November 2019 - Ffmpeg-devel-irc

ffmpeg-devel.log.20191104
by burek 05 Nov '19

05 Nov '19

[03:27:45 CET] <cone-356> ffmpeg 03Jun Zhao 07master:8f7a04360955: lavf/mpegtsenc: fix logic check error [03:27:45 CET] <cone-356> ffmpeg 03Jun Zhao 07master:ec3d3fc65d06: lavf/asfdec: Simplify the check conditions [10:47:02 CET] <jdarnley> Does anyone know what program uses ~/.debug and stores a lot of data there? [10:47:08 CET] <jdarnley> Is it gdb? [10:47:19 CET] <jdarnley> or maybe valgrind? [10:49:21 CET] <ubitux> sounds like perf [10:49:37 CET] <ubitux> no? [10:51:25 CET] <ubitux> jdarnley: man perf-config [10:51:36 CET] <jdarnley> thanks [10:51:40 CET] <ubitux> > The recording tools also stores a hard link or copy in a per-user directory, $HOME/.debug/ [16:35:28 CET] <cone-844> ffmpeg 03zhongli_dev(a)126.com 07master:fcbfdeeabe21: lavc/qsvenc: add the missing vp9 file [16:35:28 CET] <cone-844> ffmpeg 03Zhong Li 07master:f52e15df0f6a: tools/probetest: replace the deprecated API [16:35:28 CET] <cone-844> ffmpeg 03Zhong Li 07master:8ff432eb447e: tools/enum_options: replace the deprecated API [22:15:32 CET] <cone-107> ffmpeg 03Lou Logan 07master:132bc8817e66: configure: select jpegtables for rtp muxer [22:15:32 CET] <cone-107> ffmpeg 03Lou Logan 07master:16ccafad9592: configure: select riffdec for act demuxer [22:15:32 CET] <cone-107> ffmpeg 03Lou Logan 07master:82bb7900db50: avformat/Makefile: add missing pcm dependency to hcom demuxer [22:15:32 CET] <cone-107> ffmpeg 03Lou Logan 07master:5f78ca83d61a: avformat/Makefile: add missing pcm dependency to nsp demuxer [22:15:32 CET] <cone-107> ffmpeg 03Lou Logan 07master:cae7f6658cf3: avformat/Makefile: add missing pcm dependency to sdx demuxer [00:00:00 CET] --- Tue Nov 5 2019

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ffmpeg.log.20191104
by burek 05 Nov '19

05 Nov '19

[06:37:40 CET] <edenist> hi everyone o/ [06:50:14 CET] <edenist> I wonder if someone would be able to offer some insite into some latency inconsistencies with different values for -crf on libx264 encodings [06:50:19 CET] <edenist> *insight [12:08:57 CET] <BeerLover> How to upload ffmpeg directly to AWS S3? https://gist.github.com/abhyuditjain/080449ae91588dae3d38d551a4c5353f [12:09:31 CET] <BeerLover> I don't want to create files locally and want to save the segment files and index.m3u8 directly in S3 bucket. [12:12:08 CET] <BtbN> you'll have to write something yourself that does that [12:13:26 CET] <DHE> I'm no s3 expert, but ffmpeg does support HTTP PUT, so if you can give ffmpeg the necessary authentication headers shouldn't it work? [12:13:26 CET] <BeerLover> -method PUT ? [12:14:02 CET] <BtbN> I don't think uploading to S3 is as simple as HTTP PUT [12:14:17 CET] <BtbN> you need to use their API, with OAuth2 auth flow, and a bunch of other stuff [12:16:22 CET] <BeerLover> BtbN DHE you can generate Presigned urls [12:16:32 CET] <BeerLover> with PUT as HTTP Verb [12:16:46 CET] <BeerLover> how will segment%d.ts work??? [12:17:04 CET] <BtbN> If you can only do that per-file, that won't help you a whole lot. [12:18:30 CET] <BtbN> https://docs.aws.amazon.com/AmazonS3/latest/API/API_PutObject.html#API_PutO… looking at this, this won't be a reasonably possible without custom code [12:21:13 CET] <BeerLover> Okay. Then how can I use piping for it? [12:21:17 CET] <BeerLover> for one file it' [12:21:20 CET] <BeerLover> s easy [12:21:46 CET] <BeerLover> ffmpeg -i <> -f <> - | aws s3 cp - s3://<> [12:21:52 CET] <BeerLover> but for segments??? [12:22:08 CET] <BtbN> you can't [12:57:04 CET] <Kadigan> Hey. So I just found out ffmpeg supports LUTs. Sadly, I can't seem to get it to work. No matter what I specify, I get "[Parsed_lut3d_0 @ (address)] 3D LUT is empty". Here's my commandline, output, and version: https://8n1.org/16190/30f8 [12:58:20 CET] <Kadigan> Maybe it's my cube format/value format, I don't know. Not sure how to even -verify- this, as I've no idea if there are different formats, and if so - which one ffmpeg supports. [12:58:51 CET] <Kadigan> The LUTs I got from AbelCine, url to download package: https://abelcinesalesforce2.s3.amazonaws.com/article/canon-luts-for-the-c10… [12:58:58 CET] <Kadigan> Ideas? [14:03:31 CET] <kepstin> Kadigan: what does "ffprobe ./CxxxLog10toRec709_Legal.cube" say? [14:06:45 CET] <Kadigan> kepstin: https://8n1.org/16191/a177 this [14:07:28 CET] <kepstin> hmm. i'm not sure what I expected. I guess that means the file is some sort of raw data with no header. [14:07:51 CET] <kepstin> the filter says it supports ".cube" files, but i dunno the exact format needed. [14:09:10 CET] <kepstin> hmm. a camera lut is probably rgb. can you re-run your ffmpeg command with "-v verbose" so i can see if there's any inserted conversions? [14:10:09 CET] <kepstin> well, video camera lut is probably yuv :) [14:10:47 CET] <kepstin> how big is the lut file? (#bytes) [14:14:09 CET] <kepstin> anyways, it looks like your ".cube" file isn't the Iridas format the filter expects, that message is printed if it doesn't find the expected file headers [14:16:44 CET] <kepstin> it looks like most of the colour cube formats are actually text files, or at least have text headers; if yours is, the actual format can probably be identified by the first line or two. [15:03:33 CET] <BeerLover> What is the use of -re flag??? [15:04:58 CET] <Kadigan> kepstin: the file is indeed a text file, and it does indeed have a header. Begins with "# Truelight Cube v2.0", if that helps. [15:05:41 CET] <Kadigan> Size is exactly 884816 bytes [15:07:09 CET] <Kadigan> Hm... I just realized I have more than one .cube format, in the damn archive. I'm ashamed to say I missed it. Hm... Give me a sec, I'll try to see if I can get any of them to load up. [15:07:14 CET] <kepstin> Hmm. That doesn't look like one of the formats the filter can read. [15:07:42 CET] <Kadigan> (actually more than a sec, since I'm kinda at work... and need to be seen working from time to time ;P) [15:07:44 CET] <kepstin> If you can figure out how the format is laid out, patches welcome I guess. [15:09:24 CET] <kepstin> BeerLover: when reading a local file, it slows it down to approx real time. Useful to livestream with a file as a source [15:11:03 CET] <BeerLover> kepstin I have a use case where I want to transcode mp4 audio files to HLS and store them in a bucket. Then files will be served from there. Do I need this flag while transcoding? It slows the speed. [15:11:22 CET] <BeerLover> And in the man pages it's written that it's for simulation. [15:11:37 CET] <Mavrik> No, when you're transcoding you shouldn't use it. [15:11:49 CET] <Mavrik> It's only when you want the transcoding process to be locked to actual playback speed and not be faster. [15:13:26 CET] <kepstin> An example is if your input is a file and your output is rtmp to twitch [15:15:16 CET] <Mavrik> Yeah, or playing to a remote player or something [15:16:34 CET] <kepstin> Or hls that's pretending to be a live capture, encoding while people watch it [17:28:00 CET] <Chagall> I am trying to build from master, my last build was some time during last month and with the same config and system (just with some packages updated) it fails with libavcodec/qsvenc_vp9.o not found. any idea what broke? [17:30:18 CET] <Chagall> oh, another git pull brought in qsvenc_vp9.c again. [17:30:23 CET] <Chagall> guess I pulled at the wrong time [18:14:34 CET] <nerdnils> https://pastebin.com/JM3PkL9D why does this not work? [18:15:27 CET] <kepstin> nerdnils: for the reason that the error message you copy/pasted very clearly explained. [18:16:50 CET] <kepstin> i mean, i could write it out again here in chat, but i'd just be repeating what the error message already said. [18:18:02 CET] <TheAMM> I don't want to be mean but I am kind of impressed you managed to copy, pastebin and link it here without reading it [18:18:21 CET] <nerdnils> width not divisable by 2. i can see that but i gave no width. is it the video ffmpeg can not work with? [18:18:53 CET] <kepstin> nerdnils: "width not divisible by 2" isn't even an error in the pastebin you linked, that's a completely different issue [18:19:07 CET] <DHE> change -1:100 to -2:100 [18:19:09 CET] <DHE> in your scaler [18:19:18 CET] <DHE> err, :480 [18:19:36 CET] <nerdnils> i read it already gooogled it first hit told me to change the acodec from aac to copy i chhanged the b:v from 5ß00 to 512 to be a power of two neither changed the message [18:19:49 CET] <kepstin> nerdnils: neither of those things are width [18:20:30 CET] <kepstin> width is referring to the number of pixels wide that the image is. when using yuv420p (4:2:0 subsampling), the width of the picture being encoded must be a multiple of two. [18:20:52 CET] <kepstin> DHE has helpfully explained how to fix that, so do that :) [18:21:37 CET] <nerdnils> got it, thank you [18:22:29 CET] <nerdnils> should have read https://trac.ffmpeg.org/wiki/Scaling instead of googling it which proved to be useless. was thinking already what the ac has to do with width& [18:22:32 CET] <nerdnils> thank you guys [19:17:09 CET] <AmyMalik> nerdnils: are you german? [19:56:31 CET] <nerdnils> AmyMalik, yes, why? [19:56:37 CET] <AmyMalik> ß [19:56:45 CET] <AmyMalik> i have to make 3 keystrokes to type that [19:56:53 CET] <AmyMalik> it seems like you only needed to press 1 key [19:57:05 CET] <nerdnils> right [19:57:28 CET] <nerdnils> regardless that i never wrote that sign in here :þ [19:57:46 CET] <AmyMalik> lol [19:59:54 CET] <nerdnils> AmyMalik, qwertz has more keys than qwerty anyways. for example a key for <> and | left of where z is on your keyboard(where those signs are on your keyboard is öäü) [20:00:19 CET] <AmyMalik> nerdnils: aoeuidhtns ;-) [20:00:32 CET] <Diag> its disgusting [20:01:55 CET] <Diag> The linux image on a board that i have came in the UK locale and i couldnt fuggin find # [20:02:01 CET] <Diag> like what the beep yall [20:04:45 CET] <nerdnils> Diag, just reconfigure the locales? nobody needs # for a proper command. but have a qwerty keyboard and german locales and this realy #+?!%& because the key which has | is not even on the keyboard [20:07:12 CET] <Diag> nerdnils: i did change the locale, but the issue was that id killed X and dropped to tty and needed to comment a line in my apt sources [20:07:24 CET] <Diag> I didnt want to delete it [20:07:47 CET] <Diag> So im sitting there hitting all sorts of buttons, couldnt find #, had to stop what i was doing and change the locale [20:09:31 CET] <nerdnils> does apt.source tkat /* */? [20:09:41 CET] <nerdnils> i use arch, i don't know [20:12:07 CET] <Diag> I dont know if it *will* [20:12:26 CET] <Diag> ive only ever seen /etc/apt/sources.list commented with # [20:14:51 CET] <nerdnils> if there are other lines commented out yank it? [20:15:02 CET] <Diag> Nope lol [20:15:10 CET] <nerdnils> copy and paste ftw [20:15:18 CET] <Diag> taint my first rodeo [20:15:25 CET] <Diag> euro keyboard just sucks [20:18:45 CET] <nerdnils> no [20:18:49 CET] <nerdnils> this is rodeo: http://bash.org/?795779 [20:19:15 CET] <Diag> alright hunter2 [20:25:13 CET] <nerdnils> yours was: Diagana Jones: Raiders of the lost key [20:29:27 CET] <Diag> nerdnils: im actually leading the change.org petition to have the rainbow emoji key added to standard keyboards [20:38:09 CET] <nerdnils> what about the snowman? [21:10:09 CET] <SpeackerToMeat> Hello all [21:10:41 CET] <AmyMalik> that's a weird name [21:11:14 CET] <SpeackerToMeat> Question, say I have these dirs: /jpg-1/vid_######.j2c and /jpg-2/vid_######.j2c is there any way to open them all as sequences of images to convert intop a single output video? [21:11:27 CET] <SpeackerToMeat> can I use file template names with file lists for example? [21:11:38 CET] <SpeackerToMeat> AmyMalik: Huh... a c slipped in somehow... [21:11:44 CET] <SpeakerToMeat> Better [21:12:24 CET] <SpeakerToMeat> AmyMalik: thanks [21:12:45 CET] <AmyMalik> even so [21:12:52 CET] <SpeakerToMeat> Diag: Maybe you can enlist the help of Tom Scott [21:12:58 CET] <kepstin> SpeakerToMeat: with most ffmpeg builds, the image demuxer accepts unix shell glob patterns [21:13:12 CET] <SpeakerToMeat> kepstin: So I could use the file list with that [21:13:14 CET] <kepstin> (not sure, this might not work on windows) [21:13:38 CET] <SpeakerToMeat> I'm on linux [21:14:08 CET] <SpeakerToMeat> So, I should use shell globs rather than vid_%06d.j2c [21:14:24 CET] <AmyMalik> if 06d will work then use that [21:14:43 CET] <SpeakerToMeat> With the concat demuxer? [21:15:03 CET] <kepstin> no, not concat demuxer. You're using the image2 demuxer if you have separate image files [21:15:25 CET] <SpeakerToMeat> So how do I concat these directories? [21:16:06 CET] <SpeakerToMeat> With the concat filter? [21:16:23 CET] <kepstin> you could use a glob file pattern that selects both directories, or do each directory as a separate input and then concatenate those inputs (in theory the latter might be doable with the concat demuxer) [21:17:04 CET] <SpeakerToMeat> It seems I could use each directory as input (with the image demuxer) and use the concat filter.... [21:17:13 CET] <kepstin> yep, that works too [21:17:53 CET] <kepstin> (well, as long as you don't want to use -c copy) [21:19:52 CET] <kepstin> but you probably could just do `ffmpeg -pattern_type glob -i /jpg-*/vid_*.j2c` [21:20:10 CET] <kepstin> assuming the directories and files sort in the order you want [21:20:29 CET] <SpeakerToMeat> ok i'll try that [21:25:39 CET] <SpeakerToMeat> The good thing about that is thatr if I have to apply settings to the input images (set framerate and colorspace) I can do so with a single command [21:33:36 CET] <SpeakerToMeat> The way to specify what the native rate for discrete image input was... it wasn't -r I think, it had the word fps... [21:34:53 CET] <kepstin> SpeakerToMeat: the -framerate input option, see "ffmpeg -h demuxer=image2" or the online docs at https://www.ffmpeg.org/ffmpeg-formats.html#image2-1 [21:35:37 CET] <SpeakerToMeat> that was it. thank you kepstin [21:39:01 CET] <SpeakerToMeat> Do we have.... [21:39:05 CET] <SpeakerToMeat> kepstin++ [21:39:09 CET] <SpeakerToMeat> Nope [21:39:52 CET] <SpeakerToMeat> It's good anyhow, I suspect some people here would've ovreflown a karma counter by now [21:40:01 CET] <SpeakerToMeat> overflowed [21:40:23 CET] <pbox> hi, can anyone help me debug an error? I use this command https://termbin.com/m2rp and ffmpeg gives me an error: Option sample_rate not found. I searched online and couldn't find an answer [21:42:02 CET] <pbox> *troubleshoot [21:44:31 CET] <poutine> pbox: what's in $AUDIO_RATE [21:44:40 CET] <pbox> 44100 [21:44:45 CET] <kepstin> pbox: input options go before the input they apply to. you probably intended to have the -ar option between the -f jack and the -i ffmpeg [21:44:46 CET] <poutine> post a command after it is run through this [21:45:22 CET] <kepstin> pbox: alternately if you want -ar as an output option, you have to put it after the last -i option (and before the output filename) [21:46:32 CET] <pbox> first option gave the same error [21:46:44 CET] <pbox> will try to move it before the last -i [21:47:03 CET] <kepstin> well, the position to put it in depends what you want the option to do [21:47:39 CET] <kepstin> that said, yeah, the jack input device doesn't support setting a sample rate [21:48:05 CET] <pbox> maybe that was it [21:48:06 CET] <kepstin> i guess you just get whatever jack gives you :) [21:48:06 CET] <pbox> let me try [21:48:44 CET] <kepstin> if you put it as an output option, ffmpeg will resample the audio to the specified sample rate before encoding to your destination file. [21:49:39 CET] <pbox> now it's playing but I can't see the text [21:49:51 CET] <kepstin> actually, your command has two outputs, so if you want to use -ar as an output option, you need to put it on both outputs [21:49:55 CET] <pbox> btw can I just define the parameters once for both outputs? [21:50:21 CET] <kepstin> no [21:52:30 CET] <pbox> ok. Any idea why the text doesn't show? [21:53:25 CET] <kepstin> does the text show on only on the first output and not the second? :) the -vf option is per-output. [21:53:47 CET] <kepstin> note that if you use -filter_complex, that's a global option which is shared between all inputs and outputs. [21:54:36 CET] <pbox> no it doesn't show in the first either [21:55:40 CET] <kepstin> i don't see anything obviously wrong there. [21:55:46 CET] <pbox> and I think the audio stopped streaming as well [21:56:26 CET] <kepstin> if you use -filter_complex you will have to manually specify -map options to get the audio stream in, i think. [21:56:58 CET] <kepstin> other than that... I've never actually used ffmpeg with jack, i dunno how it behaves [21:57:17 CET] <pbox> and I think the audio stopped streaming as well [21:57:21 CET] <pbox> I can get it to work [21:57:30 CET] <pbox> with a different command [22:03:58 CET] <SpeakerToMeat> kepstin: glob no go: "/usr/bin/ffmpeg: Argument list too long" [22:04:18 CET] <SpeakerToMeat> I guess having a total of almost 200000 files in 8 directories was not fun for ffmpeg [22:04:40 CET] <kepstin> you need to quote it so your shell doesn't expand it [22:04:56 CET] <kepstin> ffmpeg does the glob internally [22:05:35 CET] <SpeakerToMeat> Ah [22:05:56 CET] <SpeakerToMeat> Aha [22:06:37 CET] <pbox> hey thanks for the support, I finally got it to work [22:06:49 CET] <pbox> one last question - I am using a python script to rewrite the file [22:07:15 CET] <pbox> and apparently sometimes ffmpeg tries to read the file when its being written and crashes saying: [22:07:27 CET] <SpeakerToMeat> Ok, ffmpeg glob patterns NEED an extension for file [22:07:38 CET] <pbox> The text file 'path/to/file' could not be read or is empty [22:07:59 CET] <kepstin> pbox: in your python script, write to a temp file then rename to the real filename [22:08:51 CET] <pbox> gotcha :) thanks kepstin you're great [22:09:22 CET] <SpeakerToMeat> Hmmmmm [22:10:51 CET] <SpeakerToMeat> I have /jpg-1/movie-####.j2c up to /jpg-8 and I have /wav-1/sound.wav up to /wav-8 ... yet when I try both, the wav one fails: ffmpeg -pattern_type glob -framerate 24 -i "/jpg-*/*.j2c" -i "/wav-*/*.wav" [22:11:11 CET] <SpeakerToMeat> /wav-*/*.wav: No such file or directory [22:11:14 CET] <SpeakerToMeat> Yet, they exist [22:11:37 CET] <SpeakerToMeat> I tried adding a second -pattern_type glob before the second input, no go [22:11:52 CET] <kepstin> SpeakerToMeat: only the image2 demuxer supports the glob thing [22:11:58 CET] <SpeakerToMeat> ah hmm [22:12:00 CET] <kepstin> for the wav files you'll want to use the concat demuxer [22:12:34 CET] <SpeakerToMeat> so then the image demuxer will see this as a single video, right? so I would need to -i the glob images, and -i each single wav, and then do a concat filter with v=1 and a=8 right? [22:12:44 CET] <SpeakerToMeat> or, the concat demuxer...right [22:12:55 CET] <kepstin> no, use the concat demuxer, then you'll have 1 video and 1 audio stream :) [22:13:09 CET] <SpeakerToMeat> yep [22:13:41 CET] <kepstin> note that the concat filter doesn't work that way - you'd have to use "concat=a=1:v=0:n=8" to concatentate the audio files without touching the video, since the video is already one stream. [22:18:19 CET] <SpeakerToMeat> hmmm ok [22:42:18 CET] <pbox> kepstin: I rename the file and I get the same error [22:43:21 CET] <kepstin> pbox: the sequence should be "open temp file, write to it, close temp file, rename temp file to real file". Or you could use a helper library like python-atomicwrites [22:45:03 CET] <pbox> kepstin: i closed the file and use os.rename [22:47:49 CET] <pbox> if you know python, I do with open("txt1tmp") as file: write - and outside I use os.rename("txt1tmp","txt1") [23:17:48 CET] <pbox> kepstin: I also tried using a shell call to the mv command which writes atomically and I et the same error [23:18:29 CET] <pbox> s/et/get/ [23:49:03 CET] <kingsley> May I please have the benefit of your informed thoughts on why ffmpeg used to accept certain options, but now version 7:4.2.1-1 complains? [23:49:12 CET] <kingsley> The options are [23:49:29 CET] <kingsley> ffmpeg -i /tmp/test.mp4 -r -filter_complex showspectrum=mode=separate:color=rainbow:slide=1:scale=cbrt:size=400x226 -y -acodec copy -qp 0 /tmp/test.mkv [23:49:43 CET] <kingsley> Version 7:4.2.1-1 complains with [23:49:57 CET] <kingsley> [NULL @ 0x8073ae40] Unable to find a suitable output format for 'showspectrum=mode=separate:color=rainbow:slide=1:scale=cbrt:size=400x226' [23:49:57 CET] <kingsley> showspectrum=mode=separate:color=rainbow:slide=1:scale=cbrt:size=400x226: Invalid argument [23:53:42 CET] <c_14> kingsley: -r takes an option [23:53:54 CET] <c_14> so it's consuming the -filter_complex [23:54:27 CET] <c_14> -r <number> [23:56:07 CET] <kingsley> c_14: Thank you very much! I agree, and am happy to report that appending "3-" after "-r" fixed it. I'll focus on finding out what -r wasn't passed a number by the script. Thanks again! [23:56:38 CET] <kingsley> s/"3-"/"30"/ [00:00:00 CET] --- Tue Nov 5 2019

