February 2019 - Ffmpeg-devel-irc

ffmpeg-devel.log.20190217
by burek 18 Feb '19

18 Feb '19

[01:05:14 CET] <cone-794> ffmpeg 03Carl Eugen Hoyos 07master:fe7d8c993ff9: lavc/libgsmenc: Force mono and use 13k as default bitrate. [06:27:51 CET] <cone-029> ffmpeg 03Reto Kromer 07master:7a51fed0f047: doc/muxers: grammar fix [07:00:16 CET] <cone-029> ffmpeg 03Gyan Doshi 07master:a9452fe6dc2e: doc/fftools-common-opts: add example for codec option stream specifiers [10:53:12 CET] <cone-825> ffmpeg 03Michael Niedermayer 07master:d1afa7284c3f: avformat/matroskadec: Do not leak queued packets on sync errors [10:53:12 CET] <cone-825> ffmpeg 03Michael Niedermayer 07master:7246bf365a1f: avcodec/mpegvideo_enc: Use av_assert1() instead of assert() [10:53:12 CET] <cone-825> ffmpeg 03Kevin Backhouse via RT 07master:1f00c97bc347: avcodec/htmlsubtitles: Fixes denial of service due to use of sscanf in inner loop for tag scaning [10:53:12 CET] <cone-825> ffmpeg 03Kevin Backhouse via RT 07master:894995c41e07: avcodec/htmlsubtitles: Fixes denial of service due to use of sscanf in inner loop for handling braces [10:53:12 CET] <cone-825> ffmpeg 03Wenxiang Qian 07master:a142ffdcaec0: avformat/ftp: Fix Out-of-Bounds Access and Information Leak in ftp.c:393 [10:53:13 CET] <cone-825> ffmpeg 03Wenxiang Qian 07master:85f91ed760a5: avformat/http: Fix Out-of-Bounds access in process_line() [10:53:13 CET] <cone-825> ffmpeg 03Michael Niedermayer 07master:b687b549aa0f: avformat/webmdashenc: Check id in adaption_sets [10:53:14 CET] <cone-825> ffmpeg 03Steve Lhomme 07master:9326117bf63b: avformat/matroskadec: Check parents remaining length [21:29:02 CET] <durandal_1707> anybody know if mplayer2 code can be compiled with latest FFmpeg? [21:54:54 CET] <CoreX> was wondering if theres a way to port one of the filters to a dll [21:55:10 CET] <CoreX> like is there a guide online for it because im looking and not getting the best results [21:55:21 CET] <durandal_1707> dll for what? [21:57:37 CET] <CoreX> avs [21:58:08 CET] <JEEB> well it sounds like you want to first write a general wrapper using libavfilter [21:58:11 CET] <CoreX> guessing my stupid ass is asking a stupid question on a sunday night [21:58:20 CET] <JEEB> and then you can start enabling filters [21:58:31 CET] <JEEB> (or you take the filter's name in as a string) [21:58:39 CET] <JEEB> and/or the full filter chain [21:58:41 CET] <CoreX> ok thank you time to learn more [21:58:55 CET] <durandal_1707> CoreX: which ffmpeg filter? [21:58:57 CET] <JEEB> you can build libavfilter as either static or shared, and then link your thing against it [22:02:12 CET] <CoreX> yadif/w3fdif/bwdif/fieldmatch/seperatefieldmatch main ones i want todo [22:03:59 CET] <durandal_1707> some of them are already available in AVS or VS [22:32:56 CET] <CoreX> alright i'll check it out thank for the help guys [22:47:12 CET] <uau> durandal_1707: what do you want to test with mplayer2? [22:47:49 CET] <JEEB> probably the dshow wrapper thing? [22:54:10 CET] <durandal_1707> uau: yes, i give up with that approach, i'm now compiling mplayer with -m32 [23:13:13 CET] <durandal_1707> mplayer just crashed [23:18:58 CET] <durandal_1707> what i need to compile lavfilters? [23:20:33 CET] <JEEB> mingw-w64 for the FFmpeg, then free MSVC 2017 for the actual dshow part [23:22:36 CET] <nevcairiel> (or msvc for both also works) [23:26:43 CET] <nevcairiel> most people seem to struggle with getting an msys environment for building the ffmpeg part, its odd [00:00:00 CET] --- Mon Feb 18 2019

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ffmpeg.log.20190217
by burek 18 Feb '19

