cpma> ffplay/mpv find the information, but all the id3 tools don't [18:32:29 CET] <q3cpma> kepstin: Here's the test file (1s): https://pomf.pyonpyon.moe/hyozvh.mp3 [18:32:45 CET] <kepstin> yeah, I've already made one and i'm seeing the same behaviour [18:33:44 CET] <q3cpma> I see. [18:34:12 CET] Action: kepstin is busy at work right now, but will definitely look into this more in the evening. [18:35:00 CET] <q3cpma> kepstin: Should I still make a bug report, [18:35:59 CET] <kepstin> yeah, go ahead. [18:36:03 CET] <q3cpma> Okay [18:36:09 CET] Action: kepstin should test some input formats other than flac, too [18:37:06 CET] <kepstin> hmm, it seems specific to flac input files [18:37:17 CET] <kepstin> i don't see the same thing with e.g. opus input [18:37:50 CET] <kepstin> oh, nvm, it's not reading the replaygain from this opus file at all [18:37:54 CET] Action: kepstin tries more format [18:38:06 CET] <q3cpma> Yeah, I tried wv, same story [18:38:28 CET] <q3cpma> The main difference is that ffprobe prints "Side data: track gain..." only for flac, here [19:25:18 CET] <SpeakerToMeat> Hello All. [19:25:27 CET] <SpeakerToMeat> Question, do DPX files have a metadata place for fps setting? [19:28:12 CET] <durandal_1707> yes [19:28:21 CET] <SpeakerToMeat> I want to export a 23.976 prores to DPX which I'll reinterpret at 24 (constant frame reencoding), but the text says it'll output at 23.98, I know single dpx can't be 23.98 as they're discrete, but I wonder if some metadata is being set in them that might confuse some NLEs/encoders. If so, is there any way I can tell ffmpeg to out the dpx at 24 without doing a time constant reencode? [19:28:46 CET] <durandal_1707> that is not implemented [19:28:51 CET] <SpeakerToMeat> hmm [19:29:22 CET] <SpeakerToMeat> Maybe I can export them as it is, and run a batch change with exif-tool or a library (and python or similar) and tell them all they're 24 [19:29:25 CET] <SpeakerToMeat> thanks [19:32:06 CET] <SpeakerToMeat> Btw if I have video and multiple audios, I can use map to output each on a file, right? like: "-map 0:0 -c:v dpx files%06d.dpx -map 0:1 -c:a copy file.wav" no? [19:39:59 CET] <SpeakerToMeat> Btw does ffmpeg for conversion use all possible cores (when possible)? [20:43:49 CET] <kepstin> SpeakerToMeat: individual encoders/decoders may be multithreaded, but the ffmpeg cli tool otherwise only uses a single cpu [20:47:25 CET] <SpeakerToMeat> I wonder if prores->dpx will multithread, it should be able to. [20:47:56 CET] <SpeakerToMeat> Yes, it's multithreading. thanks [22:14:45 CET] <SpeakerToMeat> for going from 23.976 to 24 without frameblending I need to do setpts=0.999 right? [22:15:55 CET] <SpeakerToMeat> ffmpeg -i my_23.976.mov -vf "setpts=0.999*PTS" -r 24 my_24.mov no? [22:16:45 CET] <Keshl> Quick comment on my issue from last night: Okay, things got weirder. If I set the bitrate to 8,000 instead of 4,500, magically it reads everything as 60 (fps, tbn, and tbc all report 60), the duplicated frames in the recording are gone from their usual spots, and dumping the frames to PNG files shows them all individually unique. MPC reports it as playing at 60 just fine. Everything's happy. ... What the *heck*? o_o' [22:16:56 CET] <Keshl> /Literally/ the only thing I changed was bitrate. [22:17:20 CET] <JEEB> Keshl: some encoders drop frames if they can't compress [22:17:23 CET] <JEEB> theora for example did that [22:17:33 CET] <JEEB> so depending on your encoder used it might be doing that [22:17:43 CET] <Keshl> My encoder was nvenc. Does that one do that? [22:18:00 CET] <JEEB> no idea, I think that's mostly controlled by the driver [22:18:02 CET] <Keshl> Oh right, wait, you weren't here last night for the weirdness. [22:18:05 CET] <JEEB> ASIC encoders yay [22:20:17 CET] <Keshl> Quick summary: I'm recording my gameplay footage because I'm trying to beat a long and difficult game, and down the road, if I start wondering if it was all a dream, I'll be able to show myself that I actually did it. Quality doesn't matter too much here since it's just for my own use and I just want a gentle little "You did it!" reminder, not a frame-by-frame analysis of my tactics. I was going to record at 360p and encode with h264 on [22:20:19 CET] <Keshl> CPU, using a CRF of like 28 or maybe 33, but due to various hardware issues and people wanting to watch me play in real-time, I had to settle for nvenc at 720 instead. [22:20:21 CET] <Keshl> Now, here's the weirdness. [22:22:33 CET] <Keshl> I'm using OBS to do all the streaming and recording, which is basically just a GUI frontend for ffmpeg with some extras thrown on top, primarily aimed at streamers who just want to click "Go" and forget about it. I have it set to use the same encoder to stream to the network and to record to disk, so logically it's only encoding once but has two outputs. /Somehow, these outputs are not equal/. The network stream magically doesn't have [22:22:34 CET] <Keshl> dropped frames anywhere, but on-disk recording does.. And weirder still, the on-disk recording somehow claims it's been recorded at 1,000 FPS (60 tbn, 120 tbc). The recording plays totally fine, thank goodness, but I came here asking is the fact that ffmpeg thinks it's 1,000 fps might cause issues with playback down the line. [22:23:03 CET] <Keshl> I was recording and streaming at 4,500 bps back then. Today I tried 8,000 just for kicks, and that change alone seems to have magically solved the fps detecition(?) issues. [22:23:26 CET] <JEEB> the "1000fps" stuff is FLV output, right? [22:23:47 CET] <JEEB> (the time base for FLV is 1000Hz) [22:23:47 CET] <Keshl> I'm recording to an MKV container, but if you're basically asking "Is that directly