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ffmpeg-devel.log.20191103
by burek 04 Nov '19

04 Nov '19

[01:30:07 CET] <Compn> hah what is this spam compile your own trojan? <filternet> Hi dudes, i brought you C madness: cOut() in C alpha test version https://pastebin.com/miw35a1r [09:46:42 CET] <cone-552> ffmpeg 03Linjie Fu 07master:e786e37326d4: lavc/qsvenc: Fix bitrate_limit to allow AVC encode in limited bitrate [09:46:42 CET] <cone-552> ffmpeg 03Zhong Li 07master:33583803e107: lavc/qsvenc: enable vp9 encoder [11:54:31 CET] <cone-552> ffmpeg 03Andriy Gelman 07master:5e3229df4ca8: avformat: Add max_probe_packets option [12:38:56 CET] <cone-552> ffmpeg 03Nicolas Frattaroli 07master:a8ec0685ac1c: avformat/ftp: add AVOptions for authentication [12:56:09 CET] <arpu> Hello can someone help to implement rtmp read/connect timeout at https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/rtmpproto.c#L2655 [12:57:03 CET] <arpu> i think something like this should work [12:57:13 CET] <arpu> ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, [12:57:13 CET] <arpu> "?timeout=%d", [12:57:14 CET] <arpu> rt->listen_timeout * 1000); [13:09:16 CET] <arpu> @michaelni, makes this sense to do? ^^ [00:00:00 CET] --- Mon Nov 4 2019

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ffmpeg.log.20191103
by burek 04 Nov '19