18 Feb '19

[01:32:19 CET] <keegans> when using the hls api i can only get the samples from the first stream [01:32:27 CET] <keegans> how can i select which stream I'd like to get samples from? [01:32:34 CET] <keegans> or are they tracks? (each segment in the HLS stream ) [01:39:56 CET] <keegans> is this a bug or am i misunderstanding the code : https://github.com/FFmpeg/FFmpeg/blob/32fb83e4318845a7f44f7bed02d5963b9e818… ? [01:40:24 CET] <keegans> because my current issue is that av_read_frame only gives me the samples in the first segment of the HLS file [01:59:56 CET] <keegans> k , i'll be back and fix this bug [08:24:44 CET] <linux_gnu> Hi ,how can I use ffmpeg to record the audio from my audio card ,like if i m listenning to a web radio???thanks [08:45:17 CET] <linux_gnu> or how to use ffmpeg to record the desktop in video and the sound from the mike?? [09:00:49 CET] <linux_gnu> anyone? [11:53:35 CET] <BtbN> Can ffprobe give me the exact duration somehow? Even if that means decoding the whole thing? [12:04:51 CET] <furq> maybe -count_frames [12:05:31 CET] <JEEB> BtbN: -show_frames and then the PTS of the last frame + duration if it's there [12:05:47 CET] <JEEB> that of course differs by stream a bit [12:06:15 CET] <JEEB> and since some streams begin from nonzero you might need to also take into account first frame's PTS [12:19:44 CET] <BtbN> I resorted to parsing the time= output of ffmpeg -i bla.mkv -f null - [12:20:42 CET] <JEEB> but yea, if you don't have discontinuities max((last_pts - first_pts) + last_duration) [12:20:58 CET] <JEEB> should be good enough and the -of json is useful to just push it to python or something [12:22:14 CET] <BtbN> This is randomly cut mpegts [12:22:29 CET] <BtbN> decoding it is really the only reliable source of its duration [12:22:34 CET] <JEEB> yes [12:22:42 CET] <JEEB> and -show_frames decodes [12:22:51 CET] <JEEB> so you don't have to care about b-frames etc [12:23:04 CET] <BtbN> Yeah, but the pts can be all over the place [12:24:12 CET] <JEEB> in what sense? discontinuities, yes you have to handle those separately if the lavf side doesn't handle that wrap-around for you (IIRC it does the first time and then it bugs out). [12:24:42 CET] <JEEB> but generally just having a nonzero start pts isn't a problem [12:24:45 CET] <BtbN> It can be a concatenation of unrelated segments. So pts can jump, reset, and so on [12:26:02 CET] <JEEB> yup, so that requires just having to handle that somehow :P remember that ffmpeg.c isn't perfect with that either, it's given me 10s A/V desync because it couldn't believe something really wanted to have nothing'ish for 10 seconds and then come back [12:26:41 CET] <JEEB> you will get a value and it might handle some stuff. which might or might not be good enough for you [12:27:52 CET] <BtbN> it seemed on point for everything I tested [15:15:52 CET] <BtbN> This is probably the most insane script I ever came up with, but it seems to work: https://github.com/esamarathon/esaupload/blob/master/tools/combine_video.sh [18:13:14 CET] <jemius> Hey folks. I'm doing a project where a rasperryPi-camera is used to observe something 24/7. I'm planning to use a video codec to archieve the pictures efficiently. Now I'm experimenting with ffmpeg and am a bit overwhelmed by all the options you can set. For example: "cat *.jpg | ffmpeg -f image2pipe -i - -c:v libx265 output.mkv" the codec makes a very short video if it's 120 pictures. If I wanted to make a photo every 0.5 [18:13:14 CET] <jemius> s and create a storage video for each minute observation-time (so 120 pictures) and want ffmpeg to produce a video that's (of course) 60s long, how'd i do that? I tried to enforce this with the framerate, but it caused very large videos and I don't really understand why, as even a 40 hour long video containing 120 pictures should only be as large as the 120 pics [18:18:49 CET] <sparrowrsword> how do i compress a lossless avi file down to something reasonable? its an 8 gig file... and its only a few minutes long at most... was trying to upload a video to youtube and used obs to record, however for some reason the last video i recorded was 360p, which is unwatchable...so i recorded in lossless [18:23:46 CET] <sparrowrsword> https://unix.stackexchange.com/questions/60042/trying-to-compress-avi-file-… [18:24:22 CET] <JEEB> pretty sure the trac wiki has stuff on x264, which is so far the best practice of how to encode stuff :P [18:24:30 CET] <JEEB> with it having relatively straightforward knobs [18:24:37 CET] <JEEB> preset for "how fast does this train go" [18:25:08 CET] <JEEB> and then CRF for variable bit rate control that could be sumamrized as being the least far from being "constant quality" kind of thing, without actually being such [18:25:31 CET] <JEEB> https://trac.ffmpeg.org/wiki/Encode/H.264 [18:25:44 CET] <JEEB> yup, so