from OBS", then yeah. [22:24:16 CET] <JEEB> ok that's interesting then because I would have thought matroska has a somewhat larger time base than that [22:24:25 CET] <Keshl> MKV because I have four audio streams (One for just game audio, one for just my mic, one for just Skype, one for everything combined for compatbility purposes.) [22:24:46 CET] <Keshl> ...Oh my gosh, that makes sense suddenly. [22:26:28 CET] <Keshl> Twitch it's self can only handle one audio stream. I'm recording to disk with four, but sending to the network with one. I remember reading somewhere that you can't just magically add or remove audio streams; adding or removing streams requires a re-encode. If that's all accurate, then that'd explain how the two outputs aren't identical. Additionally, if ffmpeg is taking audioo stream bitrates into consideration, then logically the disk' [22:26:29 CET] <Keshl> s output is getting less bits on it's video stream since I have almost 1,000 extra bits of audio being sent there, while the network stream only has 1/4th of the audio bandwidth being used up. [22:27:23 CET] Action: Keshl checks his really old recordings to see if those are identical, since they only have one audio and one video stream. [22:27:34 CET] <JEEB> yes, FLV lets you only put a single audio and video track [22:27:43 CET] <JEEB> twitch takes in FLV [22:28:37 CET] <Keshl> Yep. Totally fine. [22:28:47 CET] <Keshl> Problem started when I started adding more audio tracks. [22:29:34 CET] <JEEB> well that's a single difference at least [22:29:49 CET] <Keshl> Am I right in assuming that, locally, if I want the four audio tracks, and I want to stream to Twitch at the same time, I absoutely have to encode twice? Once for MKV locally and once for FLV to send to Twitch? [22:30:12 CET] <BtbN> that's up to the software [22:30:23 CET] <JEEB> technically nothing stops you from pushing an AVPacket into multiple muxers [22:30:34 CET] <JEEB> AVPAcket being something an encoder throws out [22:30:43 CET] <JEEB> (while AVFrames are decoded raw samples= [22:30:57 CET] <Keshl> Please excuse my ignorance here.. Muxing is a seporate step after encoding and what actually stuffs the output from an encoder into a file container, right? [22:30:57 CET] <furq> even the ffmpeg cli can deal with that [22:31:00 CET] <kepstin> in fact, ffmpeg includes a muxer that lets you send the same packets to multiple muxers, and can be used to for this purpose [22:31:07 CET] <furq> muxing puts the elementary streams into a container, yeah [22:31:45 CET] <furq> but yeah if OBS actually does this then it sucks really badly [22:31:49 CET] <Keshl> Okay, nifty. I can try that. It lets me put albitrary command line stuff in a box somewhere, though I don't actually know what it does with it all. Is there documentation on how to achieve what you guys are talking about? [22:32:04 CET] <furq> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#Teepseudo-muxer [22:32:07 CET] <Keshl> (Like, with just straight ffmpeg on a command line.) [22:32:15 CET] <furq> that's how you'd do it with just ffmpeg [22:32:17 CET] <JEEB> did the tee muxer let you limit the streams? [22:32:19 CET] <Keshl> Thank'ye! [22:32:23 CET] <furq> i obviously wouldn't recommend capturing with ffmpeg though [22:32:26 CET] <JEEB> if this guy wants one video and audio into FLV [22:32:33 CET] <Keshl> "Obviously"? Why not? XD [22:32:36 CET] <JEEB> and then all tracks into something else [22:32:38 CET] <furq> JEEB: yeah it does [22:32:41 CET] <JEEB> cool [22:32:58 CET] <JEEB> I will probably still implement something similar in my stuff instead though [22:33:15 CET] <JEEB> because I'd rather re-init at failures if possible instead of killing all [22:33:20 CET] <kepstin> Keshl: ffmpeg doesn't include a high performance screen capture for windows suitable for capturing games [22:33:27 CET] <JEEB> yea [22:33:29 CET] <kepstin> Keshl: obs has its own screen capture [22:33:44 CET] <JEEB> the win8+ APIs would be a good addition but you'd have to get it async somehow [22:33:55 CET] <furq> Keshl: i guess you'd want -f tee "[select=\'v:0,a\']local0.mkv|[select=\'v:0,a:3\':f=flv]rtmp://server0/app/instance/playpath" [22:33:59 CET] <JEEB> virtualdub is funny enough the least bad "normie" app that does that API stuff [22:33:59 CET] <Keshl> ...Ho boy. Okay, this went over my head in like two seconds, mostly due to my complete failure to understand what -map actually does. I've read the manual on it, but it assumes I have a lot of knowledge about videos that I don't actually have. I don't think this is ffmpeg's fault, considering that most typical people won't ever need to mess with that option, but yeah, it's beyond me. <É<' [22:34:12 CET] <furq> you can ignore the -map stuff in that command [22:34:29 CET] <furq> that command has a bunch of filters that complicate the example [22:35:38 CET] <furq> also yeah like JEEB mentioned, the tee muxer will stop outputting all streams if one of them dies [22:35:47 CET] <Keshl> But I have to use -map myself for my own streams, and I'm not sure how that mixes in with this. See, OBS is doing /something/ to only send the "combined audio" channel to Twitch with my video, but locally, when I re-encode stuff to downscale it and whatnot, I have to do "-map 0" to actually preserve all the independant audio channels. Since I have more than one audio channel, wouldn't I have to use -map at some point with teemux to tell [22:35:48 CET] <furq> so in this case if the rtmp connection drops then it'll stop writing to disk [22:35:49 CET] <Keshl> it to preserve all the channels locally, but only send one over the network? [22:35:51 CET] <JEEB> it has an option for that at least [22:36:05 CET] <JEEB> either