04 Nov '19

[02:04:32 CET] <DHE> API question: functions that begin with ff_* are not intended for user consumption. correct? [03:05:38 CET] <DHE> I ask because I see useful stuff in there... :/ [03:07:55 CET] <c_14> >For variables and functions visible outside of file scope, but only used internally by a library, an @code{ff_} prefix should be used, e.g. @samp{ff_w64_demuxer}. [03:08:05 CET] <c_14> according to doc/developer.texi anyways [03:10:08 CET] <DHE> well that takes all the fun out of it... [03:43:37 CET] <hendry> I have a record script that has just started failing after an upgrade/reboot: No handle set on framebuffer: maybe you need some additional capabilities? [03:43:52 CET] <hendry> here is my log file https://s.natalian.org/2019-11-03/1572748901.mp4.log and a google only shows up https://www.reddit.com/r/ffmpeg/comments/cr8jd2/how_to_use_kmsgrab/ [09:37:58 CET] <Carltoso> morning [09:38:01 CET] <Carltoso> :-)) [10:45:20 CET] <ponyrider> snooky-: your bash script is incredibly broken [10:55:59 CET] <ponyrider> 1. never use ls to get a list of files, 2. a directory in the "$dir" folder will break your script 3. sh extensions are for sh files, not bash files [10:56:43 CET] <ponyrider> snooky-: i assume you want something like this? http://ix.io/20G7 [10:58:34 CET] <furq> probably not because that ffmpeg command is still completely broken [10:58:42 CET] <Reinhilde> lal [10:59:42 CET] <ponyrider> LOL [11:05:35 CET] <ponyrider> 4. its also an infinite loop [11:06:33 CET] <Reinhilde> "sh extensions are [...not for] bash files" [11:06:54 CET] <Reinhilde> you obviously still live in windows land [11:07:27 CET] <ponyrider> no - he's calling #!/usr/bin/bash not #!/usr/bin/sh [11:07:47 CET] <Reinhilde> and still, you live in windows land [11:07:57 CET] <Reinhilde> the extension being different does not break the script [11:08:07 CET] <Reinhilde> it is merely cosmetic. you can save it as .bash on your own system if you want [11:08:40 CET] <ponyrider> sh and bash are not the same things, its misleading. if there was any extentions it should be file.bash [11:08:55 CET] <Reinhilde> true and irrelevant. [11:09:32 CET] <Reinhilde> .sh is used as an extension many times for any shell script. [11:10:02 CET] <ponyrider> for shell files [11:10:30 CET] <Reinhilde> ponyrider: how long have you been using unix? [11:10:50 CET] <ponyrider> r u serious? -.- [11:11:32 CET] <Reinhilde> file extensions are informative only and not intended to be a be all end all [11:16:16 CET] <ponyrider> Reinhilde: how about no extention? either way, i agree its a moot point considering his code is broken anyway [11:17:46 CET] <Reinhilde> then why did you bring it up if it's moot [11:18:07 CET] <ponyrider> because i am still right [11:18:34 CET] <furq> if we're really doing this then it should be #/usr/bin/env bash [11:19:00 CET] <furq> or just /bin/sh and find "$dir" -maxdepth 1 -type f | shuf -n 4 | xargs -n1 -P4 ffmpeg ... [11:22:37 CET] <ponyrider> 2019-10-30 23:39:17 snooky hi all [11:22:44 CET] <ponyrider> ^ i dont think he's listening either way [11:49:50 CET] <Carltoso> Hi [11:49:57 CET] <Carltoso> Does anyone know why I get this error [11:50:09 CET] <Carltoso> Unable to find a suitable output format for ' [11:50:26 CET] <Carltoso> my command looks like this ffmpeg -i /Users/Malthe/test//1.mkv -i /Users/Malthe/test//2.mkv -i /Users/Malthe/test//3.mkv -i /Users/Malthe/test//4.mkv -i /Users/Malthe/test//5.mkv -i /Users/Malthe/test//8.mkv -i /Users/Malthe/test//9.mkv -filter_complex [0:a][1:a][2:a][3:a]amix=inputs=4 " \ [11:50:26 CET] <Carltoso> [0:v] setpts=PTS-STARTPTS, scale=qvga [a0]; \ [11:50:26 CET] <Carltoso> [1:v] setpts=PTS-STARTPTS, scale=qvga [a1]; \ [11:50:26 CET] <Carltoso> [2:v] setpts=PTS-STARTPTS, scale=qvga [a2]; \ [11:50:27 CET] <Carltoso> [3:v] setpts=PTS-STARTPTS, scale=qvga [a3]; \ [11:50:29 CET] <Carltoso> [4:v] setpts=PTS-STARTPTS, scale=qvga [a4]; \ [11:50:31 CET] <Carltoso> [5:v] setpts=PTS-STARTPTS, scale=qvga [a5]; \ [11:50:33 CET] <Carltoso> [6:v] setpts=PTS-STARTPTS, scale=qvga [a6]; \ [11:50:35 CET] <Carltoso> [a0][a1][a2][a3][a4][a5][a6]xstack=inputs=7:layout=0_0|w0_0|w0+w1_0|0_h0|w0_h0|w0+w1_h0|0_h0+h1[out] \ [11:50:38 CET] <Carltoso> " -map "[out]" -c:v crf 23 -preset medium -movflags +faststart -c:a aac libx264 - | ffplay -autoexit -left 10 -top 10 - [11:50:42 CET] <Carltoso> works fine without [0:a][1:a][2:a][3:a]amix=inputs=4 [11:51:15 CET] <Carltoso> ah I have 9 inputs I will try to update [11:52:18 CET] <Carltoso> Sorry for the spam [11:52:24 CET] <cehoyos> "-" (the pipe) has no default output format and you do force one with the option "-f" [11:52:24 CET] <Carltoso> Now i get this error "[AVFilterGraph @ 0x7fd31fe0b5c0] Too many inputs specified for the "amix" filter. [11:52:24 CET] <Carltoso> Error initializing complex filters. [11:52:24 CET] <Carltoso> Invalid argument [11:52:24 CET] <Carltoso> pipe:: Invalid data found when processing input" [11:52:43 CET] <cehoyos> I suspect you want "-f h264" [11:52:49 CET] <Carltoso> yes [11:52:59 CET] <cehoyos> And I see now that "libx264" is missing "-c:v" in front of it [11:53:25 CET] <cehoyos> But you don't have to specify it if you use "-f h264" because iirc, that defaults to "v:c libx264" [11:53:42 CET] <furq> amix has no output label or delimiter [11:54:03 CET] <furq> also it's outside the quotes [11:54:15 CET] <Carltoso> where do i put -f h264 ? hehe [11:54:24 CET] <furq> also please use a pastebin [11:55:03 CET] <Carltoso> yes sure absolutely [11:55:19 CET] <Carltoso> It tells amix has to many inputs [12:24:53 CET] <Carltoso> I have my video somewhat working now https://pastebin.com/sKbthdib [12:25:06 CET] <Carltoso> now I just need to add audio [12:25:17 CET] <Carltoso> does anyone know how? [13:57:53 CET] <jemius> Is a 3D denoiser what it sounds like, for 3D Videos? [13:58:25 CET] <furq> no [13:58:28 CET] <furq> it means it does temporal denoising [13:58:35 CET] <furq> in addition to spatial [13:59:02 CET] <jemius> ahm... is that good? [13:59:23 CET] <furq> sure [14:01:05 CET] <jemius> not that easy to use, though. All default paremeters are 0. Defaultvalues for atadenoise worked quite well for me [14:01:35 CET] <furq> the defaults for hqdn3d aren't all 0 [14:01:58 CET] <jemius> sure: ffmpeg -h filter=hqdn3d [14:01:59 CET] <jemius> luma_spatial <double> ..FV.... spatial luma strength (from 0 to DBL_MAX) (default 0) [14:02:02 CET] <jemius> etc [14:02:33 CET] <furq> https://ffmpeg.org/ffmpeg-filters.html#hqdn3d-1 [14:02:36 CET] <furq> not sure what's going on there [14:03:27 CET] <furq> https://github.com/FFmpeg/FFmpeg/blob/master/libavfilter/vf_hqdn3d.c#L194-L… [14:03:27 CET] <furq> oh [14:03:31 CET] <jemius> probably because my version is from the middle ages [14:03:38 CET] <furq> no it's still there [14:04:08 CET] <jemius> who is where? [14:04:17 CET] <furq> the defaults are still 0 in master [14:04:35 CET] <furq> i guess it's done this way because some default params are factors of other params [14:04:47 CET] <jemius> but what you linked is 4.0, 3.0 and 6.0 [14:04:57 CET] <furq> yeah and those are the actual defaults [14:05:09 CET] <furq> they're set to that in the actual filter code [14:05:27 CET] <furq> they're just not the defaults in the AVFilter which is what -h shows [14:05:51 CET] <jemius> hmm [14:07:10 CET] <jemius> furq, anyways, would you say this filter is the "best" for denoising? I have 7 denoising filters and don't know how they compare [14:07:20 CET] <furq> there isn't a "best" [14:07:29 CET] <furq> hqdn3d does pretty well with most stuff [14:08:18 CET] <furq> nlmeans is also decent and there's an opencl version [14:12:45 CET] <jemius> do denoising filters also reduce flickering? VLC offers options for both denoise and flickering [14:13:38 CET] <furq> a temporal one might reduce it a bit [14:13:56 CET] <furq> there's at least two dedicated deflicker filters though [14:14:09 CET] <furq> deflicker and normalize [14:14:23 CET] <jemius> i really need the newest version... [14:14:32 CET] <jemius> #debianlife [14:14:33 CET] <furq> https://www.johnvansickle.com/ffmpeg/ [14:16:45 CET] <jemius> thx [14:17:31 CET] <jemius> I presume filters don't negatively interact with each other? deinterlace, denoise, deflicker altogether [14:18:14 CET] <furq> the order matters [14:18:23 CET] <furq> that order looks right though [14:19:42 CET] <jemius> I actually had a lecture about image processing in Uni once. Still a whole science for itself [14:46:43 CET] <jemius> Whenever I use filters I can not use -c copy so I have to choose the same codec as before manually to prevent ffmpeg from using the default ones? [15:22:51 CET] <DHE> jemius: you can't both modify the raw frame and copy without decoding/encoding [15:23:43 CET] <jemius> DHE, so you can also never use 'copy' when cutting? [15:38:04 CET] <klaxa> jemius: not if you are using filters [15:38:51 CET] <jemius> klaxa, no no. I mean: If you cut a video in half it will likeley have to re-encode again, right? As frames depend on each other in lossy compression algorithms [15:39:16 CET] <klaxa> if you cut at keyframes you can use -c copy and prevent re-encoding [15:40:35 CET] <DHE> but if your keyframes are not where you want them... well, too bad... [15:41:23 CET] <jemius> doesn't matter, they're at most a few seconds distance to each other, right? [15:56:19 CET] <DHE> depends. h264 runs quite well without keyframes, so if they're not needed then there's bitrate savings to be had by minimizing their use [16:02:34 CET] <jemius> I didn't even know these codecs work without key frames [16:08:40 CET] <DHE> they don't. they must decode a keyframe first. this is why on digital TV boxes channel change times are inconsistent - they're waiting for the next keyframe [16:09:02 CET] <DHE> (there is a kinda cool feature in h.264 called intra refresh that eliminates keyframes, but that's the exception to the rule) [16:09:57 CET] <jemius> DHE, I guess the goal without key frames would be that you don't loose a whole segment if the kf is destroyed? [16:10:49 CET] <DHE> keyframes are large because they must describe a whole frame from nothing. furthermore H264 supports motion reference frames from more than 1 frame back but a keyframe is a wall where back references cannot go past. [16:11:26 CET] <DHE> but unlike older codecs like MPEG-4 part 2 (aka DivX), the image quality doesn't deteriorate over time if you don't have regular keyframes. [16:11:36 CET] <DHE> so from a "minimize size" standpoint, keyframes are the enemy [16:13:32 CET] <jemius> my question was if when a keyframe is broken displaying the video has to pause until the next kf arrives? [16:14:25 CET] <DHE> usually the player makes a best effort to decode it anyway. you just get graphical glitching and errors which will persist and smudge until the next keyframe [16:14:39 CET] <DHE> do a google image search for "macroblocking" [16:15:10 CET] <jemius> k, would have been my next question; because you only ever see glitches, never the image going disappearing fully [16:21:23 CET] <jemius> One really would need one of those thread ripper CPUs for testing [16:24:05 CET] <DHE> I have some Xeon Gold servers with 20 cores in each socket. :) [16:25:26 CET] <jemius> servers are only useful to fiber-people who can push >1GB around in no time :0 [16:26:30 CET] <jemius> Am I right with the assumption that ffmpeg's filterset targets mostly at editing videos? Because I noticed that the audio filters lack most natbly the notch filter [16:27:30 CET] <DHE> there's a decent number of audio filters. dunno which one that is but maybe an existing filter can meet your needs [16:27:38 CET] <DHE> https://ffmpeg.org/ffmpeg-filters.html [16:28:39 CET] <jemius> nope, the notch filter is crucial for removing single frequency disturbances. It's a very common filter, and a very useful one. It's not a big problem, you can filter it externally. I was just wondering [17:09:59 CET] <jemius> DHE, btw I just remember that I currently have access to a clusetr with 6000 nodes :) :) [17:09:59 CET] <durandal_1707> jemius: see bandreject it is notch filter [17:10:49 CET] <DHE> jemius: well you win then [17:11:11 CET] <DHE> I just assume people have only core i7 6xxx series (or worse) unless told otherwise [17:12:22 CET] <jemius> durandal_1707, I'm not sure if you can configure a typical bandreject to be as steep as a notch [17:18:14 CET] <jemius> ffmpeg -ss 00:10:00 -i foo.mkv -to 00:20:00 bar.mkv [17:18:22 CET] <jemius> does this do what I think it will do? Because it does not [17:20:45 CET] <klaxa> move -to before -i foo.mkv i think [17:24:17 CET] <DHE> yeah, you're doing a seek on the input before starting, and setting an end time on the output. [17:26:05 CET] <jemius> https://superuser.com/a/704118 [17:26:30 CET] <jemius> a bit confusing [17:26:48 CET] <klaxa> i think that's outdated [17:26:56 CET] <klaxa> https://trac.ffmpeg.org/wiki/Seeking [17:27:18 CET] <klaxa> hmm wait... wtf i thought that was fixed [17:28:38 CET] <jemius> In Dary, no one is ever really gone [17:50:52 CET] <jemius> furq, denoising with 3D worked really well with default settings, thx for the recommendation. One is tempted to further play with the settings, but I expect that with stronger presets it would only blurr to mutch [18:16:28 CET] <snooky> hi all [18:16:33 CET] <snooky> i create an stream with ffmpeg [18:16:41 CET] <snooky> this runs fine but with an error [18:17:15 CET] <snooky> I do not know how to explain it. can I post the stream url? [18:17:49 CET] <jemius> snooky, I'd say post what your command, stream and error is [18:20:38 CET] <snooky> https://nopaste.linux-dev.org/?1270963 [18:20:44 CET] <snooky> https://nopaste.linux-dev.org/?1270964 [18:21:20 CET] <snooky> http://i.imgur.com/ElP3DM0.png [18:21:48 CET] <snooky> http://i.imgur.com/2lDb36p.png [18:23:00 CET] <snooky> http://i.imgur.com/IsXgcMg.png [18:23:34 CET] <snooky> the upper or the whole picture becomes totally funny. with such strange effects. [18:25:04 CET] <cehoyos> Please post the actual command line (without variables) and the complete, uncut console output on the paste (Poste die tatsächliche Kommandozeile, die Du getestet hast, ohne Variablen, zusammen mit dem gesamten, unveränderten Output auf der Konsole) [18:40:01 CET] <snooky> I've already found a "mistake". I believe. so as long as the input material is 1920x1080 everything is perfect and exactly as it should .. the input material but now z.b. 1920x720 then he makes these problems. [19:20:45 CET] <entourage> how do I adjust the volume when playing media with ffplay? [19:21:10 CET] <entourage> is there a volume control built in to that app? [19:32:41 CET] <cehoyos> No [19:32:59 CET] <cehoyos> (It is a mostly a testing tool, there are media players based on FFmpeg) [19:39:00 CET] <entourage> too bad since ffplay is a very capable media player and I like it a lot [19:45:56 CET] <jemius> does VLC use ffmpeg? [19:55:20 CET] <cehoyos> jemius: Yes [20:29:13 CET] <DHE> yeah, ffplay is functionally capable but treat it as a diagnostic tool. I like to use to to test ffmpeg filters before I actually encode the video... [22:13:54 CET] <void09> anyone tried to use tesseract or some other open source ocr to extract hardsubs from video ? [00:00:00 CET] --- Mon Nov 4 2019

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ffmpeg-devel.log.20191102
by burek 03 Nov '19

03 Nov '19

[15:17:23 CET] <taliho> anyone able to reproduce https://trac.ffmpeg.org/ticket/8285 ? [15:19:00 CET] <taliho> tried to set v4l2_m2m on encoding/decoding on raspi 4, but don't have any memleaks when testing on the sample [15:40:52 CET] <mkver> All I know about said ticket is that there is already an attempted fix on the ML: https://ffmpeg.org/pipermail/ffmpeg-devel/2019-October/251940.html [15:43:05 CET] <taliho> :mkver I was trying to test the fix [17:28:52 CET] <taliho> ok, I couldn't reproduce because I had a valid v4l2_m2m device on the raspi. Can reproduce on a different machine [21:36:47 CET] <tmm1> anyone working on AC4 decoder? [21:37:15 CET] <JEEB> i think durandal_1707 was but he got fed up doing it for free [21:37:23 CET] <JEEB> or something [21:37:30 CET] <JEEB> also apparently it wasslow [21:38:05 CET] <JEEB> he might have something wip in vis girhub [21:38:18 CET] <Lynne> durandal_1707 but couldn't continue without samples which used more features [21:38:33 CET] <JEEB> right [21:39:02 CET] <JEEB> i only knew the ones on dash if sample repo [22:02:35 CET] <jamrial> someone on #videolan posted a few dash broadcast samples (ac4 in isobmff) the other day [22:15:25 CET] <durandal_1707> few? [22:18:33 CET] <tmm1> i have some samples from astc3 ota and also someone interested in sponsoring dev [22:20:45 CET] <durandal_1707> i need full ac4 coverage, not some sloppy samples [22:21:05 CET] <durandal_1707> ac4 is more complex than ac3 [22:21:43 CET] <durandal_1707> with different versions and channel layouts [22:29:59 CET] <jamrial> durandal_1707: short of a test suite, you're unlikely to get full coverage for such a format. even if every broadcaster switches to it [22:34:03 CET] <durandal_1707> tmm1: i already wrote functional decoder, so i dunno what you want [22:35:11 CET] <Lynne> just call avpriv_report_missing_feature() to nag users for samples [22:41:34 CET] <tmm1> durandal_1707: have a link? [22:44:58 CET] <durandal_1707> tmm1: why you need a link? its public [23:07:41 CET] <tmm1> ok i will search [23:10:15 CET] <tmm1> https://github.com/FFmpeg/FFmpeg/compare/master...richardpl:ac4 [23:35:01 CET] <JEEB> meanwhile Japan seems to have picked MPEG-H 3-D audio [23:35:47 CET] <JEEB> which is probably why sony posted a demux patch for it [00:00:00 CET] --- Sun Nov 3 2019