I don't have to repeat my usual lines (possibly) [18:29:58 CET] <Blacker47> jemius, you need to set the framerate (maybe both, for input and output). [20:35:59 CET] <jemius> Blacker47: Well, that still results in a larger movie. And I don't know why [20:40:05 CET] <Blacker47> jemius, video is build of independend GOPs each of that starts with a I-Frame that has complete frame (compressed like jpeg). the number of frames a GOP can have is limited (say it is 60 or 120 maybe much less depending on setting). that you can try is very low output framerate and a large GOP. i don't know if framerate can go below 1. [20:40:50 CET] <Blacker47> in general it makes no sense to create a such long video for only 120 pictures. [20:41:26 CET] <JEEB> technically frame rate can be whatever as long as it can be coded in the time base required [20:42:37 CET] <Blacker47> even if strange framerate can be used for encoding a player may not be able to handle it... needs testing. [20:42:52 CET] <klaxa> practically, i've had problems with framerates near and below 1 during playback (mpv) [20:56:45 CET] <jemius> GOP? [20:57:00 CET] <jemius> So how do I find out what's the optimal solution for my scenario? [21:17:08 CET] <Blacker47> Group Of Pictures. [21:17:47 CET] <Blacker47> jemius, maybe there is some, but i don't know what codec is suitable for such a pathetic scenario... [21:44:34 CET] <jemius> Don't call the dreams of my life pathetic :O [21:47:46 CET] <Blacker47> jemius, it is easier to write a program/script that will change to the next picture every 20 minutes. [21:49:43 CET] <jemius> Blacker47: Probably; the video length isn't my concern. My concern is compression. I just came up with the length issue by the observation of changing video sizes depending on the frame rate [21:59:08 CET] <LigH> Hi. [21:59:57 CET] <LigH> Do you already know of https://github.com/OpenVisualCloud/SVT-VP9 - and may it be supported in ffmpeg some day, if not already? [22:00:53 CET] <jemius> LigH: Doesn't AV1 make VP9 obsolete? [22:01:03 CET] <LigH> It's already mentioned in https://www.freshports.org/multimedia/svt-vp9/ but I did not yet check other commit lists... [22:01:21 CET] <LigH> Some prefer high speed over best quality. [22:01:55 CET] <LigH> This encoder claims to be able to encode in real time, if I understood correctly. [22:03:27 CET] <JEEB> libvpx also can do real-time [22:03:43 CET] <furq> most things can do realtime if you murder the quality badly enough [22:03:58 CET] <JEEB> anyways, we should see people testing that one soon enough [22:04:08 CET] <JEEB> then we'll see if there's any interest in adding a wrapper for it [22:05:44 CET] <jemius> The newest version of ffmpeg already supports AV1, is that correct? [22:05:57 CET] <furq> yes [22:06:00 CET] <furq> libaom-av1 [22:06:01 CET] <durandal_1707> via wrappers [22:06:18 CET] <jemius> The codec is said to be really slow still, though [22:06:33 CET] <JEEB> for decoding it should be fast enough with dav1d [22:06:40 CET] <JEEB> encoding is slow though, yes [22:06:49 CET] <JEEB> although the same SVT-* thing is also available for AV1 [22:09:29 CET] <LigH> Indeed: https://github.com/OpenVisualCloud/SVT-AV1 [22:11:28 CET] <LigH> So I will have some patience and assume that future support is probable. [22:11:38 CET] <LigH> :) [22:11:43 CET] <JEEB> we'll see [22:11:51 CET] <LigH> Bye. [22:20:08 CET] <jemius> How come we can use HEVC freely yet? I thought there are huge license fees on that one [22:20:25 CET] <JEEB> open source generally doesn't take comments on that [22:20:52 CET] <JEEB> f.ex. FFmpeg only provides the source code, if you when distributing a binary are under a need to pay royalties - that's a separate thing [22:21:16 CET] <JEEB> same for VLC f.ex. although they're based in France and have counted that they don't need to care about the licenses [22:21:40 CET] <JEEB> IANAL (I Am Not A Lawyer) and thus I recommend that if you care you may have to look into these things [22:22:08 CET] <kepstin> jemius: it's basically "you're responsible for finding out what patent licensing / fees apply to you" [22:22:47 CET] <jemius> kepstin: So no one bothers if open source folks distribute programs able to en/decode HEVC? [22:23:09 CET] <Mavrik> Em. [22:23:15 CET] <Mavrik> Noone is stopping you from doing that. [22:23:26 CET] <JEEB> distribute source and distribute binaries are two very different things [22:23:33 CET] <Mavrik> But noone will stop the patent pools from suing you to hell and back either :) [22:24:02 CET] <jemius> Well, I get ffmpeg as a binary through debian or arch, so I don't see the difference [22:24:21 CET] <JEEB> in that case debian/arch handle the binary distribution [22:25:16 CET] <kepstin> those distributions own legal teams have decided how to handle that. [22:25:24 CET] <JEEB> yup [00:00:00 CET] --- Mon Feb 18 2019