it keeps on with the left-around outputs [22:36:07 CET] <JEEB> or it all dies [22:36:15 CET] <furq> oh ok [22:36:35 CET] <furq> it won't attempt to reconnect to rtmp though which i assume OBS will [22:36:52 CET] <Keshl> (OBS does reconnect and re-start encoders, yes. When I d -- Hey, it uses teemux!) [22:37:11 CET] <furq> i don't think you'll need -map if you use tee directly [22:37:14 CET] <Keshl> (I was just about to type "When I drop connection, the disk recording stops too, until things are reconnected and stuff.") [22:37:29 CET] <furq> the select= stuff in the tee outputs are doing the same job [22:37:50 CET] <furq> select=a selects all audio streams for that output, but select=a:3 only selects the fourth stream [22:39:00 CET] <Keshl> Okay, I think this only leaves one question, and this'll be the nail in the coffin if you guys say yes. [22:39:22 CET] <Keshl> Suppose that I actually am using teemux and stuff, and the audio streams are correctly sent to where they're supposed to go, and everything's happy. [22:41:29 CET] <Keshl> This logically means that, when I tell OBS to limit it's self to 4,500 bps, it's /somehow/ telling that to ffmpeg. As far as I know, you have to specify the audio and the video bitrate seporately, with -b:a and -b:v. Suppose it does that, with -b:a 320 and -b:v 4180 (To account for the audio's rate). Obviously this works fine for one audio channel, but what happens when I have four? Does ffmpeg figure out what's going on and lower the [22:41:31 CET] <Keshl> video's bitrate to compensate for the increased audio streams? [22:41:56 CET] <furq> it doesn't figure anything out [22:42:01 CET] <furq> it just passes the bitrates you provide to the encoder [22:42:03 CET] <JEEB> the bit rates match what you set them for [22:42:11 CET] <JEEB> -b:a is "bit rate, (all) audio" [22:42:16 CET] <JEEB> so each audio encoder gets that bit rate [22:42:18 CET] <Keshl> Dangit. I mean, that's great, I prefer that, but dangit, this means now I have to go ask the OPBS guys what the heck they're doing. XD [22:42:20 CET] <Keshl> *OBS [22:42:32 CET] <JEEB> if they're doing FLV then... [22:42:36 CET] <JEEB> they have a single audio track? [22:42:44 CET] <JEEB> but generally speaking they just set AVCodecContext.bit_rate [22:42:49 CET] <furq> i've seen smart people recommending obs, so something about this doesn't seem right [22:42:52 CET] <JEEB> or whatever is the least retarded rate control mode for an encoder [22:42:54 CET] <furq> because this sounds pretty amateur hour [22:43:04 CET] <JEEB> OBS worked for me "OK" when I tested nvidia lossless output [22:43:13 CET] <Keshl> Oh, OBS is great, in my opinion. I'm just doing really weird stuff that almost no typical OBS user does, so I expect I might break something. [22:43:18 CET] <furq> maybe [22:43:46 CET] <JEEB> but then OBS seemed to be unable to write off that lossless video to my SSD well enough [22:43:55 CET] <JEEB> so when I stopped recording it would just flush its buffers for quite a while [22:44:00 CET] <Keshl> Oh my gosh. Yeah. I have no idea why, but thank you for telling me it's not just me. [22:44:47 CET] <Keshl> I have no idea why, but a few years ago I was trying to record a video just to disk (No network stuff), and whenever I wrote to a hard drive, nothing weird happened. SSD, constant massive frame drops for no conceivable reason, despite the obviously increased bandwidth. I was recording losslessely, so I thought for sure the HDD would be causing that, not the SSD. [22:45:36 CET] <Keshl> I still have the originals if you guys get curious or anything, but considering they're lossless 1080p at 60 FPS, they're, er, huge. [22:47:12 CET] <JEEB> my short test http://up-cat.net/p/a864aa70 [22:47:15 CET] <Keshl> https://www.youtube.com/watch?v=A7Njjy3Twhs if you wanna see the general output on YouTube. Again, recorded losslessely, but for some reason it only bugs out whenever there's a "complicated" series of frames that would be hard to encode with anything else. [22:47:28 CET] <Keshl> (I know this won't suffice for actual debugging, but still.) [22:47:28 CET] <JEEB> (I still find it funny that nvidia can encode this, but not decode) [22:50:48 CET] <Keshl> 2:38 in my video, great example of things going progressively wrong as the complexity increases. (Sorry for slowness, was looking for that.) [22:51:20 CET] <Keshl> JEEB: ...Is that one gigabit per second? [22:52:06 CET] <JEEB> 2 897 564 060 bytes for 23.47 seconds [22:52:37 CET] <furq> that looks about right for lossless 4k [22:52:57 CET] <furq> does nvdec not support 4:4:4 at all [22:53:05 CET] <JEEB> seemingly not [22:53:12 CET] <furq> that's pretty lame [22:55:19 CET] <Keshl> JEEB: That's.. that's a big number. [22:56:55 CET] <Keshl> ...Curiousity. I'm aware of what things like "4:4:4" refer to, but when you have an actual "4:4:4" metric, why don't you just call it "1:1:1"? Or "1::"? Isn't it just a ratio? [22:57:51 CET] <furq> https://en.wikipedia.org/wiki/Chroma_subsampling#Sampling_systems_and_ratios [22:58:01 CET] <furq> i couldn't tell you why that system was chosen [22:58:05 CET] <furq> i assume it's just convention [22:58:10 CET] <JEEB> it's a convention, yea [22:58:25 CET] <JEEB> I have the book circles of confusion - might explain it [22:59:03 CET] <furq> To calculate required bandwidth factor relative to 4:4:4 (or 4:4:4:4), one needs to sum all the factors and divide the result by 12 (or 16, if alpha is present). [22:59:10 CET] <furq> i didn't actually realise that [22:59:18 CET] <furq> i guess that's a useful property you might not get otherwise [23:00:46 CET] <Keshl> ...This is thr most helpful diagram I've seen in a while. o_o [00:00:00 CET] --- Fri Dec 14 2018