1 0

ffmpeg.log.20191102
by burek 03 Nov '19

03 Nov '19

[00:00:59 CET] <phobosoph> hi [00:01:04 CET] <phobosoph> "Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080, 30 fps, 30 tbr, 90k tbn, 60 tbc" [00:01:08 CET] <phobosoph> looks nice + youtube live friendly [00:01:12 CET] <phobosoph> except the yuvj420p thing [00:01:30 CET] <phobosoph> I could use a raspberry (with good cooling) and hardware accelerated ffmpeg to re-encode to other colour format [00:01:38 CET] <phobosoph> hm, or does youtube live do this already for all its users? [00:09:33 CET] <nicolas17> youtube video upload will reencode whatever you send even if it's already compatible [00:09:39 CET] <nicolas17> not sure about live [00:17:35 CET] <phobosoph> nicolas17: pro question: is it possible to ffprobe the youtube live stream, too? or another way for finding out whether it is yuvj420p? [00:17:38 CET] <phobosoph> :) [00:17:40 CET] <phobosoph> that would be so cool [00:18:07 CET] <nicolas17> youtube-dl --get-url https://youtube.com/... [00:18:12 CET] <nicolas17> then pass the result to ffprobe [00:18:21 CET] <nicolas17> I think I never tried it on live ... might work [00:18:34 CET] <phobosoph> oh cool [00:18:40 CET] <phobosoph> pass result to ffprobe, so the downloaded file [00:18:40 CET] <phobosoph> ok [00:19:09 CET] <nicolas17> oh with --get-url it gives you the video URL instead of downloading anything [00:19:54 CET] <nicolas17> I guess downloading and passing the file to ffprobe would work too ^^ [00:25:10 CET] <phobosoph> nicolas17: alright, installing youtube-dl from source (master) because apparently older versions have issue with latest youtube [00:25:20 CET] <furq> pretty sure live won't work with yuvj [00:25:29 CET] <furq> there's an easy way to find out though [00:27:26 CET] <phobosoph> furq: how? the stream runs currently, but I don't know what yuvj thing it got [00:29:40 CET] <phobosoph> furq: is there another way except using youtube-dl? [00:30:12 CET] <nicolas17> stream yuvj live and see if it works on the browser? :) [00:30:35 CET] <furq> yeah that [00:35:21 CET] <phobosoph> it works nicely in chrome and on my smartphone [00:36:54 CET] <phobosoph> so youtube-dl can download that stream thing [00:36:55 CET] <phobosoph> hm [00:37:07 CET] <phobosoph> but ffprobe is unhappy because youtube-dl can't finish because well, the stream doesn't finish [00:37:35 CET] <phobosoph> ah , I found it [00:37:50 CET] <phobosoph> Stream #0:1: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 30 fps, 30 tbr, 90k tbn, 60 tbc [00:37:56 CET] <phobosoph> hurra [00:38:03 CET] <phobosoph> youtube live converts it to yuv420p!!! [00:38:18 CET] <phobosoph> no need for re-encoding it (though I plan to do this with hardware acceleration on my raspberry, just for being able to) [00:38:20 CET] <phobosoph> good [00:38:35 CET] <furq> the hwenc on the rpi is really bad [00:38:45 CET] <furq> so you probably shouldn't do that [00:39:51 CET] <phobosoph> furq: it isn't necessary :D [00:39:51 CET] <furq> at least the 0-3 quality is terrible, idk what the 4 is like [00:40:08 CET] <phobosoph> furq: 0-3 quality from the raspy hw encoder? [00:40:18 CET] <Banana51> I am looking at this class in python that is using ffmpeg with a source file specified (https://github.com/imayhaveborkedit/discord.py/blob/voice-recv-mk2/discord/…) - How can I use ffmpeg to fetch audio from system microphone? I have tried with "ffmpeg -f alsa -i hw:0 -ar 48000 -ac 2 -loglevel warning", but it wants me to specify output file. However, I just want it to be streamed [00:40:20 CET] <Banana51> like an input file (see link) [00:40:27 CET] <furq> the 4 has a new hwenc block which does hevc [00:40:37 CET] <phobosoph> raspberry 4? [00:40:39 CET] <furq> yeah [00:40:41 CET] <phobosoph> ah [00:40:44 CET] <furq> no idea if the h264 quality got any better [00:40:47 CET] <furq> it can't have got any worse [00:41:10 CET] <phobosoph> mine is a raspby 3 :/ well [00:41:23 CET] <phobosoph> currently I use a vps without traffic issues for now [00:41:32 CET] <phobosoph> but a raspby could remove the need for it [00:42:29 CET] <phobosoph> hm, nvidia jetson (120 bucks) could do it, too [00:42:34 CET] <phobosoph> well, if it supports it [00:42:37 CET] <phobosoph> but that would be awesomely fast [00:43:35 CET] <JEEB> IIRC nvidia decided with that stuff that using standard APIs with standard APIs is not fun [00:43:50 CET] <JEEB> so it has v4l2 hwdec for stuff, but you have to utilize some custom nvidia lib in the middle? [00:44:02 CET] <JEEB> which would be less bad if it was open source. it isn't [00:45:02 CET] <Banana51> Seems like I have to use pipe:1 [00:45:18 CET] <JEEB> I think someone was interested in implementing the jetson stuff on -devel, but I had to stop his description of it after I heard "linking against nvidia lib" [04:22:50 CET] <Adcock> What ffmpeg command will show me what hardware accelerated capabilities I have on my computer? [05:19:25 CET] <TheDcoder> Hi, is it possible to process multiple files with ffprobe? [05:20:01 CET] <TheDcoder> I tried passing multiple files in the command but I got an error about another file already being specified [05:21:11 CET] <TheDcoder> Currently I am using a script which uses ffprobe to process a few hundred video files in a loop, this process is very slow and it is just supposed to be get the dimensions of each video [05:21:56 CET] <TheDcoder> So I was thinking I can speed this up by passing all of those files to ffprobe in a single command and get the output of all files at once [05:25:53 CET] <furq> you can't and it wouldn't speed it up anyway [05:25:58 CET] <TheDcoder> Actually it took me about 20 minutes to process ~700 files (mp4, h256) [05:26:10 CET] <TheDcoder> furq: oh... why is that? :( [05:26:34 CET] <TheDcoder> I imagine just getting the dimensions of a video is a simple enough operation? [05:26:39 CET] <furq> the bottleneck is disk speed [05:27:00 CET] <furq> you could try running it with xargs -P or something but i doubt it'd be significantly faster [05:27:01 CET] <TheDcoder> oh, does ffprobe read all of the file to determine the dimensions? [05:27:11 CET] <furq> no but it doesn't actually decode anything [05:27:20 CET] <furq> it just reads the first 5MB or so of the file [05:27:45 CET] <furq> you could try reducing -probesize [05:27:48 CET] <TheDcoder> I see... but isn't it sufficient enough to read the metadata alone to determine the video dimensions? [05:28:20 CET] <furq> i don't think ffprobe is optimised to only read as much as necessary to get the entries you selected [05:28:34 CET] <furq> afaik it just reads 5MB, gets all the properties and then displays the ones you asked for [05:28:55 CET] <TheDcoder> hmmm... [05:29:00 CET] <furq> but all those properties are stored in headers so it's not doing anything cpu intensive [05:29:23 CET] <TheDcoder> furq: I cannot find the -probesize option here: https://ffmpeg.org/ffprobe.html [05:29:47 CET] <furq> it's a libavformat option [05:29:57 CET] <furq> it works for all the ff* tools [05:30:14 CET] <furq> https://www.ffmpeg.org/ffmpeg-formats.html#Format-Options [05:30:59 CET] <furq> -probesize 1M or something [05:31:14 CET] <TheDcoder> ah, I see [05:31:27 CET] <TheDcoder> would -probesize 1024 suffice? [05:31:35 CET] <furq> probably not [05:31:37 CET] <furq> depends on the file [05:31:40 CET] <furq> hence it being configurable [05:31:48 CET] <TheDcoder> okay, will try it, thanks [05:32:53 CET] <TheDcoder> I have another question, is it possible for several input videos with constant frame rate to output a video with variable frame rate when I am using the copy option? [05:34:07 CET] <furq> maybe [05:34:19 CET] <furq> the input files need the same timebase but afaik they don't need to be the same framerate [05:34:25 CET] <furq> you'll need -vsync vfr anyway [05:38:18 CET] <TheDcoder> what if the input have the same fps with constant frame rate? [05:38:24 CET] <TheDcoder> *input files [05:38:49 CET] <TheDcoder> Is it possible to merge them into a single file with the same constant fps in that case? [05:40:08 CET] <furq> if they have the same timebase, sure [05:41:02 CET] <TheDcoder> I am not sure that I really understand what "timebase" means... [05:41:36 CET] <TheDcoder> but the input files are from the same source, so I imagine there is a pretty good chance that they have the same timebase [05:41:48 CET] <furq> yeah they will do [05:42:18 CET] <TheDcoder> assuming that this is true, do I need to use any special options to specify constant frame-rate output? [05:42:21 CET] <furq> no [05:43:26 CET] <TheDcoder> so if the output has vfr, that means some of the videos don't have the same timebase, am I right? [05:43:46 CET] <furq> it won't be vfr unless you set -vsync vfr [05:44:01 CET] <furq> and they have the same timebase or else the concat demuxer will throw an error [05:44:36 CET] <TheDcoder> very strange, I am not using -vsync vfr [05:44:49 CET] <TheDcoder> but MediaInfo says the output has VFR [05:45:18 CET] <TheDcoder> and the concat demuxer is not throwing any errors as far as I know [05:45:22 CET] <furq> is this a 23.97/29.97fps mkv [05:45:32 CET] <furq> mediainfo always reports those as vfr unless you muxed them with merge [05:45:33 CET] <furq> mkvmerge [05:45:58 CET] <furq> or 59.94 or whatever [05:46:24 CET] <TheDcoder> it is not a mkv [05:46:29 CET] <TheDcoder> the output is an mpv [05:46:31 CET] <TheDcoder> *mp4 [05:46:57 CET] <furq> actually nvm i forgot -vsync auto is the default [05:47:02 CET] <TheDcoder> ... [05:47:06 CET] <TheDcoder> :) [05:47:20 CET] <TheDcoder> so, what should I set -vsync to turn it off? : [05:47:22 CET] <TheDcoder> :) [05:47:39 CET] <furq> try -vsync cfr but i'm not sure that'll work with copy [05:47:46 CET] <furq> since it'd have to drop frames, which it can't do [05:48:02 CET] <furq> or -vsync drop [05:48:13 CET] <TheDcoder> how about -vsync passthrough [05:48:36 CET] <furq> yeah maybe [05:48:42 CET] <TheDcoder> sounds good [05:48:58 CET] <furq> i assume the concat demuxer corrects the timestamps [05:49:05 CET] <furq> since that obviously won't work otherwise [05:49:54 CET] <TheDcoder> as long as it doesn't drop any frames or make the output vfr, that is not a problem [05:50:15 CET] <furq> copying will never drop frames [05:50:25 CET] <TheDcoder> got it [05:50:28 CET] <TheDcoder> thanks [05:50:29 CET] <furq> you can't just drop frames out of most video codecs without breaking it [05:50:41 CET] <TheDcoder> makes sense [05:50:59 CET] <TheDcoder> by the way, any idea what the -safe option is for? [05:51:37 CET] <furq> https://www.ffmpeg.org/ffmpeg-formats.html#Options [05:51:40 CET] <TheDcoder> I can't find it anywhere in the documentation [05:51:49 CET] <TheDcoder> oops [05:51:50 CET] <furq> it's specific to the concat demuxer [05:52:09 CET] <furq> if the concat list has 'ffconcat version 1.0' at the top then it auto-enables it [05:52:35 CET] <TheDcoder> thanks, was just curious [05:52:37 CET] <furq> or disables it, rather [05:52:47 CET] <TheDcoder> I understand [05:56:24 CET] <TheDcoder> -vsync passthrough seems to have done the trick [05:56:46 CET] <TheDcoder> I tested with a small sample and it shows constant frame rate in MediaInfo... [06:00:22 CET] <TheDcoder> Now only if there was a faster way to get the dimensions of the video... -probesize 1024 works actually [06:00:36 CET] <TheDcoder> so that should in theory ease the disk speed bottleneck [06:01:18 CET] <TheDcoder> but I think the real bottle neck is the overhead of starting a new process for every video [06:01:41 CET] <TheDcoder> and not to mention ffmpeg will perform some common initialization actions every time [13:13:22 CET] <snooky> hi all [14:03:22 CET] <pagios> -i plyalist.m3u8 -live_start_index -3 -t 10 -acodec copy -vcodec copy out.mp4 <<-- is this how i convert from hls to mp4? it is not working as expected. I am not seeing any video playing on mp4 [14:15:50 CET] <pagios> JEEB, hi [15:14:01 CET] <pagios> any idea how to extract 1 frame image from a livestream? [16:13:33 CET] <snooky> Is there a way to calculate the required power for ffmpeg? [16:13:47 CET] <JEEB> not really [16:14:13 CET] <JEEB> depends on the speed you require, what modules you utilize and the drivers even [16:16:52 CET] <snooky> So I wrote a little script. and depending on the input material ffmpeg breaks down. or it hacks everything. or a huge delay [16:52:17 CET] <snooky> https://imgur.com/a/nLJMaD3 [16:53:32 CET] <snooky> does ffmpeg get it? so live. no 2 pass or similar. directly, live. [17:13:08 CET] <ddubya> I'm using nv-codec-headers git master and the version supplied is not supported anymore. I'll look elsewhere for the header I guess [17:16:09 CET] <ddubya> nvm, I may have another issue [17:21:01 CET] <ddubya> multiple versions installed [18:08:22 CET] <pagios> i am transcoding a source RTMP stream in HD (1280x720) to 480p and 320p and i used -vf scale=.. -vb.. params, are those enough or idealy i should set a preset too? [18:15:07 CET] <DHE> if you're using a non-standard preset, you will want to set that on all outputs [18:15:14 CET] <DHE> *non-default [18:44:18 CET] <pagios> DHE, thanks. any other parametersthat would ease the cpu and optimze stuff/ [19:02:43 CET] <analogical> what is wrong with this command? "ffmpeg -i audio.wav -b 320 audio.mp3" ?? [19:03:45 CET] <furq> 320k [19:04:14 CET] <pagios> thats 320kbps not kB [19:06:53 CET] <analogical> when I type "ffmpeg -i audio.wav -b 320k audio.mp3" it still creates a file that is 128kbps :( [19:07:34 CET] <furq> did you try reading the warning it throws [19:09:15 CET] <analogical> ffmpeg is an extraordinary tool but it has the worst syntax of any program I've ever used [19:10:28 CET] <Reinhilde> ffmpeg -i input.wav -codec:a libmp3lame -b:a 320k output.mp3 [19:10:41 CET] <furq> how do you suggest the syntax should be improved [19:20:10 CET] <klaxa> when compared with most other programs with long command line arguments it seems rather logical and sane [19:20:38 CET] <klaxa> then again, i don't use many cli programs with as many arguments as ffmpeg [19:29:38 CET] <another> analogical: -b:a 320k [19:30:39 CET] <analogical> another, :D [19:30:55 CET] <another> there is even a warning, that informs you of -b beeing ambiguous [19:33:59 CET] <analogical> what a stupid program hehe [19:37:37 CET] <analogical> the problem with https://www.ffmpeg.org/ffmpeg-codecs.html is that there are too few examples [19:37:48 CET] <analogical> examples makes everything MUCH easier [19:43:11 CET] <analogical> for example https://www.ffmpeg.org/ffmpeg-codecs.html#Options-7 doesn't tell what syntax to use when chosing compression level [19:45:07 CET] <analogical> like I said before ffmpeg is an extraordinary tool but obviously the developers don't want it to be easy to use [19:50:44 CET] <another> i don't believe that to be true [19:51:38 CET] <another> what syntax help are you missing? there's the option and the allowed values it can have [19:56:35 CET] <klaxa> i think there's plenty of examples on the internet, most stuff has examples, everything is documented [19:57:02 CET] <klaxa> once you "grok" it, it's very intuitive [19:57:27 CET] <another> hmm.. well, actually compression_level could use a bit more info [19:57:46 CET] <analogical> the correct syntax for flac was "-compression_level" but on ffmpeg.org they typed "compression_level" which doesn't work [19:57:58 CET] <another> like which value are outside of the subset [19:58:02 CET] <TheWild> hello [19:58:22 CET] <analogical> they need to offer examples!! [20:00:20 CET] <TheWild> I have a 11 GB movie which has been copied multiple times over many computers. One computer had slightly damaged RAM. That was just one bit, but enough to cause mess during copying. [20:00:21 CET] <TheWild> The movie plays seemingly fine, but I'd like to check the consistency of the file. I don't have any checksum to check against. [20:00:21 CET] <TheWild> Is there a way to use ffmpeg to simply go through every frame and tell the errors? [20:00:46 CET] <another> https://ffmpeg.org/ffmpeg-all.html#Examples-1 [20:03:47 CET] <TheWild> okay, I'll just try ffmpeg -i video.mkv -f null - [20:16:10 CET] <kepstin> note that ffmpeg can only tell you if there's an error that causes something like a bad reference or syntax issue - it's possible (depending on the format/codec) that there are errors left that still decode fine but cause visible differences. [20:16:32 CET] <carltoso23> hi [20:19:11 CET] <ddubya> is there a way to see what pixel formats a filter supports [20:19:29 CET] <carltoso23> I tried to make a mosaic video with 4 videos all mkv videos, but the audio does not follow when playing the new mosaic mkv video [20:19:41 CET] <carltoso23> it doesn't have any audo [20:19:43 CET] <carltoso23> audio* [20:20:37 CET] <ddubya> so the file has no audio at all then? [20:21:02 CET] <carltoso23> yes no audio at all [20:21:39 CET] <ddubya> try a simpler example and see if you get audio.. like -vf unsharp or something [20:22:15 CET] <kepstin> carltoso23: when you'd using -filter_complex with video, you have to explicitly say what to do with the audio too, or you'll just get no audio [20:22:28 CET] <kepstin> what do you want to do? take the audio from one video? mix them all together? [20:22:54 CET] <carltoso23> I want audio from all of the 4 videos mixed into one [20:23:06 CET] <carltoso23> the final mosaic video [20:23:21 CET] <kepstin> carltoso23: then you have to add an "amix" filter to your filter chain to mix them together. [20:23:29 CET] <carltoso23> amix [20:23:34 CET] <carltoso23> i will look that up! [20:24:44 CET] <kepstin> it should work similarly to the filter you're using to make the video mosaic; you put multiple sets of filters into a single -filter_complex option by separating them with ; [20:25:23 CET] <carltoso23> "ffmpeg -i 1.mkv -i 2.mkv -i 3.mkv -i 4.mkv -filter_complex: -filter_amix "nullsrc=size=640x480 [base]; [0:v] setpts=PTS-STARTPTS, scale=320x240 [upperleft]; [1:v] setpts=PTS-STARTPTS, scale=320x240 [upperright]; [2:v] setpts=PTS-STARTPTS, scale=320x240 [lowerleft]; [3:v] setpts=PTS-STARTPTS, scale=320x240 [lowerright]; [base][upperleft] [20:25:23 CET] <carltoso23> overlay=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=320 [tmp2]; [tmp2][lowerleft] overlay=shortest=1:y=240 [tmp3]; [tmp3][lowerright] overlay=shortest=1:x=320:y=240" -c:v libx264 outpu2t.mkv" [20:25:25 CET] <carltoso23> ? [20:27:25 CET] <kepstin> uh, i have no ide what you just did [20:27:59 CET] <kepstin> you want to add ";[0:a][1:a][2:a][3:a]amix=inputs=4" fo the end of the existing -filter_complex argument [20:28:38 CET] <carltoso23> ah! [20:28:41 CET] <carltoso23> wonderful [20:28:55 CET] <carltoso23> i just added "-filter_complex: (-filter_amix) " [20:29:39 CET] <kepstin> (newer ffmpeg versions have filters that can do the video mosaic in fewer steps, but that'll work fine) [20:33:17 CET] <carltoso23> ah xstack? [20:33:22 CET] <carltoso23> -filter_complex;[0:a][1:a][2:a][3:a]amix=inputs=4 [20:33:25 CET] <carltoso23> correct ? [20:34:05 CET] <another> replace the semicolon after filter_complex with a space [20:37:26 CET] <carltoso23> ah thanks [20:37:39 CET] <carltoso23> " Unable to find a suitable output format for 'nullsrc=size=4608x2880 [base]; [0:v] setpts=PTS-STARTPTS, scale=1152x720 [upperleft]; [1:v] setpts=PTS-STARTPTS, scale=1152x720 [upperright]; [2:v] setpts=PTS-STARTPTS, scale=1152x720[lowerleft]; [3:v] setpts=PTS-STARTPTS, scale=1152x720 [lowerright]; [base][upperleft] overlay=shortest=1 [tmp1]; [20:37:39 CET] <carltoso23> [tmp1][upperright] overlay=shortest=1:x=320 [tmp2]; [tmp2][lowerleft] overlay=shortest=1:y=240 [tmp3]; [tmp3][lowerright] overlay=shortest=1:x=320:y=240'nullsrc=size=4608x2880 [base]; [0:v] setpts=PTS-STARTPTS, scale=1152x720 [upperleft]; [1:v] setpts=PTS-STARTPTS, scale=1152x720 [upperright]; [2:v] setpts=PTS-STARTPTS, scale=1152x720[lowerleft]; [20:37:40 CET] <carltoso23> [3:v] setpts=PTS-STARTPTS, scale=1152x720 [lowerright]; [base][upperleft] overlay=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=320 [tmp2]; [tmp2][lowerleft] overlay=shortest=1:y=240 [tmp3]; [tmp3][lowerright] overlay=shortest=1:x=320:y=240: Invalid argument" [20:37:44 CET] <carltoso23> ran into this error [20:37:56 CET] <carltoso23> ah I have to edit the last part [20:38:00 CET] <snooky> how can I insert a video device as an overlay in a video? [20:39:02 CET] <carltoso23> "ffmpeg -i 1.mkv -i 2.mkv -i 3.mkv -i 4.mkv -filter_complex [0:a][1:a][2:a][3:a]amix=inputs=4 "nullsrc=size=4608x2880 [base]; [0:v] setpts=PTS-STARTPTS, scale=1152x720 [upperleft]; [1:v] setpts=PTS-STARTPTS, scale=1152x720 [upperright]; [2:v] setpts=PTS-STARTPTS, scale=1152x720[lowerleft]; [3:v] setpts=PTS-STARTPTS, scale=1152x720 [lowerright]; [20:39:03 CET] <carltoso23> [base][upperleft] overlay=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=1152 [tmp2]; [tmp2][lowerleft] overlay=shortest=1:y=720 [tmp3]; [tmp3][lowerright] overlay=shortest=1:x=1152:y=720" -c:v crf 23 -preset medium -movflags +faststart -c:a aac libx264 outpu2t.mp4" [20:40:23 CET] <snooky> https://nopaste.linux-dev.org/?1270917 [20:41:47 CET] <snooky> and there i would have /dev/video2 add as "overlay" [20:41:56 CET] <furq> you can't use -vf twice and you can't use filters with -c:v copy [20:42:04 CET] <furq> and you can't use overlay with -vf [20:43:09 CET] <snooky> with this script [20:43:19 CET] <snooky> i became an rtsp stream WITH logo as overlay [20:46:12 CET] <snooky> can I then add / dev / video1 and / dev / video2 together and then put the overlay over it? [00:00:00 CET] --- Sun Nov 3 2019