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ffmpeg-devel.log.20190216
by burek 17 Feb '19

17 Feb '19

[17:52:41 CET] <philipl> BtbN: I think your cuda_check rework breaks any cuda_check users that directly link cuda. [17:53:14 CET] <philipl> The vf_scale_cuda won't compile for me here, because the inline function references tcuGetErrorName etc and those are only defined in the dynamic loader header and that cannot be included at the same time as cuda.h [17:54:39 CET] <philipl> It probably requires a completely separate impl, which i guess is fine. [17:58:13 CET] <jamrial> can't you make the function avpriv and share it between libs? [18:01:55 CET] <philipl> jamrial: that is where we started... [18:02:22 CET] <jamrial> you were using ff_ and #including the c file on every library [18:02:43 CET] <nevcairiel> please avoid avpriv [18:02:54 CET] <nevcairiel> what exactly was the problem with the include way anyway? [18:02:58 CET] <nevcairiel> that works fine for many other functions [18:24:26 CET] <philipl> I don't know what the case was that failed. The report was symbol duplication. Could that happen if someone statically links all libav* into a program? [18:53:51 CET] <philipl> BtbN: https://github.com/philipl/FFmpeg/commit/f028d813709cb14bdf08de898bc560e41e… [19:26:09 CET] <cone-203> ffmpeg 03Philip Langdale 07master:f4ea930a1192: avcodec/hevc_ps: Expose all SPS and PPS range extension flags [19:26:10 CET] <cone-203> ffmpeg 03Philip Langdale 07master:e06ccfbe1d33: avcodec/nvdec: Add support for decoding HEVC 4:4:4 content [19:26:11 CET] <cone-203> ffmpeg 03Philip Langdale 07master:83c7ac2e47ef: avcodec/nvdec: Explicitly mark codecs that support 444 output formats [19:26:12 CET] <cone-203> ffmpeg 03Philip Langdale 07master:317b7b06fd97: avcodec/cuviddec: Add support for decoding HEVC 4:4:4 content [19:44:35 CET] <cone-203> ffmpeg 03Philip Langdale 07master:d6fc5dc24aa0: avcodec/version: Bump micro-version for nvdec/cuviddec changes [20:03:30 CET] <nevcairiel> the way this stuff usually works is that in static builds, only one of the .c files is actually build, and all static libraries reference the same function (ie. from libavutil or whichever is the "primary" one), which works since static has no problems referenceing ff* symbols from another archive [20:03:38 CET] <nevcairiel> and the duplicate .c f iles are only build with dynamic builds [20:03:55 CET] <nevcairiel> if this is not how it was done, then it was done wrong in the first place :) [20:04:29 CET] <nevcairiel> if you in fact have the same ff symbol in every .a file, there is a good chance it'll collide in some toolchain [20:05:01 CET] <philipl> That is very likely what happened. [20:05:33 CET] <philipl> There was no conditional logic for static vs dynamic. [20:05:39 CET] <nevcairiel> sounds to me like someone suggested thats how we solve it, but then not enough research was done how exactly its done [20:07:32 CET] <philipl> Yep. [20:07:51 CET] <nevcairiel> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/Makefile;h=c010f… for example [20:07:59 CET] <nevcairiel> log2_tab.c is only included in shared builds [20:08:30 CET] <philipl> Yes, I was just looking. [20:08:59 CET] <nevcairiel> unfortunately makefile logic is a bit limited so having two conditions to include a fi le (ie. cuda and shared) gets a bit annoying [20:09:02 CET] <philipl> To go back and do this correctly, it would need to be conditioned on both SHARED and CUDA [20:09:05 CET] <philipl> yeah [20:09:22 CET] <philipl> I guess we could add a specific condition in configure for it [20:30:43 CET] <cone-203> ffmpeg 03Michael Niedermayer 07master:1f686d023b95: avcodec/mpeg4videodec: Clear interlaced_dct for studio profile [20:30:44 CET] <cone-203> ffmpeg 03gxw 07master:1466dc140d49: avcodec/mips: [loongson] optimize theora decoding with mmi. [00:00:00 CET] --- Sun Feb 17 2019

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ffmpeg.log.20190216
by burek 17 Feb '19

17 Feb '19

[01:12:30 CET] <keegans> JEEB: re: the HLS implementation, yeah it is broken but i'm not sure how to pinpoint what is exactly broken [01:12:42 CET] <keegans> right now i took a URL and fed it to ffmpeg and it reporst the wrong number of samples [01:12:53 CET] <keegans> perhaps it is only the first audio file -- but it cuts off and then stops giving me frames [01:13:11 CET] <keegans> ffprobe also has the same issue (only shows like 250 frames when there is like 2000+) [01:13:20 CET] <keegans> (spread across many chunked mp4 files) [01:38:27 CET] <keegans> ffprobe -v trace shows AVIndex sample 215, says 1/22050 after the sample# and using a av_read_frame loop gives me 645 samples [01:38:28 CET] <keegans> so wtf [02:53:43 CET] <_Vi> How do I specify ffmpeg to map all streams from all input files? [02:54:12 CET] <_Vi> Is there something like `-map *:*`? [03:02:49 CET] <_Vi> (Keep forgetting that `ffmpeg -i input -c copy -y output` spills away extra languages; remembering only after deleting original file) [03:22:53 CET] <c_14> _Vi: afaik there is no such option, you have to add a -map $number for every input [03:26:39 CET] <_Vi> Maybe FFmpeg should warn more clearly and more red-coloured-ly by default if it decides to omit some audio or subtitle tracks? Like "unused input streams: ..." [03:27:25 CET] <_Vi> Omission of additional tracks may be not noticeable initially, only later when those tracks become needed. [03:27:52 CET] <c_14> probably should, I'm pretty sure it's just legacy behavior at this point [03:28:53 CET] <_Vi> Are patches to add `-map '*'` or to implement that warning welcome? [03:29:09 CET] <c_14> should be, yeah [03:35:41 CET] <furq> probably don't use * just because it'll be annoying having to escape it [03:35:46 CET] <furq> maybe e [03:39:28 CET] <_Vi> What do you suggest instead? `-map all`? `-map_all`? [03:39:49 CET] <_Vi> Or just change the default to include everything? [03:40:16 CET] <furq> i'm going to assume changing the default would get vetoed [03:40:20 CET] <furq> so i'd probably just go with -map e [03:40:36 CET] <furq> A is technically free but that would be needlessly confusing [03:43:33 CET] <furq> on second thoughts i would actually go with -map all so it's clear that it's handled specially by map and not a stream specifier [04:15:06 CET] <keegans> how can i read a fragmented mp4 using ffmpeg ? [04:52:48 CET] <Hackira> Hello guys, I try to compile ffmpeg with ndi support but I get a where ld can't find -lndi, can someone help me figure out what am I doing wrong ? [05:15:48 CET] <zap0> can anyone help me understand how v210 works [07:42:15 CET] <MrDave20161> Hi: I am using swr_convert in my application and it works fine on my VM(Ubuntu 18.04 x64) development machine but when I compile the same code over on my server(Debian 9 x64), it is throwing a seg fault in 'ff_float_to_int16_sse2' Does anyone have any suggestions on additional swr_convert parameters? Occurs with ffmpeg 3.4 as well as 4.1 also when just converting FLTP to FLTP [11:31:46 CET] <Jan-> hihi ffmpeg people [11:32:08 CET] <Jan-> I have an AVI file and a WAV audio file and I need an mp4 output file with the video and audio. [11:32:19 CET] <Jan-> Right now I have a command line that turns the AVI into an mp4 [11:32:27 CET] <Jan-> how do I go about adding in the wave file for audio? [12:24:43 CET] <_Vi> Jan-, Something like `ffmpeg -i yourvideo.avi -i youraudio.wav -c:v copy -map 0:v -map 1:a output.mp4` [12:25:00 CET] <Jan-> thanks [12:28:56 CET] <DHE> the map commands should be unnecessary since the default is for ffmpeg to take the last source for each of audio and video... [12:29:22 CET] <Jan-> I did wonder [00:00:00 CET] --- Sun Feb 17 2019