1 0

ffmpeg-devel.log.20181212
by burek 13 Dec '18

13 Dec '18

[01:44:57 CET] <cone-650> ffmpeg 03Lou Logan 07master:11817c3316d9: doc/indevs: document libdc1394 options [01:58:36 CET] <cone-650> ffmpeg 03Lauri Kasanen 07master:1046cba24be4: swscale/output: VSX-optimize nbps yuv2plane1 [04:08:35 CET] <cone-650> ffmpeg 03James Almer 07master:0e833f615b59: avcodec/libdav1d: add support for 12bit streams [12:51:18 CET] <durandal_1707> how to reset frame number in encoder AVCodecContext? [12:51:41 CET] <durandal_1707> from outside, via avformat_* [17:09:53 CET] <durandal_1707> ubitux: currently gif encoding via image2 muxer is broken, i fixed it on top of my patch by adding user option to encoder - there is no way to signal from muxer to encoder that instead? [17:11:43 CET] <ubitux> no idea [17:47:47 CET] <ubitux> durandal_1707: that duplicated offsetting logic is ugly [17:48:11 CET] <ubitux> i'd rather have the offsetting computation factored-out in a function and re-used [17:48:22 CET] <ubitux> (patch 2/6) [17:53:36 CET] <durandal_1707> ubitux: i do not see how [18:28:31 CET] <JEEB> > The x32 subarchitecture may be removed [18:28:34 CET] Action: JEEB claps [18:30:19 CET] <atomnuker> good riddance [18:32:56 CET] <iive> JEEB, macosx ? [18:33:05 CET] <JEEB> linux kernel [18:33:13 CET] <JEEB> x32 wasn't really a thing elsewhere I thought? [18:33:32 CET] <iive> is that the 32bit code executed in 64bit mode? [18:33:46 CET] <JEEB> > it runs the processor in the 64-bit mode, but uses 32-bit pointers and arithmetic. The idea is to get the advantages of x86-64 without the extra memory usage that goes along with it. [18:34:07 CET] <JEEB> except it came so late to the party nobody really cared about the overhead :P [18:34:15 CET] <JEEB> https://sites.google.com/site/x32abi/ [18:34:27 CET] <iive> yeh. it never took off. [18:51:42 CET] <ePirat> JEEB, iirc apple does it for its new watch or something [19:43:16 CET] <durandal_1707> atomnuker: why are you against fixing that serious bug in opus encoder? [19:49:00 CET] <atomnuker> its not a serious bug [19:49:43 CET] <atomnuker> there might be a small difference but we still decode the testvectors correctly (according to one interpretation) [19:50:16 CET] <atomnuker> to begin with checking for that bug would involve checking the entire celt part of the decoder against the reference [19:50:40 CET] <nevcairiel> he said encoder [19:50:42 CET] <durandal_1707> ENCODER [19:51:08 CET] <atomnuker> ah [20:10:16 CET] <atomnuker> why the hell is the afq adding the overlap into remaining_samples on init? [20:11:32 CET] <durandal_1707> you coded it that way? [20:13:19 CET] <atomnuker> no, it was justing ruggles, who went to vimeo never to be heard from again [20:13:59 CET] <durandal_1707> atomnuker: where is that code? [20:14:01 CET] <atomnuker> I think its a hack to adjust the delay (and hence the timestamps), and it wasn't taken into account the very first frame might be null and the encoder might depend on the afq [20:14:08 CET] <atomnuker> audio_frame_queue.c [20:33:01 CET] <atomnuker> grr, I'd rather not touch anything related to afq or rely on its proper initialization to check whether to flush, as that'll nullify its error-detection logic [20:33:34 CET] <atomnuker> I'll do it via the encoder state [21:10:00 CET] <cone-980> ffmpeg 03Rostislav Pehlivanov 07master:83db1efd42bd: opusenc: fix infinite loop if flushing encoder upon init [22:38:00 CET] <thardin> duckduckgo bought on2's old duck.com [22:40:30 CET] <JEEB> classic level domain :) [22:41:38 CET] <gnafu> Time to update all my bookmarks. [23:54:52 CET] <cone-980> ffmpeg 03Carl Eugen Hoyos 07master:32601fb82117: lavfi/signalstats: Cast the return value of AV_RN16() to int. [00:00:00 CET] --- Thu Dec 13 2018