1 0

ffmpeg-devel.log.20191101
by burek 02 Nov '19

02 Nov '19

[00:24:38 CET] <philipl> So, I'm continuing to look at improving the handling of HLS discontinuities. Clearly, I have problems. Discontinuous segments are marked in the m3u8. That's great. One idea would be to mark the first packet out of the segment as discontinuous (a new packet flag) and then we'd consider how to treat that in other places. But there's a pretty thick layer between segments and packets. We use avio to [00:24:44 CET] <philipl> create the appearance of a continuous stream and packets are read out of that. This means that at the time a segment is opened, and we detect a discontinuity, we're actually ahead of packets being read out of the stream. The next packet isn't the one that needs to be marked; it might be a few packets later. [00:25:51 CET] <philipl> I can't see a way to carry the discontinuity marker out-of-band. Do I have to insert an artificial 'barrier' into the avio stream? Seems that way. [00:26:10 CET] <nicolas17> you mean when a discontinuity is found in the m3u8 you're still halfway processing packets in the previous .ts? [00:33:28 CET] <philipl> Well, at least for ffmpeg.c, av_read_packet seems to be a couple of packets behind. [00:34:01 CET] <philipl> And I can't attach a flag to the packet until it's read. [00:43:27 CET] <nicolas17> I thought of adding extra readahead for the (misguided?) use-case of transcoding from a remote hls vod (for fast encodes, the latency of downloading new segments becomes a bottleneck) and it didn't seem easy to implement [01:18:00 CET] <philipl> Looks like what we'd really have to do is modify the first packet out of the stream to set the mpeg ts discontinuity indicator - it's the only inline way to do it, and probably semantically reasonable. [01:18:13 CET] <philipl> Then the mpegts demuxer can set the packet flag. [01:22:49 CET] <philipl> The hls demuxer feeds everything into a new stream that goes into the mpegts demuxer and that's what returns the packets. so it's completely decoupled. [10:30:56 CET] <cone-972> ffmpeg 03Steven Liu 07master:2dd71bf95e36: doc/filters: fix overlay_opencl document typo [11:38:09 CET] <JEEB> > color callback is unsigned int > color is "int" in style [13:40:04 CET] <cone-927> ffmpeg 03James Cowgill 07master:bb718d11ed99: avcodec/libtwolame: fix mono default bitrate [15:58:17 CET] <JEEB> This text should be bold, italics and underline This text should be small and green [15:58:23 CET] <JEEB> welp, now my code generates :things: [20:00:37 CET] <cone-187> ffmpeg 03Mattias Wadman 07master:e447a4d112bc: libavformat/flacenc: reject too big picture blocks [20:00:37 CET] <cone-187> ffmpeg 03Limin Wang 07master:a5e24be52ab0: swscale/swscale_unscaled: fix gbrap10be md5 different on big endian system [20:00:37 CET] <cone-187> ffmpeg 03Limin Wang 07master:99fc5059fbee: avfilter/vf_lut3d: simplify code [20:00:37 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:f011572e66c8: avcodec/utils: Check block_align [20:32:25 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:9ee5096068f6: avcodec/truemotion2: Fix several integer overflows with *Yo, *Uo, *Vo [20:32:26 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:2b655f55eaf0: avcodec/truemotion2: Fix several integer overflows in tm2_low_res_block() [20:32:27 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:7730bacb413f: avcodec/aacdec_template: Check samplerate [20:32:28 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:47a017aaaae3: avcodec/rv10: Check input for minimal frame size [20:32:29 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:98a257c3235b: avcodec/aptx: Check the number of channels [20:32:30 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:fd3ee7a92e92: tools/target_dec_fuzzer: Adjust threshold for G2M [23:20:30 CET] <cone-187> ffmpeg 03Andriy Gelman 07master:1aec1183f3e8: avcodec/v4l2_buffers: Fix infinite loop [23:20:31 CET] <cone-187> ffmpeg 03Andriy Gelman 07master:53c21c2d6beb: avcodec/v4l2_m2m: Fix indentation [23:22:13 CET] <tmm1> i'm trying to teach ffprobe about FF_CODEC_PROPERTY_CLOSED_CAPTIONS but it appears not be set inside show_stream() [00:00:00 CET] --- Sat Nov 2 2019