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ffmpeg-devel.log.20190215
by burek 16 Feb '19

16 Feb '19

[00:27:59 CET] <cone-124> ffmpeg 03Michael Niedermayer 07master:b0d8b7cb8e86: avformat/mov: Do not use reference stream in mov_read_sidx() if there is no reference stream [00:44:42 CET] <cone-124> ffmpeg 03Timo Rothenpieler 07master:9e1e5213933d: avutil/cuda_check: fix usage of removed .c file [10:32:39 CET] <OmniBlade> Hello, I'm trying to add support to the aud demuxer for a pcm format found in the Blade Runner game, but I'm having trouble correctly returning an end of data state. [10:33:24 CET] <JEEB> AVERROR_EOF? [10:34:00 CET] <OmniBlade> I'll give that a try, I take it I should return that when I know I have no more data to read [10:38:01 CET] <OmniBlade> Thanks, tha appears to have solved the issue. [10:38:50 CET] <OmniBlade> How do I go about submitting a patch, generate a patch set with git and then submit to the mailing list for review? [10:39:48 CET] <JEEB> commit changes as usual, create format-patch and either use git send-email or attach the patch to a mail that you send to ffmpeg-devel(a)ffmpeg.org [10:40:05 CET] <JEEB> format-patch is basically a patch that contains commit info [10:40:18 CET] <JEEB> if it's multiple patches you can mkdir -p my_patch_set [10:40:34 CET] <JEEB> and then git format-patch master..HEAD -o my_patch_set/ [10:40:51 CET] <OmniBlade> With each patch being a commit? [10:41:00 CET] <JEEB> format-patch does that yes [10:41:13 CET] <JEEB> you get 0001-*.patch, 0002-*.patch etc [10:42:13 CET] <OmniBlade> Okay, thanks. I was planning to add a muxer for aud and extending the adpcm encoder to write the stream that it expects at some point as well. [10:43:23 CET] <JEEB> you can see the documentation on git format-patch and send-email if you want. the --help output of at least format-patch seems pretty good [17:42:44 CET] <BBB> I'm using an older version for something and I get this message "Past duration 0.999992 too large", what does it mean? [17:43:37 CET] <durandal_1707> which container? [17:43:58 CET] <BBB> ivf [17:46:50 CET] <durandal_1707> huh? it should not be present there afaik [17:48:26 CET] <durandal_1707> some vfr/cfr nonsense [18:15:45 CET] <nevcairiel_> JEEB: afaik US CC can have language codes, but its independent of the CC track, and instead shipped in some metadata in the TS container [18:16:03 CET] <nevcairiel_> i remember seeing something a bout that from work [22:00:02 CET] <JEEB> nevcairiel_: ok, so if you only transfer the cc packets it's not gonna be there [22:09:46 CET] <kepstin> BBB: usual case where I see that is when the output on a filter chain is set to cfr, but the pts values on the frames are closer together than that (higher framerate). Some frames will be dropped in this case. [22:10:35 CET] <kepstin> an easy way to trigger it is to use the concat filter, with the first input being lower framerate than the second - concat sets its output to the framerate of the first input, so everything gets messed up when it passes through frames from the second. [22:10:58 CET] <kepstin> (that's a bug in the concat filter, it should just set the output framerate to undefined in that case) [00:00:00 CET] --- Sat Feb 16 2019

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ffmpeg.log.20190215
by burek 16 Feb '19