1 0

ffmpeg.log.20181212
by burek 13 Dec '18

13 Dec '18

[00:17:01 CET] <xiando> ffmpeg -i "$a" -vn -codec:a libvorbis -qscale:a 9 "${a[@]/%flac/ogg}" creates a ogg file with song metadata but no cover image. without -vn ffmpeg makes a video track with the cover which makes it unplayable in mpd and audacious with the ffmpeg plugin but not with that disabled. [00:17:21 CET] <furq> the ogg muxer doesn't support cover art [00:17:43 CET] <furq> oggenc and opusenc both support flac input so just use those [00:18:00 CET] <xiando> Oh. So I have been searching for solution all this time and there is none. :) [00:18:26 CET] <xiando> I did try oggenc but I didn't get the cover image with that either. [00:20:31 CET] <furq> nvm apparently opusenc supports it but oggenc doesn't [00:20:40 CET] <furq> you should be using opus anyway nowadays unless you need to support a specific device [00:23:02 CET] <xiando> I must support the specific device, a Benjie T2 which could support Rockbox and have the Opus support but not Bluetooth [00:24:03 CET] <xiando> My Xiaomi Redmeme 4 also doesn't play Opus OGG files which is interesting because it plays webm files with VP9 video and Opus audio just fine [00:26:54 CET] <xiando> My choice is flac or convert to vorbis or aac or mp3 since those are supported by music player and the phone. [01:19:07 CET] <Hello71> put the audio in mkv then?? [01:19:23 CET] <Hello71> well i guess it isn't v [01:19:28 CET] <Hello71> mka [01:20:13 CET] <furq> this is some wonky android bug isn't it [01:20:20 CET] <furq> iirc it'll work if you just rename the ogg files to .opus [01:45:20 CET] <xiando> furq: sadly, no [01:45:52 CET] <xiando> I also have a tablet with an earlier Android version and volla ogg and opus files with opus audio work just fine. [01:46:36 CET] <xiando> The no opus in ogg containers or .opus but it's fine with .mkv except audio player can't recognize those files seem very specific to Xiaomi Redmeme 4 [01:49:04 CET] <xiando> Opus doesn't work on my benjie music player anyway. Unless I flash rockbox. But then I can't have bluetooth. Guess I could also flash custom ROM on the phone but that's usually a similar disaster, always some "minor" thing like the front camera or that stops working [06:02:53 CET] <konrados> Hi! I have a question regarding making animated png. Here I put the files, both source and converted - https://kpion.github.io/tests/ffmpeg/ - first go the 'OK' ones, a "typical" *movie* (mp4) converted to apng looks OK. Then I tried to convert another mp4, this time an "animation" of text being entered slowly. You can see it works fine when converted to gif, but the animated png seems to 'catch' only a few frames [06:03:33 CET] <konrados> I tried to google but I can't see any options when it comes to 'animated png' except for looping with the 'plays' option, but that doesn't change anything [06:04:29 CET] <konrados> I've just started toying with ffmpeg yesterday, so please forgive me my ignorance :) The command was simply `ffmpeg -i lorem.mp4 lorem.apng` [06:04:43 CET] <TheAMM> You could try -vsync 0 but I don't see why apng would drop frames like that, and don't know any further [06:05:20 CET] <konrados> ok, I'll try, thanks! [06:11:42 CET] <konrados> nothing changed :( I tried `ffmpeg -i lorem.mp4 lorem.apng -vsync 0` as well as `ffmpeg -i lorem.mp4 -vsync 0 lorem.apng` - same with vsync 1, with no success. [06:12:22 CET] <furq> are you sure you want to use apng [06:12:36 CET] <furq> i guess chrome and safari finally started supporting it but it's still not a great format [06:14:26 CET] <konrados> furq, well, no, it looks like I'll go with .gif, I thought apng is already ... stable and common :) So yeah, I'll just use .gif, thanks! [06:14:51 CET] <konrados> it's 10 years since initial release according to wiki :) [06:15:17 CET] <furq> it's more like 14 years [06:15:29 CET] <furq> there was a big fuss a while back because some people supported mng and some people supported apng [06:15:35 CET] <furq> and then that never got resolved until like last year [06:16:15 CET] <furq> but now it looks like firefox and safari are going to support webp, so that's probably a better choice if you need lossless 24-bit animations that work in browsers [06:16:33 CET] <furq> edge already supports webp and will probably never support apng [06:18:12 CET] <furq> anyway if you're making gifs with ffmpeg then make sure to use palettegen/paletteuse [06:21:29 CET] <konrados> furq, oh wow, with you and wiki I learned about mng and webp :) Thanks! And yeah, I did find http://blog.pkh.me/p/21-high-quality-gif-with-ffmpeg.html yesterday but then thought that it's a lot of effort to bring an old 256 colors technology to modern day, and that maybe apng is the way... now it turns out it apparently isn't (yet). So indeed I'll stick to this one, thanks again! [06:21:57 CET] <konrados> *this one == gif [08:49:54 CET] <Karel|> Hey guys! I just released a first version offfmpeg-cpp, a easy-to-use C++ wrapper around the libav* libraries [08:49:57 CET] <Karel|> check it out here: [08:49:58 CET] <Karel|> https://github.com/Raveler/ffmpeg-cpp [08:50:14 CET] <Karel|> might be useful if you want to integrate libav*/ffmpeg functionality in your project without having to resort to command-line [10:16:57 CET] <tommy``> is it possible download