1 0

ffmpeg.log.20191101
by burek 02 Nov '19

02 Nov '19

[01:08:41 CET] <KombuchaKip> JEEB: Alright, so found the problem. Not a damn thing wrong with ffmpeg. I was just doing some pointer arithmetic wrong. Thanks for your help. [01:58:23 CET] <a0z> I'm trying to rip subs from a few VOB files, but the subtitle streams doesn't start until 25megs into the VOB file, how do I use ffmpeg to start the stream processing from the starting point of the subs stream? see here for what i tried: http://paste.debian.net/1112440/ interesting around lines 57-64 [01:59:20 CET] <pink_mist> why does it matter if it starts 25MB in? [02:00:08 CET] <a0z> well, when i try to rip the subs ffmpeg says there is no subtitle stream, however when transcoding the audio/video there was no problem as ffmpeg finds those streams okay [02:02:58 CET] <nicolas17> a0z: note that DVD subtitles are image overlays, so you can't get them into srt (text) format... unless you do character recognition [02:03:09 CET] <a0z> ah, i see [02:03:38 CET] <nicolas17> but that's not an answer to your problem :) [02:04:41 CET] <furq> a0z: -probesize 30M [02:05:05 CET] <nicolas17> furq: that goes before -i right? [02:05:10 CET] <furq> yeah [02:05:21 CET] <furq> you might also need to set -analyzeduration to some value higher than 5M [02:31:35 CET] <a0z> thanks, that did it! [06:26:42 CET] <pk08> hi, is there any way to find muxing mode of closed caption from mpeg-ts? [06:27:15 CET] <pk08> in ffprobe it shows "ATSC A53 Part 4 Closed Captions" for every CC types [06:27:42 CET] <pk08> but i need to know if its SCTE 128 or A53 [10:13:23 CET] <jancoow8> Hi, this should cut a video from 06:39:00 to 07:39:30 right [10:13:24 CET] <jancoow8> ffmpeg -ss 06:39:00 -i ThunderDome\ 2019\ Full\ Version\!-YtxMfkRjhjM.mkv -codec copy -t 07:39:30 sefa.mkv [10:13:43 CET] <JEEB> t is duration [10:13:59 CET] <jancoow8> Ah right, that's why it isn't working [10:14:00 CET] <JEEB> so that cuts from the closest random access point before 06:39:00 [10:14:10 CET] <JEEB> (or at that point, if it has a random access point at that point) [10:14:30 CET] <JEEB> and then cuts off a thing with that duration [10:14:37 CET] <jancoow8> so how can I set an end time [10:15:06 CET] <JEEB> I think there might be something like -to [10:15:39 CET] <JEEB> -to time_stop record or transcode stop time [10:15:52 CET] <jancoow8> Ah right! [10:15:53 CET] <jancoow8> Thanks! [10:35:56 CET] <Botik2> hello [10:36:24 CET] <Botik2> I need help with retrieving 1 second interval frames from 29.97 fps video. Here is what I have tried: [10:36:38 CET] <Botik2> ffmpeg -i rugby\ football_1013.mp4 -vf "select='not(mod(n,30))',setpts='N/(30*TB)-(3*round(N/3000))/(30*TB)'" -f image2 rugby%08d.jpg [10:37:49 CET] <Botik2> The idea was to just retrieve every 30th frame and shift time by 3 frames every 3000 frames to compensate for difference between 29.97 and 30 fps. But it is not giving me desired results [10:40:22 CET] <Botik2> but i am getting 15 second shift around 84 minute. What am I doing wrong? [12:18:43 CET] <CounterPillow> JEEB: doesn't -ss before -i break -to? [12:19:38 CET] <CounterPillow> as in, -to starts behaving like -t because fast seek sets the time base to 0 [12:22:09 CET] <JEEB> CounterPillow: no idea tbqh :P [12:22:16 CET] <JEEB> I prefer -t since it's specific [12:22:41 CET] <CounterPillow> I just do the math myself, but it would be really nice if that thing were fixed [12:30:42 CET] <CounterPillow> Though to fix this we probably first need to look into bofa [12:39:44 CET] <Adcock> when will rav1e [12:39:59 CET] <Adcock> be usable using ffmpeg [12:40:01 CET] <Adcock> ? [12:43:13 CET] <JEEB> as soon as rav1e makes a release so that if they make changes to the C API we can check for the version [12:43:27 CET] <JEEB> I think the person who posted the patch for rav1e is one of the rav1e devs [12:43:52 CET] <JEEB> currently they're not doing semantic versioning so there's no way to check :P [12:44:06 CET] <Adcock> Okay. [12:44:17 CET] <Adcock> Thanks. [12:44:18 CET] <JEEB> in other words, until rav1e makes a release or at least starts using semantic versioning, the patch is not going in [12:44:24 CET] <JEEB> unfortunately [12:45:02 CET] <Adcock> Do you think rav1e will be a big impact in terms of encoding for normal people? [12:45:21 CET] <Adcock> Like me who has no dedicated hardware of encoding? [12:46:47 CET] <Adcock> *for [13:00:12 CET] <Adcock> Taking it to #daala.....:D [13:07:13 CET] <void09> anyone got a oneliner to convert a Bluray folder to .iso (properly) ? [13:07:20 CET] <void09> on linux [13:13:58 CET] <AiNA_TE> mkisofs -o blah.iso /home/user/dir_you_wanna_isoify [13:15:52 CET] <AiNA_TE> man mkisofs [13:18:36 CET] <void09> it's not that simple, filesystem has to be udf.. some version of it [13:23:20 CET] <AiNA_TE> google-fu says [13:23:23 CET] <AiNA_TE> mkisofs -udf -allow-limited-size -input-charset "utf-8" -o "$TargetFile" "$SourceDir" [13:23:34 CET] <AiNA_TE> but i cannot verify it, give it a shot though [13:31:02 CET] <void09> needs to be version 2.50 [15:53:16 CET] <Adcock> Guys how do I know if my GPU or CPU has hardware decoding for x264 or x265? [15:53:38 CET] <Adcock> Any way to test it? [15:58:46 CET] <DHE> you still need drivers, so you'll have to consult those drivers [15:59:07 CET] <DHE> on linux I'd run nvidia-settings and check what its capabilities are for vdpau [15:59:44 CET] <Mavrik> IIRC ffmpeg can list that too [15:59:49 CET] <Mavrik> Depending on OS [16:18:11 CET] <void09> how to output the resolution of a png image ? [16:18:48 CET] <void09> ffprobe.. ok [16:19:04 CET] <furq> ffprobe -v error -select_streams v:0 -show_entries stream=width,height -of csv=nk=1:p=0 foo.png | tr ',' 'x' [16:19:46 CET] <void09> nice thanks furq [16:23:20 CET] <relaxed> xsv [16:28:58 CET] <Radioactive> heyyyy [16:29:01 CET] <Radioactive> [ac3 @ 00000183297DE4C0] Specified channel layout '7.1' is not supported [16:29:16 CET] <Radioactive> since when was that a thing? ac3 used to support 7.1 just fine [16:30:07 CET] <furq> you're thinking of eac3/dd+ [16:30:37 CET] <Radioactive> this is for encoding [16:31:10 CET] <Radioactive> with libavcodec [16:31:28 CET] <Radioactive> basically i used to be able to select AV_CODEC_ID_AC3 [16:31:45 CET] <JEEB> changes related to that seem to be from like 2010 [16:31:58 CET] <Radioactive> is there any way to export ac3 with 8 channels? [16:32:02 CET] <Adcock> Mavrik: How? [16:32:07 CET] <JEEB> I would have been surprised if it ever worked [16:32:25 CET] <Radioactive> before i would do something like this _codecContext->channel_layout = av_get_default_channel_layout(numChans); [16:32:27 CET] <Radioactive> and it would work fine [16:32:37 CET] <Radioactive> with ac3, and mp2 as well [16:32:41 CET] <Radioactive> now both give that error [16:32:48 CET] <Radioactive> where numChans could be 8 [16:34:21 CET] <JEEB> 1aeb88b77ddbd3802127a61bf8610e55b6c49a6f from 2011 added the check for both AC3 encoders [16:34:27 CET] <JEEB> the integer only and the float one [16:35:51 CET] <JEEB> although that just renames the list it seems [16:35:59 CET] <JEEB> the list itself being there is even older [16:37:03 CET] <Radioactive> what does that mean "the check"? [16:38:05 CET] <JEEB> added a .channel_layouts attribute to the AC-3 encoder AVCodecs. although, as I noted I then checked the commit and the change was actually just renaming the lists [16:38:09 CET] <JEEB> not adding them there [16:38:34 CET] <Radioactive> is there somewhere i can see a list of channel layouts supported for ac3? (and mp2) [16:38:51 CET] <JEEB> b33dae5efbc23cafaeb646231a949c296574540b [16:38:59 CET] <JEEB> that is where the list seems to have gotten added [16:39:21 CET] <JEEB> Radioactive: yes, definitely [16:39:25 CET] <JEEB> since ffmpeg.c can also list them [16:39:27 CET] <furq> Radioactive: -h encoder=ac3 [16:39:37 CET] <JEEB> ffmpeg -h encoder=ac3 will output them, and they are available through the API [16:39:56 CET] <Radioactive> hmm cool [16:41:01 CET] <JEEB> yea, so it loops through the array of channel_layouts in the AVCodec [16:41:37 CET] <JEEB> so if you have a AVCodec *codec , you can if (codec->channel_layouts) { for (...) } [16:44:53 CET] <JEEB> (such lists should end with 0/NULL entry at the end) [16:45:07 CET] <Radioactive> mp2: "Supported channel layouts: mono stereo" [16:45:09 CET] <Radioactive> whaaaaaaa [16:45:18 CET] <Radioactive> i could have sworn mp2 supported multichannel audio [16:45:25 CET] <Radioactive> am i going crazy? [16:45:53 CET] <JEEB> yea I don't remember it ever doing that [16:46:04 CET] <Radioactive> maybe i didn't define an explicit channel layout before? [16:46:27 CET] <Radioactive> if i just set the number of channels but not the channel layout [16:46:34 CET] <Radioactive> and encode that, what happens [16:46:39 CET] <JEEB> maybe, but I have no idea if that would actually do [16:46:42 CET] <JEEB> or if it would let you pass [16:49:44 CET] <Radioactive> only one way to find out [17:02:10 CET] <Radioactive> "[mp2 @ 0000023219E34150] encoding 8 channel(s) is not allowed in mp2" [17:02:19 CET] <Radioactive> idk i feel like i've entered an alternate timeline [17:02:22 CET] <Radioactive> i swear this used to work [17:04:30 CET] <durandal_1707> never worked [17:06:16 CET] <DHE> maybe you encoded to a different format? [17:12:52 CET] <FOSSie_bear> hi, I have been searching for some solution to my problems when I try to convert a black white video to h264. Unfortunatley the picture gets a "greenish" tint all the time. I guess it is due to color space conversion... [17:13:27 CET] <FOSSie_bear> Has anyone had these problems before? Can you recommend a guide to correct these issues? [17:16:15 CET] <durandal_1707> full command and uncut console output missing [17:17:02 CET] <FOSSie_bear> Actually the video input is "ffv1 with bgr0(progressive)" and the output is "h264_nvenc with bgr0". Perhaps I quickly try libx264 first and then post the outputs [17:22:39 CET] <FOSSie_bear> https://pastebin.com/rcnY83Ct [17:25:35 CET] <durandal_1707> i do not think that is ffmpeg bug, both input and output are bgr0 [17:28:57 CET] <FOSSie_bear> ok. I was not sure whether h264_nvec has compatibility issues with bgr0 [17:29:32 CET] <FOSSie_bear> perhaps there are some internal conversions before hardware upload and after download :/ [19:09:41 CET] <realies> would it possible to use something like this as a ffmpeg filter? https://github.com/xiph/rnnoise/blob/master/examples/rnnoise_demo.c [19:11:35 CET] <durandal_1707> realies: already done [19:12:35 CET] <durandal_1707> http://ffmpeg.org/ffmpeg-filters.html#arnndn [19:13:26 CET] <realies> lol, guess I just need to feed it the model? [19:13:39 CET] <durandal_1707> yes [19:13:53 CET] <durandal_1707> see relevant github repo [19:15:25 CET] <realies> not sure were that is in xiph's rnnoise [19:15:40 CET] <realies> or where you could find trained models? [19:15:46 CET] <durandal_1707> another repo [19:15:54 CET] <durandal_1707> use search [19:16:23 CET] <durandal_1707> rnnoise-models [19:17:41 CET] <realies> :o [20:35:48 CET] <Retal> hello guys, i have many errors: Error while decoding stream #0:1: Error number -16976906 occurred [20:36:20 CET] <Retal> what means error 16976906? [21:04:22 CET] <realies> seems like this filter chain does not apply the noise reduction 'dynaudnorm=b=true:m=100:c=true, arnndn=model=/Users/realies/Documents/Projects/rnnoise-models/beguiling-drafter-2018-08-30/bd.rnnn, compand=0:1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0, [21:04:22 CET] <realies> loudnorm=i=-19:tp=-1:lra=6:measured_I=-25.97:measured_LRA=6.70:measured_TP=-3.34:measured_thresh=-36.40:offset=0.05:linear=true:print_format=json' [21:04:24 CET] <realies> any ideas why? [21:05:05 CET] <realies> separating it in two forks of ffmpeg seems to do it though [21:11:06 CET] <realies> weird [21:17:08 CET] <realies> nah, running in two forks does nothing either [21:23:51 CET] <durandal_1707> realies: whatever you do, you do nonsense filtering [21:23:59 CET] <realies> how come? [21:24:59 CET] <durandal_1707> you can not apply arnndn after dynaudnorm [21:25:14 CET] <realies> I thought the filters process the input one after another [21:27:06 CET] <realies> aaand wrong paths, seems to be working ok [21:28:07 CET] <realies> durandal_1707, why not arnndn after dynaudnorm though? [21:28:37 CET] <durandal_1707> because it changes audio volume [21:30:18 CET] <realies> arnndn seems to be able to handle it well [21:30:28 CET] <realies> at least bd.rnnn [21:31:03 CET] <realies> gotta do some A/B comparisons, though that by exaggerating the noise floor it might perform better suppressing the noise [23:39:41 CET] <Retal> i have many errors: Error while decoding stream #0:1: Error number -16976906 occurred What means error 16976906? [23:40:34 CET] <JEEB> try av_err2str ? [23:40:51 CET] <JEEB> unless that's what ffmpeg.c outputs and you're just looking at its terminal output [23:41:16 CET] <pink_mist> it certainly looks like terminal output to my inexperienced eyes [23:41:36 CET] <pink_mist> why are error numbers in the negative though? that seems very odd as well [23:42:49 CET] <klaxa> because positive numbers could be valid [23:43:09 CET] <klaxa> for whatever function returns the error [23:44:38 CET] <pink_mist> ah, I guess I see the point of it then [00:00:00 CET] --- Sat Nov 2 2019

1 0

ffmpeg-devel.log.20191031
by burek 01 Nov '19

01 Nov '19

[06:42:57 CET] <filternet> Hi dudes, i brought you C madness: cOut() in C alpha test version https://pastebin.com/miw35a1r [07:38:02 CET] <cone-005> ffmpeg 03Limin Wang 07master:7c872df3e175: avformat/hlsenc: fix the av_dirname path isn't include separator in the end of string [10:25:34 CET] <cone-005> ffmpeg 03Paul B Mahol 07master:86a42e954e8e: avfilter/af_afade: start crossfading only when first stream reached end [10:25:47 CET] <j-b> hello [10:30:35 CET] <durandal_1707> nobody's here [10:32:40 CET] <j-b> durandal_1707: you are. [10:47:58 CET] <cone-005> ffmpeg 03Paul B Mahol 07master:b414cff63040: avfilter/vf_median: add radiusV option [11:31:55 CET] <JEEB> huh, someone actually asked for AC-4 [11:36:14 CET] <durandal_1707> nothing to see, move on. [11:56:28 CET] <j-b> I do. [13:43:15 CET] <JEEB> did we have XML writing code with attributes somewhere? [13:43:26 CET] <JEEB> like <thing attr1 attr2> [13:43:48 CET] <JEEB> mostly thinking how to handle this best with the spacing etc, and if there's existing solutions we already have [13:45:11 CET] <thardin> ... there's an xml parser in the codebase now? [13:45:25 CET] <thardin> ah writing [13:45:29 CET] <thardin> yeah I think ffprobe has one? [13:45:53 CET] <JEEB> and yes, the code base does utilize libxml2 already, but I'm 100% not sure if I want to touch that for just writing [13:45:56 CET] <JEEB> ah yes [13:46:07 CET] <JEEB> I got looking at dashenc and that wasn't nice [14:41:35 CET] Action: JEEB has a crazy idea for using av dicts for attributes [16:22:37 CET] <JEEB> pfft. at least I'm gaining experience in ASS splitting [16:23:33 CET] <gnafu> {:-] [16:25:35 CET] <JEEB> so basically since a single styled section is a in TTML [16:26:18 CET] <JEEB> I have to have an on-going style override buffer [16:27:00 CET] <JEEB> then when I get the text callback, I write it out. [16:27:24 CET] <JEEB> in case of cancel_overrides or end I reset it [16:28:28 CET] <JEEB> although that will end up extra fun with <style overrides>TEXT<more style overrides> [16:28:36 CET] <JEEB> (with TEXT at the end) [16:30:20 CET] <JEEB> in which case I have to keep the current overrides still around (cached), quit the span, and update the cached style with the new override and be ready to write it out in case more text is received [16:35:03 CET] <JEEB> ah no [16:35:31 CET] <JEEB> I can actually have spans within spans [16:38:20 CET] Action: JEEB checks if xml:id has to be unique within the document [16:39:33 CET] <JEEB> > The values of all attributes of type ID (which includes all xml:id attributes) within a document are unique. [16:59:26 CET] <Lynne> durandal_1707: does the new channel layout api make ambisonics easier and able to be properly described? [17:08:55 CET] <durandal_1707> Lynne: yes, by adding new ambisonic channel order enum [17:10:10 CET] <durandal_1707> Lynne: i need to wait for some other developer with his channel layout patches to be posted around VDD days, because that certain other anonymous is actually paid to do it, unlike poor me [17:11:15 CET] <Lynne> so you know of someone else who's is working on ambisonics or just waiting for a drive-by patch? [17:11:50 CET] <Lynne> (the google and china people won't add the API because they're apparently uninterested in properly described ambisonics) [17:12:43 CET] <durandal_1707> Lynne: i do not know anonymous guy at all, i just know because someone else told me that, koda [17:16:24 CET] <Lynne> huh, nice [17:16:54 CET] <Lynne> how are you awake at such time btw? jetlag? [17:18:35 CET] <durandal_1707> what time? I'm in UTC [17:31:35 CET] <Lynne> oh, I thought you were in japan [17:31:59 CET] <durandal_1707> haha, i'm still lost in Tokyo [18:03:52 CET] <Lynne> exit google street view's VR mode. [20:00:18 CET] <durandal_1707> is there a way hot at runtime define functions in C? [20:00:26 CET] <durandal_1707> s/hot/how [20:01:01 CET] <JEEB> yes, having them in something you load up and then using known symbols to figure them out? [20:01:09 CET] <JEEB> or what do you exactly mean? [20:03:00 CET] <durandal_1707> i need to create N functions depending on user supplied number at runtime [20:03:21 CET] <BtbN> That's not how C works, no [20:13:23 CET] <kierank> durandal_1707: thats self modifying code like swscale [20:13:29 CET] <kierank> But just template the user cases [20:13:40 CET] <kierank> See how I did it for bit depth and intermediate for idct [20:14:49 CET] <JEEB> -41 [20:16:20 CET] <Lynne> if you don't mind gpu stuff you can use the vulkan code to generate glsl [20:40:45 CET] <durandal_1707> i can not use for loops in macro? [20:41:28 CET] <JEEB> why not? [20:41:56 CET] <JEEB> have the macro have f.ex. { } ? [20:42:01 CET] <durandal_1707> i mean with for loop define bunch of functions? [20:54:19 CET] <taliho> durandal: maybe use a linked list with function pointer as member? [20:56:20 CET] <durandal_1707> i need to define 32 * 4 functions, how to do it nice way ? without repeating #DEFINE_FUNCTION(0) - #DEFINE_FUNCTION(31) [20:58:14 CET] <jamrial> macro that expands to four macros that each expand to four macros that etc [22:48:08 CET] <BBB> what is the option again to not resize the output stream to a common resolution, but keep all images at their input's original resolution (even if they contain resizes)? [23:04:46 CET] <Lynne> as in disable filter autoinsertion/reinit when image size changes? -reinit_filter 0 [23:12:23 CET] <Lynne> BBB: as in disable filter autoinsertion/reinit when image size changes? -reinit_filter 0 [23:15:20 CET] <BBB> ty [00:00:00 CET] --- Fri Nov 1 2019