16 Feb '19

[01:09:31 CET] <pmacdiggity> I'm having a hard time getting clarity on the current state, is x265 the only way to produce 10-bit HEVC that's compatible with iOS? This seems to require yuv420p10le(tv, bt2020nc/bt2020/smpte2084), which means no hardware acceleration like hevc_nvenc? [01:28:58 CET] <iive> pmacdiggity, no idea about your question, just want to point out that for the encoded bitstream it doesn't matter if the input format is yuv420p10le or yuv420o10be [01:29:23 CET] <iive> o/p [01:30:39 CET] <pmacdiggity> I mean on the output pix_fmt, hevc_nvenc says no dice, and subs in p010le, but only yuv420p10le seems to play back correctly on iOS [01:32:10 CET] <pmacdiggity> or at least the yuv420p10le is the only discernible difference I can find with x265 vs hevc_nvenc [01:37:13 CET] <pmacdiggity> and actually, it seems to be the bt2020nc/bt2020/smpte2084 part in particular that's important vs yuv420p10le(tv, progressive) not playing correctly [01:37:14 CET] <iive> ffmpeg swscale filter should be able to convert yuv420p10 to yuv420 [01:37:33 CET] <pmacdiggity> yeah, but I want 10bit output [01:38:13 CET] <iive> sorry, isn't the problem that the output is 10bit? [01:39:02 CET] <pmacdiggity> no, it's that it's 10bit, but it's washed out on playback [01:39:02 CET] <furq> p010le is 10-bit [01:39:15 CET] <furq> it's the 10-bit nv12 iirc [01:39:18 CET] <pmacdiggity> yes, but it seems that's not the particular flavor that Apple likes [01:39:25 CET] <furq> 4:2:0 with interleaved chroma planes [01:40:39 CET] <pmacdiggity> it plays back like a 10bit HEVC plays back at 8bit without correct tone mapping [01:41:01 CET] <furq> do both files have the same primaries etc [01:41:21 CET] <pmacdiggity> not sure what you mean as primaries? [01:42:23 CET] <furq> the "bt2020nc/bt2020/smpte2084" bit of the thing you pasted [01:46:41 CET] <pmacdiggity> ah, not on the output, but I don't seem to be able to output bt2020nc/bt2020/smpte2084 with hevc_nvenc, it it says not supported and outputs with p010le, but looking at the outputs with ffprobe, the correct file is yuv420p10le(tv, bt2020nc/bt2020/smpte2084), using x265, and the incorrect file is output to yuv420p10le(tv, progressive) (from hevc_nvenc, which it said was p0120le [01:46:49 CET] <pmacdiggity> er... p010le [01:48:11 CET] <furq> yeah i have no experience with nvenc but nv12/p010le are just internal formats afaik [01:50:13 CET] <furq> try adding -colorspace bt2020nc -color_primaries bt2020 -color_trc smpte2084 [01:54:07 CET] <iive> yuv420 uses 3 planes for y u v. nv12 uses 2 planes, first is Y, the seconds is interleaved UV [01:58:03 CET] <pmacdiggity> ok, I'll give that a shot, thanks! [02:14:51 CET] <pmacdiggity> success! thank you! [04:06:51 CET] <BullHorn> hello! [04:07:00 CET] <BullHorn> how can i convert a variable bitrate video to constant bitrate? [04:30:38 CET] <shdown> Hello, Im trying to use the ffmpeg API to decode, encode, and then stream a video via UDP. [04:30:40 CET] <shdown> The problem is that it generates packets with duration=0, and, even if I manually set duration to, say, 1, mpv still refuses to play anything but the first frame it receives. [04:30:46 CET] <shdown> Here is the code: https://gist.githubusercontent.com/shdown/0b88bd67b5a99516dba237ca1ec63842/… [04:31:11 CET] <shdown> (ffmpeg.pastebin.com does not seem to exist.) [09:00:11 CET] <th3_v0ice> shdown: Packet duration can be left at 0, but if you need to set it yourself then it cant be 1 it needs to be 1 / framerate for video and 1 / timebase * nb_samples for audio. But generating it like this will give you trouble trying to sync audio and video streams. One more thing that can happen is that your audio and video are so much out of sync that your player is simply waiting for packets. [09:00:11 CET] <th3_v0ice> Try to print timestamps that go to the muxer. [10:17:28 CET] <forgon> What is the best way to record only a specific window, known from its PID? [10:40:38 CET] <relaxed> forgon: I don't know it's possible using the PID, but here's a one liner that will turn your mouse pointer to an +, then click on the window you want to record: x=($(xwininfo|awk '/Wid/||/Hei/||/Cor/{gsub(/+/,",");print $2}') );ffmpeg -video_size "${x[0]}"x"${x[1]}" -framerate 60 -f x11grab -i :0.0"${x[2]/,/+}" -t 30 -c:v rawvideo -pix_fmt yuv420p -y output.mkv [10:42:21 CET] <forgon> relaxed: Thanks. [13:35:24 CET] <devp> Hello. How i can capture a rtp stream with their all tracks ? at this moment i only can capture one track of audio, how i could capture all tracks? [13:42:28 CET] <th3_v0ice> devp: I think you need to use -map and map all streams from input to output. https://trac.ffmpeg.org/wiki/Map [13:45:02 CET] <devp> th3_v0ice, and how i can do that if i don't know exaclty how many streams it has ? [13:49:26 CET] <devp> ffmpeg -i input -map 0 -c copy output.mp4 # copies all video and audio channels from input one to output, not just one video [15:09:49 CET] <sybariten> hey [15:11:08 CET] <sybariten> (pretty sure there are a number of discussions about this online, but to my surprise my google-fu wasnt strong enough.... keywords?) I'm guessing it's quite doable to define only absolute values when doing split cutting with ffmpeg (start time and end time) instead of using duration. But this is a bit too complicated to be worth it perhaps? [15:13:27 CET] <TheAMM> I'm not sure what you're asking [15:13:57 CET] <TheAMM> You should be able to use -ss <start> -i file -to <end> to grab a section of the file? [15:14:29 CET] <DHE> keep in mind that syntax "-to" is interpreted more as duration than actual endtime [15:14:55 CET] <TheAMM> "Stop writing the output or reading the input at