with ffmpeg segmented video from a stream (list of .m4s files) then merge togheter? [10:36:16 CET] <friendofafriend> tommy``: Take a look at the concatenate filter. https://trac.ffmpeg.org/wiki/Concatenate [10:36:33 CET] <tommy``> hey friendofafriend, good day ;) [10:38:29 CET] <friendofafriend> Howdy, tommy``! ffmpeg sure is handy for operations like merging video. [10:52:56 CET] <guoyunhe> Hi, how could I install ffserver? I am using openSUSE and installed ffmpeg package. But cannot find ffserver command. [10:53:45 CET] <furq> ffserver was removed a few releases back [10:54:04 CET] <furq> it was unmaintained and also it never really worked properly anyway [10:54:10 CET] <guoyunhe> furq: Is here any alternative solution? [10:54:17 CET] <furq> what are you trying to do [10:54:37 CET] <guoyunhe> I want to stream my desktop over the web [10:55:08 CET] <guoyunhe> so someone in the same wifi can watch my screen casting [10:55:31 CET] <furq> if you don't need low latency then nginx-rtmp is the usual recommended streaming server [10:56:07 CET] <guoyunhe> No need low latency. [11:00:19 CET] <friendofafriend> I'm using Icecast with no complaints. [11:00:38 CET] <furq> are you using webm [11:00:52 CET] <furq> or are you one of those mythical people who's been able to make h264 over icecast work properly [11:02:31 CET] <friendofafriend> That's mythical? [11:02:45 CET] <furq> people keep telling me it works but it never works well for me [11:02:57 CET] <furq> it'll either work perfectly or bail out before connecting [11:04:01 CET] <friendofafriend> I'm using it in an MPEG-TS. [11:06:00 CET] <friendofafriend> Something like this: ffmpeg -re -i /dev/video0 -preset ultrafast -vcodec libx264 -tune zerolatency -b 1400k -content_type 'video/ts' -f mpegts icecast://source:password@server:port/file.ts [11:08:38 CET] <guoyunhe> thanks for the information. i will try to play it by myself. [11:24:24 CET] <guoyunhe> friendofafriend: Is my URL correct? icecast://source:hackme@localhost:8754/stream.ts [12:08:42 CET] <guoyunhe> I tried icecast and it works very well. [12:09:40 CET] <guoyunhe> It has about 1s latency for 1080p desktop streaming in WebM format [14:39:18 CET] <HickHoward> i need help with some out-there format [14:50:22 CET] <durandal_1707> HickHoward: what ffmpeg -i file says? [14:50:25 CET] <relaxed> HickHoward: be more specific [14:53:05 CET] <HickHoward> i'll send a sample soon enough [14:56:02 CET] <HickHoward> https://www.sendspace.com/file/3oyqbs [15:00:14 CET] <HickHoward> i might have a lot of samples hanging around my back so... [18:30:40 CET] <forgon> I'd like to convert RPL to OGG with the best possible quality. [18:31:27 CET] <forgon> Unfortunately, I am clueless about what I should do. So far I am executing `ffmpeg -i input.rpl output.ogg` and the quality is quite poor. [18:33:30 CET] <relaxed> try, ffmpeg -i input.rpl -q:a 10 output.ogg [18:34:16 CET] <relaxed> I think the range is 1 - 10, with 3 being the default [18:36:15 CET] <forgon> relaxed: Does a apply only to audio? [18:38:30 CET] <relaxed> sorry, I assumed audio only [18:38:55 CET] <relaxed> https://trac.ffmpeg.org/wiki/TheoraVorbisEncodingGuide [18:39:45 CET] <relaxed> why do you need ogg? [18:40:53 CET] <forgon> relaxed: Used by the project I'm trying to help, probably a choice made based on licensing and hardware compatibility. [18:42:24 CET] <relaxed> https://trac.ffmpeg.org/wiki/Encode/VP9 would be considered ogg's replacement [19:15:56 CET] <kerio> hardware compatibility? [19:16:06 CET] <kerio> for ogg vorbis? [19:43:46 CET] <RingoTheDog> I would like to preserve/copy the EIA-608/EIA-708 captions ( Ancillary data / CDP) in an MXF while added a watermark... [19:43:53 CET] <RingoTheDog> CLI: ffmpeg -i IN.MXF -i WATERMARK.PNG -filter_complex "[0:v][1:v] overlay=(W-w)/2:(H-h)/2" -pix_fmt yuv420p -vcodec mpeg2video -flags +ildct+ilme -top 1 -b:v 35000k -minrate 35000k -maxrate 35000k -bufsize 36408333 -g 15 -bf 2 -aspect 16:9 -r 29.97 -s 1920X1080 -map 0:a -f mxf -y OUT.MXF [19:44:01 CET] <RingoTheDog> I have tried -c:s copy & -map 0:S ...no luck [21:17:16 CET] <pixelou> Hi everyone, I'm an ffmpeg newbie and I am a bit confused with the deprecated and new C APIs. The modern way to decode a video is via avcodec_receive_packet which takes an AVCodecContext but the avformat API does not provide such an object, so how am I supposed to feed packets? [21:23:27 CET] <Alina-malina> is there a way to stream to youtube channel an mp4 file with ffmpeg? [21:23:36 CET] <furq> Alina-malina: youtube-dl [21:23:55 CET] <furq> either just use youtube-dl or use youtube-dl -g to get a url you can give to ffmpeg [21:24:16 CET] <Alina-malina> hmmm to stream to my youtube channel? [21:24:36 CET] <furq> oh right sorry [21:24:51 CET] <furq> -i foo.mp4 -c copy -f flv rtmp://youtube.whatever [21:25:08 CET] <Alina-malina> oh thats great, thanks let me try that [21:25:28 CET] <furq> if the mp4 is h264 and aac then that'll work, otherwise you'll need to reencode the video and/or the audio [21:26:54 CET] <Alina-malina> what about the key? i need to provide a key too [21:28:56 CET] <furq> rtmp://youtube.url/key [21:29:28 CET] <Alina-malina> oh yes yes, jsut a moment please let me try [21:30:42 CET] <Alina-malina> ok it worked thank you so much friend [00:00:00 CET] --- Thu Dec 13 2018