1 0

ffmpeg.log.20191031
by burek 01 Nov '19

01 Nov '19

[00:27:50 CET] <AlexApps> Hello, as I have discussed in previous messages, I am creating a script to generate slidehows for me. I want to have a unique caption drawn to the screen for each image using drawtext, I cannot find any way to pass a unique caption to drawtext every frame, is there any way to refer to the line of a text file correlating to the frame number? [00:29:54 CET] <furq> AlexApps: drawtext supports https://ffmpeg.org/ffmpeg-filters.html#Timeline-editing [00:30:14 CET] <furq> with that said you should probably just write a subtitle file and burn it in with the subtitles filter [00:35:46 CET] <AlexApps> furq: Thanks, will look into them both. :) [01:05:13 CET] <KombuchaKip> To decode an audio file and read its metadata, is it necessary to call read_header()? When is it necessary? https://ffmpeg.org/doxygen/trunk/structAVInputFormat.html#a286d65d159570516… [01:05:57 CET] <DHE> generally you should not call any function unless it begins with av* (except avpriv_*) [01:13:31 CET] <KombuchaKip> DHE: Makes sense which is why I was confused at seeing it used here in the OP's question. https://stackoverflow.com/questions/13592709/retrieve-album-art-using-ffmpeg [01:15:00 CET] <JEEB> that is weird [01:15:48 CET] <JEEB> usually if you want to do some pre-reading, you call avformat_find_stream_info [01:16:17 CET] <KombuchaKip> JEEB: Yeah, hence my confusion. But I noticed none of the streams in an audio file contain any disposition bits set, including AV_DISPOSITION_ATTACHED_PIC. When I use ffprobe I see them set. So I was wondering if there was something I had to call first. [01:20:51 CET] <JEEB> KombuchaKip: you can literally see what it does in open_input_file() in fftools/ffprobe.c [01:21:59 CET] <KombuchaKip> JEEB: Yeah, I'm aware of that. It calls avformat_open_input() which in turn can call read_header(). So I'm not sure why my disposition bits are always all zero when ffprobe shows they're not. [01:24:03 CET] <JEEB> it does open input, by default it also does avformat_find_stream_info since "find_stream_info" is 1 by default [01:24:25 CET] <KombuchaKip> JEEB: So then I'm stumped. [01:24:43 CET] <JEEB> does set some options for the stream info opening and the lavf context opening, [01:24:52 CET] <JEEB> but that's as far as it seems to go [01:25:22 CET] <JEEB> well, it does seem to also just in case open decoders for all streams too [01:25:32 CET] <JEEB> but that shouldn't affect the AVFormatContext [01:25:52 CET] <JEEB> since a new lavc context is opened and just parameters taken from the AVStream's codecpar [01:26:56 CET] <KombuchaKip> JEEB: ffprobe(1) shows DISPOSITION:attached_pic=1 is set on the input file, but my own code the bit is never set. [01:28:10 CET] <JEEB> same file and open+find stream info? [01:29:31 CET] <JEEB> and you're looking at the place where ffprobe.c's show_stream is? [01:29:42 CET] <KombuchaKip> JEEB: I call avformat_open_input(), avcodec_alloc_context3(nullptr), then loop over each stream looking for disposition bit. [01:30:16 CET] <KombuchaKip> JEEB: Yeah, I looked at that last night. It's just checking to see if disposition bit is set, same as my code. [01:30:20 CET] <JEEB> try adding avformat_find_stream_info [01:30:31 CET] <JEEB> since that definitely is getting done by default in ffprobe [01:30:44 CET] <JEEB> (you can disable it in ffprobe by giving it an option it seems) [01:30:48 CET] <KombuchaKip> JEEB: Yeah, I see that. Let me give it a go. [01:32:01 CET] <KombuchaKip> JEEB: I'm looking through the man page now. [01:33:15 CET] <KombuchaKip> JEEB: I'm guessing it's -find_stream_info=false, based on --help, but it's not documented in the man page. [01:34:09 CET] <JEEB> probably just -find_stream_info 0 [01:34:53 CET] <KombuchaKip> JEEB: Tried that. Chokes on it. 'Argument 'foo.mp3' provided as input filename, but '0' was already specified. [01:38:21 CET] <JEEB> that is weird [01:38:39 CET] <JEEB> I mean I look at cmdutils how OPT_BOOL is supposed to work [01:38:42 CET] <KombuchaKip> JEEB: It's argument is declared as a OPT_BOOL. I'm trying to figure it out. [01:38:44 CET] Action: KombuchaKip nods [01:38:46 CET] <KombuchaKip> haha [01:40:45 CET] <JEEB> right [01:40:48 CET] <JEEB> -no-blah [01:40:50 CET] <KombuchaKip> JEEB: I can't find anywhere online of anyone actually using it. [01:41:02 CET] <JEEB> if (!po->name && opt[0] == 'n' && opt[1] == 'o') { [01:41:28 CET] <KombuchaKip> JEEB: lol, yeah there we go. [01:42:19 CET] <KombuchaKip> JEEB: Ok, so even then with disabling find_stream_info branch ffprobe still shows DISPOSITION:attached_pic=1. [01:43:00 CET] <JEEB> (Ž4@) [01:43:26 CET] <JEEB> need to get some sleep but you're API wise doing everything right with open and read_info [01:43:37 CET] <JEEB> only things that could be affecting it are some probing options [01:43:45 CET] <JEEB> like probesize and analyzeduration [01:44:00 CET] <JEEB> but if you're not setting any of those and ffprobe isn't either [01:44:02 CET] <KombuchaKip> JEEB: Yeah, but possibly read_info isn't being called on my end. I'll build ffmpeg with debugging symbols, link and debug. [01:44:03 CET] <JEEB> then nfi [01:44:24 CET] <KombuchaKip> JEEB: Yeah I don't think I'm fiddling with those. [01:55:31 CET] <KombuchaKip> JEEB: Thanks for your help. Have a good night. I'll figure it out. [02:13:41 CET] <TechnicalMonkey> so can any one help me to figure out what I did wrong in my ffmpeg live stream attempt? [02:14:03 CET] <TechnicalMonkey> https://pastebin.com/kLBnHhzd [02:15:16 CET] <TechnicalMonkey> I guess I came at a bad time [02:15:22 CET] <furq> you definitely don't want -pix_fmt bgr24 [02:16:04 CET] <TechnicalMonkey> oh...? why not....? [02:16:27 CET] <TechnicalMonkey> I don't mind that the video gets encoded in 420 [02:16:42 CET] <TechnicalMonkey> but I need the first gen input to be RGB [02:16:56 CET] <furq> then you want -pixel_format bgr24 before the input [02:17:05 CET] <TechnicalMonkey> and that is the colorspace that it lists [02:17:07 CET] <furq> you've got it as an output option [02:17:16 CET] <TechnicalMonkey> oooohh [02:17:57 CET] <TechnicalMonkey> does it default to 420 if I delete that swtich? [02:18:17 CET] <furq> it'll be yuv444p i think [02:18:21 CET] <furq> so yeah explicitly set yuv420p [02:20:08 CET] <furq> also if -s and -r are supposed to apply to the input then move those as well [02:20:41 CET] <furq> actually nvm you have -video_size already [02:23:34 CET] <TechnicalMonkey> cool [02:23:48 CET] <TechnicalMonkey> this is the farthest I've ever gotten [02:24:09 CET] <TechnicalMonkey> but now it just keeps giving me these red messages [02:24:49 CET] <TechnicalMonkey> rtbufsize too full or near too full [02:27:10 CET] <furq> yeah that means what you think it does [02:27:36 CET] <furq> if you can't bump rtbufsize then set a lower video_size or framerate on the input [02:27:59 CET] <furq> if your capture source supports that [02:33:42 CET] <TechnicalMonkey> so I lowered my input to 720p [02:35:01 CET] <TechnicalMonkey> it lasted longer but still came up with that message about the rtbufsize [02:35:08 CET] <furq> what cpu is that [02:35:23 CET] <TechnicalMonkey> i7-3770 [02:35:45 CET] <furq> that should be able to keep up with encoding then [02:35:56 CET] <furq> obviously if the status line shows the speed dropping below 1x then that's the issue [02:36:01 CET] <TechnicalMonkey> overall CPU usage didn't go past 40% [02:36:58 CET] <furq> i take it your internet can keep up with 8mbit upload as well [02:38:48 CET] <TechnicalMonkey> I have gigabit FiOS [02:41:05 CET] <TechnicalMonkey> I'm only uploading to the most reliable of ingest servers [02:41:34 CET] <TechnicalMonkey> every time I try to use NYC servers [02:41:38 CET] <TechnicalMonkey> I have issues [02:42:05 CET] <TechnicalMonkey> I can try it [02:52:05 CET] <TechnicalMonkey> so I no longer get those messages about rtbufsize in red [02:52:13 CET] <TechnicalMonkey> but now I get yellow messages [02:53:16 CET] <TechnicalMonkey> wait... if I touch the scroll bar in CMD while streaming it can pause?!? [02:54:37 CET] <TechnicalMonkey> the yellow message is about something being too large [02:57:12 CET] <ponyrider> CFS-MP3: that command works for me [03:02:36 CET] <TechnicalMonkey> furq: thanks for the help [03:02:53 CET] <TechnicalMonkey> this really opens things up now [03:03:16 CET] <TechnicalMonkey> both the audio and the video worked without any flaws [03:03:44 CET] <TechnicalMonkey> I got a whole bunch of messages from ffmpeg screaming at me [03:03:55 CET] <TechnicalMonkey> but the audio and the video worked [03:05:51 CET] <TechnicalMonkey> I also tested in 1080p as well [03:05:57 CET] <TechnicalMonkey> both at 60fps [03:06:04 CET] <TechnicalMonkey> and it worked really well [06:16:05 CET] <derlg> Hey all, I was hoping someone here might be able to point me in the direction of where I might find some answers as to ffmpeg usage of a specific command... [06:16:39 CET] <derlg> I was wondering how(if at all possible) to save a video made using libcaca? [06:17:13 CET] <derlg> secondly, if i wanted to add audio to that video, would i re-run ffmpeg, appending audio to it? [06:18:03 CET] <derlg> my initial command ive been using is "ffmpeg -i input.mp4 -pix-fmt rgb24 -f caca out.mp4" [06:18:33 CET] <derlg> i was hoping to get something similar to "mpv video.mp4 -vo caca" but be able to save it as a file [06:25:36 CET] <furq> derlg: you can't do that in a single command [06:25:44 CET] <furq> caca is an output device that renders directly to a window or to stdout [06:25:56 CET] <furq> so you'd have to capture that window or something [06:32:21 CET] <derlg> furq: thanks :) would you know how I could go about using ffmpeg to capture the output from that window? i vaguely recall seeing something about capturing streams [06:33:06 CET] <furq> https://www.ffmpeg.org/ffmpeg-devices.html#x11grab [06:35:39 CET] <derlg> thanks furq :) [07:07:41 CET] <KombuchaKip> JEEB: So it turns out that the disposition bits are being set, but something is clobbering them between after avformat_open_input() and my reading them later. I'll experiment with a hardware breakpoint tomorrow. [08:52:18 CET] <BeerLover> How can I pipe the output of ffmpeg ? [08:53:43 CET] <BeerLover> This creates multiple segment files and an index.m3u8 file "ffmpeg -re -y -i song.mp4 -profile:v baseline -b:a 320k -hls_time 10 -hls_allow_cache 1 -level 4.0 -hls_segment_filename segment%d.ts -f hls index.m3u8". I want to pipe those files to "aws s3 cp <file> s3://<bucket>/<file>" [09:04:57 CET] <furq> BeerLover: that doesn't make sense for a few reasons with the hls muxer [09:05:16 CET] <furq> probably just use inotifywait or something [09:24:55 CET] <BeerLover> What exactly does -re do? I don't understand from the man pages [09:25:04 CET] <BeerLover> it slows the transcoding [09:27:43 CET] <JEEB> BeerLover: basically looks at the input timestamps and sleep()s the difference [09:27:50 CET] <JEEB> that way attempting to simulate a live input [09:28:02 CET] <JEEB> that can also fail in hilarious ways so unless you really need it, don't use it [10:10:42 CET] <Rhada> Hello, i'm using ffmpeg to create a dash stream from hls live source. Unfortunately, if the source stream is restarted, the media sequence is reset and ffmpeg is stuck complaining about it ([hls @ 0x5588820eaf40] Media sequence changed unexpectedly: 21 -> 0). I can't achieve to make my command exit on this input error, or take the new media sequence as the one to use ? [10:13:16 CET] <JEEB> yea it warns about that but then does nothing :P [10:13:31 CET] <JEEB> it's technically invalid I think according to the HLS RFC [10:13:41 CET] <JEEB> but still what the HLS reader does is not really optimal [10:13:47 CET] <JEEB> since it will start waiting for the next time it gets 22 [10:13:52 CET] <JEEB> if that ever comes, that is [10:13:57 CET] <Rhada> yes but after this it keep looping over the same playlist without doing any encoding [10:13:57 CET] <JEEB> it gets even more fun with larger values [10:14:05 CET] <JEEB> yes [10:14:15 CET] <JEEB> since it's waiting until it gets back to "sync" [10:14:19 CET] <Rhada> yes you get the point [10:14:46 CET] <JEEB> basically the hls module needs a rework but I think it's unlikely to happen unless someone sponsors it (or someone cares enough) [10:14:49 CET] <Rhada> if needed, the command i use : ffmpeg -re -i http://mydomain.source/playlist.m3u8 -c:a copy -c:v copy -vtag avc1 -atag mp4a -bsf:a aac_adtstoasc -map 0:p:5 -map 0:p:4 -map 0:p:3 -map 0:p:2 -b:v:0 2500k -b:v:1 1500k -b:v:2 1000k -b:v:3 500k -streaming 1 -use_template 1 -index_correction 1 -window_size 30 -extra_window_size 30 -seg_duration 2 -remove_at_exit 1 -adaptation_sets "id=0,streams=v id=1,streams=a" -f dash /tmp/index.mpd [10:15:09 CET] <JEEB> this specific bug could be improved by just making the HLS part reset sync [10:15:22 CET] <JEEB> as in, re-init playback and buffer those three segments again [10:15:30 CET] <JEEB> but in general there's various issues with the HLS reader [10:16:53 CET] <Rhada> yes, i was searching for a magic option doing this :( [11:06:01 CET] <Spring> Shot in the dark but thought perhaps someone here may know. I used OBS with x264 as the encoder and it captured the contents of a white (#ffffff) desktop screen as a darker #fdfdfd instead. Anyone know if that's someone on x264's side? [11:07:00 CET] <Spring> it's an extremely subtle difference but will mean that I won't be able to use a blend mode in a video editor as-is and will require curve adjustment to compensate, unfortunately. [11:08:33 CET] <Mavrik> Spring: side effect of colorspace conversion perhaps? [11:08:44 CET] <Mavrik> Did you try capturing into RGB instead of YUV if OBS supports that? [11:09:37 CET] <TheAMM> (It does) [11:09:49 CET] <Spring> Mavrik, I also just tested an ffmpeg conversion to VP9 and the same thing occurs [11:10:06 CET] <Mavrik> Yes, pretty much all video formats will do YUV420 color conversion. [11:10:24 CET] <Mavrik> So changing codecs up and down won't make a difference :P [11:11:09 CET] <Spring> TheAMM, is this an option of a later version of OBS? [11:11:37 CET] <TheAMM> It is in whatever version I have installed, which is months old by now [11:12:00 CET] <TheAMM> It's in the advanced tab iirc, next to the renderer [11:12:32 CET] <Mavrik> Apparently, https://4.bp.blogspot.com/-YEE1aCJRvxE/W2rceTE8EWI/AAAAAAAAAJ4/EoDEli6O9H43… [11:13:03 