position." [15:15:12 CET] <TheAMM> "-ss start -to end -i file" should work properly then? [15:15:25 CET] <TheAMM> Or does that still work as a duration [15:16:41 CET] <DHE> I believe on the input that will work as you want [15:16:51 CET] <TheAMM> It does, tested it [15:16:58 CET] <DHE> it's a matter of telling it to stop at the indicated INPUT timestamp, or OUTPUT timestamp. those are two different things [15:17:16 CET] <TheAMM> Well aware [15:17:20 CET] <TheAMM> Hopefully this helps sybariten [15:28:01 CET] <devp> there's some way to take a screenshot in grayscale colors ? [15:32:45 CET] <TheAMM> ffmpeg -ss 00:00:05 -i file.mkv -vf format=gray -frames:v 1 out.png [15:33:18 CET] <sybariten> heh, i dont really get what you are saying. :) [15:33:38 CET] <sybariten> the syntax is -t , no? And its a duration in hh:mm:ss [15:34:12 CET] <TheAMM> -t is duration, yes [15:36:12 CET] <sybariten> ok so there is another option to use absolute times ? [15:36:31 CET] <TheAMM> <TheAMM> "-ss start -to end -i file" [15:37:28 CET] <sybariten> oh! This must be a bit mnore recent then. All the googling i did only talked about -t [15:37:31 CET] <sybariten> thanks [17:33:40 CET] <th3_v0ice> devp: I am not exactly sure what can be done in that particular case. If you do not expect the stream to change hardcode the values, otherwise write a simple bash/batch script to do the work for you and select needed streams. [20:08:39 CET] <TheWild> hello [20:09:19 CET] <TheWild> I want to concatenate few mp4 files into one, without reencoding. I did "ffmpeg -i 'concat:1.mp4|2.mp4|3.mp4' -c copy output.mp4", but it only encodes first file. What I'm doing wrong? [20:10:07 CET] <durandal_1707> TheWild: use concat demuxer not protocol [20:12:17 CET] <TheWild> ok, now I used "ffmpeg -f concat -i filelist.txt -c copy output.mp4" [20:12:34 CET] <TheWild> but I remember I used concat protocol at least once in the past and it worked [20:14:28 CET] <durandal_1707> TheWild: it worked for ts probably, but can not work for mp4 [20:14:47 CET] <TheWild> mmmm, okay. Thanks durandal_1707 [20:14:58 CET] <TheWild> weird ffmpeg didn't even print warning [20:19:15 CET] <TheWild> 1037 MB (546 MB if you don't count commercials ;)) [20:53:43 CET] <TheWild> how I can extract *raw* subtitles/teletext data? [20:54:53 CET] <TheWild> this doesn't work: "ffmpeg -i 20190215.mts -scodec copy teletext.dat" [22:46:35 CET] <mfolivas> guys, many of my customers are asking me to add a frame or watermark on their videos. Can I do that with ffmpeg? [22:47:58 CET] <furq> !filter overlay @mfolivas [22:47:58 CET] <nfobot> mfolivas: http://ffmpeg.org/ffmpeg-filters.html#overlay-1 [22:48:07 CET] <mfolivas> oh nice [22:48:42 CET] <mfolivas> thank you guys [23:12:30 CET] <lindylex> If my video is being sped up using this : setpts=0.1*PTS What is the formula to get the audio speedup value? [23:13:44 CET] <furq> lindylex: the inverse [23:13:48 CET] <furq> so 10 [23:14:00 CET] <lindylex> But it only can go to 2 [23:14:14 CET] <furq> it goes up to 100 in new versions [23:14:17 CET] <furq> https://ffmpeg.org/ffmpeg-filters.html#atempo [23:14:36 CET] <lindylex> Reading now. [23:14:43 CET] <lindylex> Thanks [23:14:57 CET] <furq> looks like that works in 4.1 so it didn't change super recently [23:15:03 CET] <furq> not sure exactly when though [23:16:01 CET] <lindylex> Thanks I found old documentation that stated i would have to string multiple together to get it done. [23:16:17 CET] <furq> if you have an old version then i guess it'd be 2,2,2,1.25 [23:17:11 CET] <lindylex> This is the version I am using ffmpeg version 4.1-1 [23:17:23 CET] <furq> 10 should work fine then [23:19:13 CET] <lindylex> I just became aware of the set time for when an affect you start to work. I need to look into the syntax some more. I want to add it to my pythin script that I wrote to make ffmpeg commands shorter. [23:21:12 CET] <lindylex> What is the formula? Can see what the formula looks like? [23:21:28 CET] <lindylex> If i have this setpts=0.05*PTS[v] [23:21:46 CET] <lindylex> What is the formula to get the value for audio? [23:22:00 CET] <lindylex> is .1*100? [23:22:35 CET] <kepstin> lindylex: not sure what you're trying to do here. [23:22:51 CET] <kepstin> lindylex: are you doing slow motion video, then also trying to slow down the audio to match? [23:22:59 CET] <lindylex> What value would I use for the audio? [23:23:08 CET] <lindylex> Speed up the video and audio/ [23:23:17 CET] <kepstin> if so, you'll need to use a different filter on the audio, which would take different arguments [23:23:34 CET] <lindylex> I know thi sis what I did ffmpeg -i c116_sp.mov -filter_complex "[0:v]setpts=0.05*PTS[v];[0:a]atempo=10.0[a]" -map "[v]" -map "[a]" c116_Done.mkv [23:23:36 CET] <kepstin> if you're using e.g. the rubberband filter, it takes a "speed" option, I think? [23:23:44 CET] <lindylex> I want to speed it up even more. [23:24:13 CET] <lindylex> I have no idea what this is. That filter is new to me. [23:24:57 CET] <kepstin> if you do setpts=0.5*PTS on the video, then the video is twice as fast, so you'd use 1/0.5 = 2 as the speed setting an the audio filter, like atempo=2 [23:24:58 CET] <furq> lindylex: you could just use setpts=PTS/20 and atempo=20 [23:26:09 CET] <lindylex> kepstin it is .05 not .5 [23:26:44 CET] <lindylex> furq : That is interesting. This sets the video to match the audio speed [23:26:51 CET] <lindylex> That is interesting. [23:26:53 CET] <kepstin> lindylex: I told you how to do the math to make them match. [23:27:20 CET] <lindylex> kepstin: sorry thanks I get it now. [23:29:04 CET] <lindylex> Thanks both of you. That was extremely helpful. [00:00:00 CET] --- Sat Feb 16 2019