1 0

ffmpeg-devel.log.20181211
by burek 12 Dec '18

12 Dec '18

[00:42:58 CET] <cone-680> ffmpeg 03Carl Eugen Hoyos 07master:73abde61bb07: lavf/matroskadec: Do not use strncat() to limit copying a one-char constant. [00:50:57 CET] <cone-680> ffmpeg 03Carl Eugen Hoyos 07master:73251678c83c: lavd/v4l2: Use "int request" as second parameter for ioctl() on Android. [00:57:32 CET] <cone-680> ffmpeg 03Carl Hetherington 07master:6190f873eb6d: lavc/tiff: Fix leak of yuv_line in TiffContext. [01:20:40 CET] <cone-680> ffmpeg 03Lou Logan 07master:0e8eb07980de: doc/indevs: fix hdvbuffer typo in iec61883 example [06:31:06 CET] <cone-709> ffmpeg 03kjeyapal(a)akamai.com 07master:1d01ab399fc7: avformat/movenc: Added an option to disable SIDX atom [06:31:06 CET] <cone-709> ffmpeg 03kjeyapal(a)akamai.com 07master:876ed08b0d13: avformat/dashenc: Used the movenc option skip_sidx instead of global_sidx [10:53:19 CET] <Compn> i wonder if you can ssh > psexec localhost hmmmm [10:53:55 CET] <Compn> ah yes you can [10:54:02 CET] <Compn> that seems like the workaround then :) [12:22:49 CET] <cone-456> ffmpeg 03Peter Ross 07master:b41f378fc4dd: os_support: define socket shutdown SHUT_xxx macros if they are not defined [14:28:01 CET] <Compn> wat [14:28:06 CET] <Compn> carrrrrrrrl [16:22:16 CET] <cone-453> ffmpeg 03Gyan Doshi 07master:3b825b2f81e7: fate: add prog suffix in fate-mov-mp4-with-mov-in24-ver [16:43:36 CET] <cone-453> ffmpeg 03James Almer 07master:9c49933a3e16: configure: require the first stable version of libdav1d [17:18:04 CET] <jya> with mp4, in a stsd box defining multiple entries, each with a different codec valid? and if so does ffmpeg handle those? [17:19:18 CET] <JEEB> hmm [17:19:26 CET] <JEEB> 14496-12 for the first question [17:19:31 CET] <JEEB> second will be answered after the first [17:21:22 CET] <JEEB> > quantity: exactly one [17:21:55 CET] <JEEB> so at least in a single stbl there should be one and only one stsd [17:22:18 CET] <JEEB> ok, so stbl is also exactly one [17:22:21 CET] <JEEB> which is in minf [17:22:34 CET] <JEEB> ok, exactly one minf in mdia [17:22:48 CET] <JEEB> exactly one mdia in trak [17:23:13 CET] <JEEB> and of course there can be N traks [17:23:31 CET] <JEEB> jya: does this answer your first question? [17:24:19 CET] <jya> JEEB: well, a stbl has a single stsd yes [17:24:37 CET] <jya> but stsd can have multiple entries, for video a VisualSampleEntry [17:24:59 CET] <JEEB> right, I missed that part and started going from stsd upwards :P [17:25:33 CET] <jya> and each VisualSampleEntry can define a codec [17:25:42 CET] <jya> i've just never seen those files in the wild [17:26:45 CET] <jya> we started to have to look into this case, because for some encrypted file, we see multiple encryption data stored across multiple entries [17:28:14 CET] <JEEB> 1) multiple SampleDescriptions are of course valid [17:28:37 CET] <JEEB> now 2) would be regarding actually different video/audio formats, or just a different identifier for the same stuff? [17:29:28 CET] <jya> different all together [17:30:19 CET] <jya> in firefox, we kind of allow different h264 content in the stream, but that only works because we assume it's all h264 and parse the raw bytestream to check if there are inband SPS [17:30:46 CET] <jya> if you were to say have a group of sample using h264 and nother using vp9 that won't do [17:30:49 CET] <JEEB> yea, I think we should have support for multiple initialization data [17:30:55 CET] <JEEB> for the same codec [17:31:11 CET] <JEEB> but not the "lol these samples have completely different content" [17:31:15 CET] <jya> yes, same codec can do. different opens a bug can or worms [17:31:19 CET] <jya> of worms [17:31:32 CET] <JEEB> yes [17:31:41 CET] <JEEB> I'm not sure if that's technically allowed in 14496-12 [17:31:49 CET] <JEEB> will have to take a look a bit at where different IDs are [17:32:08 CET] <JEEB> (of course in the worst case it'll be "lol ISOBMFF" and someone goes get his whiskey) [17:33:41 CET] <jya> JEEB: i can't see anything in there related to the use of different values for the codec type [17:34:01 CET] <JEEB> yea [17:34:02 CET] <jya> now I only have the 2015 version of it, that may have changed [17:34:08 CET] <JEEB> because it's in stsd [17:34:17 CET] <JEEB> yes, technically it ain't saying anything [17:34:19 CET] <JEEB> (jesus christ) [17:34:53 CET] <jya> looking at their stsd summaries: sample descriptions (codec types, initialization etc.) [17:34:58 CET] <jya> its all plural :) [17:35:08 CET] <JEEB> yes, there's no question about that [17:35:32 CET] <JEEB> I mostly had to verify where the format identifiers were again by looking at boxdumper's output in less :P [17:35:34 CET] <jya> not in 8.5.2.1 however [17:35:46 CET] <jya> "The sample description table gives detailed information about the coding type used, and any initialization information needed for that coding." [17:35:50 CET] <jya> that's singular [17:36:23 CET] <jya> we've written our own moov parser in rust [17:36:27 CET] <JEEB> yes, but the actual ID is in the sample descriptor [17:36:48 CET] <jya> due to the multiple security issues we got when we unfortunately decided to use stagefright [17:36:48 CET] <JEEB> > Multiple descriptions may be used within a track. [17:37:08 CET] <JEEB> which is in 8.5.2.1 on the next page [17:37:10 CET] <jya> (reason for stagefright was that ffmpeg had no support for fragmented mp4 back then) [17:38:09 CET] <jya> luckily there can only be 65536 entries :) [17:39:45 CET] <JEEB> and yea, stsc lets you switch around [17:39:48 CET] <JEEB> christ this is perverse [17:40:03 CET] <JEEB> at least if you take all the freedoms the spec (seems to) give you [17:48:58 CET] <jya> we're going to parse all entries for the various configuration available, but error if the codec ever change. [17:49:01 CET] <jya> thanks for your help [17:49:52 CET] <JEEB> yes, that sounds like the correct way to do it unless you want a world of pain [17:56:27 CET] <durandal_1707> who bumps versions ALLLL THE TIME, STOOOOOOOOOOOOP!!!!!!!!! [17:57:52 CET] <cone-453> ffmpeg 03Paul B Mahol 07master:dd7d6034f1fe: avfilter/vf_paletteuse: fix flags for alpha_threshold option [19:14:18 CET] <cone-453> ffmpeg 03Carl Eugen Hoyos 07master:464aa60eb5e5: tests/api-flac-test: Rename NUMBER_OF_FRAMES as NUMBER_OF_AUDIO_FRAMES. [20:12:45 CET] <kierank> durandal_1707: nice project for you [20:12:58 CET] <kierank> dealing with masterpiece that is mpegvideo.c, h263.c and ituh263.c [20:15:04 CET] <atomnuker> I wonder if you can make an mpeg4 version of https://pics.onsizzle.com/the-the-father-is-not-ll-on-is-god-0-34080567.png [22:03:39 CET] <durandal_1707> kierank: i do not want crappy toys for xmas [00:00:00 CET] --- Wed Dec 12 2018