CET] <Mavrik> Note that a lot of software won't read RGB encoded video, but you can use it to check if that's an issue for you. [11:13:41 CET] <Spring> so YUV420 can't encode to pure white or is it the conversion incorrectly mapping the color/shade? [11:14:19 CET] <Spring> *doesn't support (rather than 'can't encode') [11:16:00 CET] <Mavrik> Might be it's just the mapping [11:16:12 CET] <Mavrik> There's an SO article where someone has the same issue with BMPs: https://video.stackexchange.com/questions/19944/ffmpeg-bmp-to-yuv-x264-colo… [11:16:19 CET] <Mavrik> And it seems forcing the format before conversion fixed it [11:17:22 CET] <Mavrik> Although in my experience no tweaking will fix the red color shift [11:17:27 CET] <Mavrik> Except using RGB or YUV444 [11:23:15 CET] <Spring> hmm, OBS states I'm up to date. Apparently OBS Studio is the successor that TheAMM must be referring to. [11:23:32 CET] <TheAMM> Oh, yes [11:23:36 CET] <TheAMM> Studio happened years ago? [11:24:13 CET] <BeerLover> JEEB I want to transcode mp4 to HLS, so do I need it? [11:24:33 CET] <BeerLover> It's just for simulation, won't affect transcoding in any way right? [11:24:57 CET] <Spring> TheAMM, quick question, do you know if Studio installs on Windows to the same location? [11:25:11 CET] <TheAMM> I don't [11:26:42 CET] <Spring> it's a pity I spent hours capturing this only to discover in the end the mapping issue. RIP. [11:27:26 CET] <Spring> actually, is there any way after the fact I could use ffmpeg to map that particular color to pure white? [11:28:59 CET] <TheAMM> You can adjust the ranges with different filters [11:29:38 CET] <TheAMM> Abusing format to convere between the yuv ranges, using some other filter to do the same [11:30:19 CET] <TheAMM> Can't remember the invocations now but manual has the parameters [11:31:29 CET] <durandal_1707> lutyuv [11:33:17 CET] <Spring> would this require conversion to YUV444 btw? [11:33:29 CET] <Spring> or does YUV420 support pure white? [11:33:29 CET] <durandal_1707> no [11:34:06 CET] <durandal_1707> or lutrgb if your capture is RGB [11:36:46 CET] <Spring> Oh boy, the methodology for the lut filter is a bit over my head. I'm only familiar with some file-based LUTs. Any tips on what syntax I'd need for this purpose? [11:41:20 CET] <TheAMM> I was supposed to calibrate my capture card and make the output proper with a LUT but never bothered [11:41:43 CET] <TheAMM> I have some format filter in there to make the result look "eh, ok" [11:42:00 CET] <durandal_1707> lutrgb=r='if(eq(val,0xfd),0xff,val)':g='if(eq(val,0xfd),0xff,val)':b='if(eq(val,0xfd),0xff,val)'" [11:42:57 CET] <Spring> durandal_1707, thank you. If the input was YUV240 would I be able to just switch the 'lutrgb' to 'lutyuv'? [11:43:04 CET] <durandal_1707> nope [11:43:19 CET] <TheAMM> Doesn't that just map FD to FF? [11:43:33 CET] <durandal_1707> yes [11:43:51 CET] <TheAMM> Well it's going to make gradients look weird [11:44:54 CET] <durandal_1707> it could map also FE to FF [11:45:10 CET] <Spring> I think it might be alright in my case since I don't mind so much about the transition but rather surrounding white which I need to use a Multiply blend mode with. [11:45:39 CET] <Spring> (hence it needs to #fff for that, which is just a solid fill) [11:45:52 CET] <TheAMM> I meant that there's going to be a sharp jump to pure white instead of adjusting the entire (limited) range of colors to the full scale [11:46:07 CET] <TheAMM> If it's good enough it's good enough [11:47:02 CET] <Spring> so for that above command the process would be conversion to RGB w/ lut filter then transcoding again to YUV I suppose. [11:50:28 CET] <durandal_1707> Spring: it your capture input is YUV you will need to use different lutyuv incarnation, no need to do another colorspace conversion which is not lossless [11:51:32 CET] <durandal_1707> but if you need to process blending in RGB while your input is YUV than above command should be applied before blending [12:01:20 CET] <Spring> it's more that I'm not familiar with what the lutyuv values should be :p Having a bit of trouble though with converting to RGB using -pix_fmt' as it comes out looking all distorted. Is there something I'm missing? [12:18:05 CET] <Spring> Hmm, it seems I need some script for the conversion? Wish I could find a Python script to run or some result that isn't ambiguous code. It's actually for a video for a funeral in just over a day so weighing whether I should re-capture it all in the next few hours or try to get this post-processing working. [12:19:39 CET] <Spring> *result online, that is. Maybe I should try Google rather than DDG. [12:24:30 CET] <mifritscher> JEEB: my wild guess was right - it crashes at muc.c - st->internal->priv_pts->val = pkt->dts; (st being an AVStream) [12:24:47 CET] <mifritscher> st->internal is ok, sp priv_pts seems to be NULL [12:40:50 CET] <Spring> Oh my goodness... Turns out that after all that the video editor displays the white as #fff... I actually found an easier YUV->RGB conversion in the meantime though: converting to PNG frames. But unnecessary now. [12:41:01 CET] <Spring> sorry for all the wall of texts btw [14:01:41 CET] <ocx32> hi all, is it possible to use ffmpeg to extract part of a live stream as an mp4 file? for example i am watching now a livestream and want to extract the last 30secs of this stream? [14:06:19 CET] <DHE> most likely, as long as you can play the stream with ffmpeg [14:06:29 CET] <DHE> or ffplay I guess [14:10:17 CET] <ocx32> DHE i was thinking of increasing the hls segment to say 1minute, and then when i want to extract i extract current segment and previous segment and then convert the hls segments to an mp4? [14:10:22 CET] <ocx32> is this the way to do it ? [14:11:15 CET] <DHE> umm.. if you're your own livestream source, I guess that would work. I was assuming ffmpeg would join the livestream as an ordinary client and do what you want [14:11:40 CET] <DHE> most HLS players intentionally start back ~3 segments before playing, so if you have 1 minute segments that's 210 seconds of lag (on average) to players [14:11:42 CET] <ocx32> i am using my own livestream source which is mainly writing to disk hls segments [14:12:06 CET] <ocx32> but should i add keyframes or so to join the hls segments into an mp4? [14:12:09 CET] <ocx32> technically speaking [14:12:31 CET] <DHE> the first frame of any segment is a keyframe. that's a requirement and enforced by the hls muxer [14:12:51 CET] Action: DHE uses 6 second segments [14:13:42 CET] <ocx32> ok so for this last 30 seconds extraction of a livestream should i just take the last 2 segments to be on the safe side, convert into mp4 and cut from the beginning till i have the last 30 sec? is this optimal? [14:21:25 CET] <ocx32> DHE can i do something like ffmpeg -i playlsit.m3u8 and pick the last 2 segments of the playlist ? [14:21:34 CET] <ocx32> any options for that? [14:24:17 CET] <DHE> ffmpeg -h demuxer=hls [14:25:44 CET] <ocx32> DHE -live_start_index <int> .D...... segment index to start live streams at (negative values are from the end) (from INT_MIN to INT_MAX) (default -3) [14:25:53 CET] <ocx32> so i should use -2 in my case? on that playlist ? [14:27:19 CET] <DHE> sounds about right [16:02:16 CET] <ocx32> DHE ffmpeg never stops when -2 is supplied on a LIVE stream as new segments are being generated always [16:13:33 CET] <DHE> ocx32: then specify -t 30 [16:19:08 CET] <ddubya> I have an HDV source rendered from Final Cut, interlaced 29.97. When I deinterlace some frames go backwards, what could cause it? [16:19:59 CET] <ddubya> it seems to be when a time dilation was applied in Final Cut, otherwise its ok [16:22:32 CET] <ocx32> DHE ok now it stops after 30 seconds, i am using -i playlist.m3u8 -live_start_index -2 -t 30 -acodec copy -vcodec copy out.mp4 , for some reason the video out.mp4 doesnt play i only see the first frame of the video [16:36:50 CET] <dont-panic> maybe this is a dumb question, but I use ffmpeg to stream to twitch.tv sometimes and I figured out how to use it send a stream through my rtmp server. I'm curious if there's a way I can stream some low latency audio and video in a way where I can host it on my linux computer. I'm trying to build a webcam baby monitor that I can put on a raspberry pi and make it portable and accessible from my phone [16:36:56 CET] <dont-panic> using the browser and stuff I already have [16:37:45 CET] <dont-panic> I'm currently using a program called motion in ubuntu 19.04 I think and that's very good for realtime video, but it doesn't include audio as far as I can tell [16:44:48 CET] <ddubya> dont-panic, I would try point-to-point streaming, https://trac.ffmpeg.org/wiki/StreamingGuide has examples [16:48:35 CET] <dont-panic> ddubya: I saw some stuff about that, but they all mention mplayer as the client for viewing the stream... is there a way to use the browser to view the stream like a live youtube/facebook video or something? [16:48:49 CET] <dont-panic> or is that a javascript/webdev question? [17:00:25 CET] <ddubya> dont-panic for the browser to view the stream I'm guessig none of those protocols will work. I've never tried though [17:55:37 CET] <fling> Can't I losslessly rotate mjpeg? [17:56:07 CET] <fling> Or should I just rotate via container metadata? [18:56:48 CET] <ocx32> DHE still on it... any idea? [19:28:07 CET] <kepstin> fling: in theory it should be possible to losslessly rotate mjpeg like tools can do with jpeg, in practice... patches to add a bsf to do that in ffmpeg might be accepted? [19:46:43 CET] <phobosoph> hi [19:46:58 CET] <phobosoph> anyone here? :) [19:46:59 CET] <phobosoph> :) [19:50:41 CET] <another> no [19:53:48 CET] <phobosoph> :) [19:53:57 CET] <phobosoph> so ffmpeg streams to Youtube Live and it works nicely for some hours [19:54:00 CET] <phobosoph> speed=1.x, also good [19:54:18 CET] <phobosoph> but then suddenly Youtube Live stops recognizing the stream and nothing can be seen on YouTube [19:54:24 CET] <phobosoph> in the meanwhile ffmpeg is still happily streaming on [19:54:32 CET] <phobosoph> but the speed dropped down from 1.x to much lower [19:54:34 CET] <phobosoph> and keeps dropping [19:54:42 CET] <phobosoph> and ffmpeg doesn't notice there is something wrong and exits with error [19:54:49 CET] <phobosoph> it just streams on and on and youtube finished the stream [19:54:55 CET] <phobosoph> restarting ffmpeg fixes the issue [19:55:09 CET] <phobosoph> so either ffmpeg doesn't terminate itself when it detects an issue with the connection to youtube rtmp [19:55:35 CET] <phobosoph> or better, the underlying issue for the dropping speed stat and youtube stopping recognizing the stream is found and fixed [19:55:37 CET] <phobosoph> any ideas? :/ [19:56:16 CET] <fling> paste your commandline [19:56:23 CET] <fling> and full output [19:56:30 CET] <fling> and wait for long time :> [20:05:36 CET] <phobosoph> pastie.org seems to be not working currently, I get a nginx welcome page [20:07:59 CET] <another> there are dozens of pastebin sites. just choose one [20:19:37 CET] <phobosoph> szre [20:20:29 CET] <phobosoph> here is the command: https://pastebin.com/u0z0mmZ9 [20:20:46 CET] <phobosoph> fling, another: ^ [20:21:10 CET] <phobosoph> when checking the ffmpeg logs after some hours when youtube suddenly stops showing the stream feed: [20:21:14 CET] <phobosoph> frame=20619 fps=1.9 q=-1.0 size= 516529kB time=00:11:27.10 bitrate=6158.3kbits/s speed=0.0649x [20:21:23 CET] <phobosoph> all the time fps=30 and speed=1.x [20:21:41 CET] <phobosoph> when I restart ffmpeg (kill + start again), youtube picks the stream up again instantly [20:21:57 CET] <phobosoph> sure, I could make some contraption that checks for the fps + speed and auto-kills ffmpeg [20:22:10 CET] <phobosoph> but it would be much better to find the reason - or let ffmpeg terminate itself with error when this anomaly happens [20:24:44 CET] <filpAM> http://filpam.com/pub/out.mp4 [20:25:12 CET] <filpAM> this plays with ffplay but not on my android phone or discord [20:25:48 CET] <another> yuv444 [20:25:56 CET] <filpAM> ffmpeg -f rawvideo -s 128x128 -r 60 -pix_fmt rgb24 -i ./tmp.rgb8 -c:v libx264 -vf scale=600:400:flags=neighbor out.mp4 [20:25:57 CET] <another> probably not supported [20:26:27 CET] <filpAM> this the command line that used to encode the video from rgb24 raw frames [20:26:49 CET] <filpAM> *that [20:27:47 CET] <another> add '-pix_fmt yuv420p' as an output option [20:28:14 CET] <filpAM> ok [20:30:54 CET] <filpAM> yep it works [20:31:07 CET] <filpAM> http://filpam.com/pub/out.mp4 updated [20:32:16 CET] <filpAM> thanks [20:45:31 CET] <jemius> There are so many video denoise filters in ffmpeg... how do I know which one to use for my scenario? [20:45:57 CET] <JEEB> see their age, and then make a comparison or so? [20:46:14 CET] <jemius> JEEB, newer = better ? [20:47:35 CET] <fling> is hqdn3d the best? [20:47:36 CET] <JEEB> not necessarily, but you get an idea of in which order things were added [21:01:25 CET] <jemius> Additionally, they all have ton of parameters :( [21:47:04 CET] <TechnicalMonkey> is q controlled by crf? [21:47:39 CET] <TechnicalMonkey> I keep getting yellow messages when I stream [21:48:43 CET] <TechnicalMonkey> Past Duration 0.xxxxx Too Large [21:49:30 CET] <saml> where do you stream? [21:49:47 CET] <TechnicalMonkey> to....? Twitch [21:51:57 CET] <saml> what's full command you use to stream? [21:53:36 CET] <furq> TechnicalMonkey: you can usually ignore that warning [21:53:59 CET] <furq> unless you're getting dropped frames or desyncs or something [22:02:09 CET] <kepstin> that message does often indicate dropped frames - the usual cause is that there's a filter which has set an output framerate (indicated cfr) but actually outputs vfr or at a framerate higher than indicated. [22:16:42 CET] <phobosoph> :( nobody [22:16:43 CET] <phobosoph> damn [22:31:16 CET] <CountPillow> how can I tell whether my ffmpeg has been built with ligma? [22:49:59 CET] <kepstin> CountPillow: what's ligma? [22:50:16 CET] Action: kepstin doesn't see anything by that name in the ffmpeg source tree [22:50:17 CET] <FooNess> (Don't fall for it.) [22:50:40 CET] <CountPillow> LIGMA NUTS LMAO GOT EM [22:50:44 CET] <FooNess> ... [22:50:48 CET] <FooNess> What are you, 12? [00:00:00 CET] --- Fri Nov 1 2019

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