1 0

ffmpeg.log.20181211
by burek 12 Dec '18

12 Dec '18

[02:00:34 CET] <tytyt> Is it possible to convert from a memory stream to a file using ffmpeg? [02:02:40 CET] <furq> if you can pipe it into ffmpeg, sure [02:08:48 CET] <tytyt> I'm using a wrapper right now, but does it even take in bitmaps? [02:09:34 CET] <furq> -f rawvideo i - [02:09:37 CET] <furq> -i - [02:09:54 CET] <furq> or if you mean bitmaps as in .bmp files then -f image2pipe -i - [02:10:10 CET] <tytyt> Awesome! [02:10:15 CET] <furq> rawvideo is just raw pixel data so you'll need to specify the resolution and framerate [02:10:56 CET] <furq> -video_size 123x456 -framerate 25 -f rawvideo -i - [02:11:24 CET] <jin> how do i increase contrast for a mp4 [02:11:50 CET] <TheRock> you can do it in your monitor [02:12:10 CET] <jin> no srsly [02:12:27 CET] <furq> colorlevels, curves, eq, etc [02:12:27 CET] <jin> looking for an example command to increase contrast [02:12:41 CET] <furq> https://ffmpeg.org/ffmpeg-filters.html#eq [02:14:32 CET] <jin> trying to find the most simple example [02:14:50 CET] <jin> is 1 the default? [02:14:53 CET] <jin> for everything? [02:17:38 CET] <tytyt> "you can do it in your monitor" [02:17:39 CET] <tytyt> Lol [02:18:33 CET] <jin> idk which is more painful to learn ffmpeg or blender [02:19:10 CET] <jin> I imagine I could write a command like: ffmpeg -i infile -vf contrast=1.5 outfile [02:22:30 CET] <jin> https://www.ffmpeg.org/ffmpeg-filters.html#toc-eq why is there no examples here? [02:28:23 CET] <fred1807> where can I learn to understand the x1,y1,x2,y2 rationale for video playback croping? [02:30:00 CET] <jin> ffmpeg -i in.mp4 -vf mp=eq2=gamma:contrast:brightness:saturation:rg:gg:bg:weight out.mp4 -y a 3 year old commentgave this example [02:30:32 CET] <jin> do I really have to do all the values like mp=eq=1:1.5:1:1:1:1:1:1 just to increase contrast by 50% [02:30:50 CET] <jin> seems excessive [02:31:01 CET] <furq> no you don't and also don't use mp [02:31:12 CET] <furq> just -vf eq=contrast=2 [02:31:49 CET] <jin> okay and replace ffmpeg with ffplay to preview? [02:32:05 CET] <jin> 2 = increase contrast 100%? [02:32:47 CET] <jin> guess I'll trial and error [02:41:05 CET] <tytyt> So, the Licence agreement for FFmpeg is not met for my project so I'm wondering if anyone has any other options for encoding to AV1 format? [02:41:53 CET] <TheRock> why is it not met [02:42:17 CET] <furq> you can use libaom and libwebp directly [02:42:21 CET] <furq> those are bsd derived iirc [02:42:39 CET] <furq> although yeah you should be able to comply with the lgpl [02:43:14 CET] <tytyt> I was under the understanding that libaom did not support encoding. [02:43:18 CET] <tytyt> I was just being stupid then. [02:43:34 CET] <TheRock> linking dynamic is fine for lpgl [02:43:38 CET] <furq> also i meant libwebm obviously [02:43:54 CET] <tytyt> On no sorry, I meant encoding from png or bitmap. [02:44:39 CET] <tytyt> I'm trying to encode images to AV1 [09:26:33 CET] <friendofafriend> Hello. I've got a concatinated MP4 audio file. When I run "ffmpeg -i inputfile -c copy outputfile", ffmpeg gives the expected output with a "Found duplicated MOOV Atom. Skipped it" warning. [09:27:06 CET] <friendofafriend> Is there a way to split the output file where the duplicate MOOV Atom is found? [11:18:34 CET] <RazWelles> Is nvenc supported on msys for ffmpeg? [11:18:56 CET] <RazWelles> I'm about to open a new project but I can't tell which compiler will play nice for the two [11:19:57 CET] <BtbN> should work. I'd recommend msvc though [11:24:15 CET] <JEEB> msys isn't what you'd build FFmpeg with for windows, though :P [11:24:32 CET] <JEEB> although msys2 is often the environment in which you do the building within windows (unless you're doing WSL) [11:24:50 CET] <JEEB> because mingw-w64 is what does the (normal) windows binaries [11:25:04 CET] <JEEB> msys(2) binaries are such that depend on msys(2)'s cygwin.dll etc [11:25:16 CET] <JEEB> heck, I call MSVC from msys(2)'s environment, too :P [12:07:30 CET] <RazWelles> JEEB: Oh geez I didn't even know you could do that [12:07:41 CET] <RazWelles> msvc it is [12:07:44 CET] <RazWelles> thank you [14:10:23 CET] <andbubble> Hi [14:10:30 CET] <andbubble> How can I install FFMPEG with all things enabled by default? I tried using the docker container but it doesn't have libass for subtitles [14:11:04 CET] <andbubble> I tried with both Alpine and Ubuntu versions [14:17:35 CET] <masterkorp> hello everyone [14:17:59 CET] <masterkorp> I want to concatenate a series of videos, but only after a few seconds o each video, is that possible? [14:25:17 CET] <andbubble> I disconnected, did anyone answer the question I posted about using docker or other package to get a complete FFMPEG installation with all the libraries like lib-ass [14:30:33 CET] <mateothegreat> @andbubble: pilfer from https://github.com/jrottenberg/ffmpeg [14:35:53 CET] <andbubble> @mateothegreat I tried using that docker container [14:35:58 CET] <andbubble> but it doesn't have evrything enabled [14:36:31 CET] <mateothegreat> right, so clone the Dockerfile and enable it yourself ^_^ [14:36:44 CET] <andbubble> But in the dockerifle it seems to have the part for enabling libass [14:37:22 CET] <andbubble> ie. here: https://github.com/jrottenberg/ffmpeg/blob/master/docker-images/4.1/ubuntu/… [14:39:49 CET] <mateothegreat> I don't understand your problem [14:39:54 CET] <andbubble> and here https://github.com/jrottenberg/ffmpeg/blob/master/docker-images/4.1/ubuntu/… [14:40:08 CET] <andbubble> well I went ahead, used that docker container but when I run ffmpeg --help [14:40:13 CET] <andbubble> it doesn't show --enable-libass [14:42:34 CET] <andbubble> see: https://hastebin.com/idetorabig.bash [15:04:08 CET] <BtbN> You'll have a hard time enabling _everything_ [15:16:44 CET] <andbubble> Any solution to 'No usable fontconfig configuration file found, using fallback. [15:16:54 CET] <andbubble> I don't get any subtitles burned to video [15:40:42 CET] <Karel|> hey guys [15:40:49 CET] <Karel|> I'm working on a C++ wrapper around the ffmpeg library [15:41:22 CET] <Karel|> it's going quite well, but I have a few remaining issues I can't seem to fix [15:42:03 CET] <Karel|> anyone here I could bounce some things off of? [15:43:15 CET] <Karel|> right now I'm struggling with the mpeg2video codec [15:43:44 CET] <Karel|> if I use the command line tool, it somehow sets the proper bit_rate somewhere in the code since the final result looks well, but in my system it doesn't [15:43:58 CET] <Karel|> I've stepped through the entire ffmpeg.c file to figure out where it happens and I can't find it [15:44:57 CET] <Karel|> this is the command I'm running, so nothing fancy there, just all defaults: ffmpeg.exe -i .\samples\big_buck_bunny.mp4 -c:v mpeg2video bla.mpg [15:45:42 CET] <Karel|> nothing different in the codecContext's either, the bit_rate, global_quality etc settings are identical in my system and in ffmpeg.c [17:51:39 CET] <visub> when I try to write a bash script with ffmpeg to find PSNR and SSIM for multiple files I get "ffmpeg:command not found" error. the same command works fine from the terminal [17:52:15 CET] <visub> Could someone tell me what I could be doing wrong? [17:52:58 CET] <pink_mist> you've changed your PATH [17:57:35 CET] <visub> Thank you:) [18:11:41 CET] <dv_> does anybody have VP9 examples with altref frames? [18:11:52 CET] <dv_> I have trouble finding videos with altref frames in them [18:12:06 CET] <dv_> or dont VP9 bitstreams have invisible frames? [18:15:48 CET] <JEEB> dv_: they have the concept at least, that's the way they kept having totally-not-bframes [18:15:49 CET] <realies> is there any way to get more info on how a particular mp3 file is broken? [18:16:02 CET] <realies> i think ffmpeg died because it was out of memory and it left out a partial file [18:16:45 CET] <dv_> JEEB, and I've seen these frames in action in various VP8 streams. not in VP9 ones yet though. [18:52:13 CET] <realies> actually I get LAME3.99.5 and many AAAAAAA chars at the end of the file, seems like it finished writing to file correctly? [18:52:14 CET] <kepstin> realies: mp3 files are pretty simple, the only issue you might have with a partial file is some tools report the wrong length or have strange seek bar behaviour [18:53:56 CET] <realies> ffprobe only says [warning] Estimating duration from bitrate, this may be inaccurate [18:54:17 CET] <realies> but then the file seems to have a correct footer (just looking at the hex viewer) [19:02:17 CET] <realies> is there any way to figure what in particular went wrong? [20:18:29 CET] <furq> realies: you can try mp3diags [20:43:47 CET] <jin> how do I shave 2 pixels from the top of a video? [20:43:53 CET] <jin> off the top* [20:46:54 CET] Action: jin searches through pages of documentation looking for an answer [20:47:07 CET] <furq> jin: if it's h264 then you can do it without reencoding [20:47:16 CET] <jin> oh man I already converted to webm :( [20:47:24 CET] <jin> but thats okay [20:47:40 CET] <furq> did you not keep the source [20:47:44 CET] <jin> of course I did [20:47:51 CET] <furq> ok good [20:48:11 CET] <jin> I have a collection of videos with minor imperfections now though, a few pixels here n there I gotta shave off [20:48:14 CET] <furq> well yeah -bsf:v h264_metadata=crop_top=2 to do it without reencoding [20:48:23 CET] <jin> Sweet [20:48:30 CET] <furq> otherwise -vf crop=in_w:in_h-2:0:2 [20:51:19 CET] <jin> thanks furq ! [20:51:29 CET] <jin> I think I understand crop filter a lot more now [00:00:00 CET] --- Wed Dec 12 2018

1 0