1 0

ffmpeg-devel.log.20190214
by burek 15 Feb '19

15 Feb '19

[00:06:15 CET] <feliwir_> What would be the proper name for a struct shared between Vp3DecodeContext and Vp3EncodeContext? Vp3State seems a little bit weird [00:12:19 CET] <BradleyS> Vp3SharedContext ? [00:12:33 CET] <feliwir_> seems good [00:13:51 CET] <BradleyS> the two most difficult things in computer science: cache invalidation and naming things and off-by-one errors [00:14:17 CET] <atomnuker> feliwir_: Vp3Context [00:15:57 CET] <BradleyS> Vp3StructSharedWithVp3EncodeDecodeContexts [00:16:07 CET] Action: BradleyS runs [00:16:31 CET] <J_Darnley> are you missing some factory contructors there? [00:18:21 CET] <BtbN> At least it's not a BoundTreeRewriterWithStackGuardWithoutRecursionOnTheLeftOfBinaryOperator [00:19:58 CET] <BradleyS> struct Kevin [01:03:34 CET] <JEEB> me gets eaten by wchar_t [01:03:46 CET] <JEEB> why on earth did someone decide to use wchar_t [04:49:36 CET] <rcombs> JEEB: was it within a #ifdef _WIN32 [05:08:03 CET] <rcombs> wow the lack of PTS generation code in the VC1 parser is kicking my ass [09:05:04 CET] <JEEB> rcombs: no [09:05:18 CET] <JEEB> it was probably code that would have only been tested on *nix [09:05:24 CET] <rcombs> was it wcwidth or wcswidth [09:05:25 CET] <JEEB> which made it all that much "wut" [09:05:38 CET] <JEEB> no [09:05:46 CET] <rcombs> okay I'm fresh out of legitimate reasons to use wchar_t then [09:06:05 CET] <JEEB> looking at the value it was defined to it looked like either uint16_t or uint32_t [09:06:12 CET] <JEEB> I think it was 16bit value [09:06:29 CET] <JEEB> (or 32bit, I was getting tired at that point) [09:06:58 CET] <rcombs> mmm, 16-bit values in 32-bit arrays(?) for no reason [09:08:27 CET] <rcombs> I love how wchar_t having implementation-defined size makes it so useless that C(++)11 added char16_t and char32_t [09:08:43 CET] <rcombs> (which are also mostly useless but at least char32_t's good for some obscure C++ stuff) [09:11:20 CET] <nevcairiel_> I've only ever used wchar_t in windows code, because you have to. Anywhere else I would gladly stick to UTF8 [09:11:37 CET] <rcombs> on most platforms wchar_t is UTF-32 [09:12:09 CET] <rcombs> which is sometimes useful (like, if you're actually trying to render text), but there you just use int because wchar_t is something completely different on windows [09:12:33 CET] <rcombs> (and also because there's really no need for a dedicated C type for that) [09:17:13 CET] <rcombs> it's actually literally defined to be the same as int_least32_t [09:17:51 CET] <rcombs> erm, char32_t is [09:18:12 CET] <rcombs> and it's just useless because int32_t exists [09:18:39 CET] <rcombs> (at least in C++ one could _vaguely_ imagine wanting some sort of overload for char32_t that's distinct from int, since char32_t is explicitly meant for UTF-32, but I can't really think of anything specific) [09:33:11 CET] <JEEB> rcombs: do you remember if the US closed captions in digital broadcast had language codes? [09:33:25 CET] <JEEB> or is it just caption slot XYZ [09:34:17 CET] <rcombs> I don't _think_ they do but I'm not positive [09:34:34 CET] <feliwir> Btw i saw there is an open ticket for DICOM support in FFmpeg. I work at a company that has a lot to do with DICOM and i've done some DICOM stuff myself. What kind support is wanted there? [09:35:03 CET] <thardin> feliwir: I'd ask in the ticket [09:35:19 CET] <rcombs> if they exist, neither VLC nor MediaInfo shows them [09:35:21 CET] <feliwir> I don't have a ticket login :D [09:35:33 CET] <JEEB> rcombs: ok, that's what I expect [09:35:40 CET] <JEEB> thanks for semi-confirming :) [09:35:46 CET] <thardin> ah. maybe an admin can help :) [09:36:18 CET] <feliwir> Ah, nvm i'll register [09:37:17 CET] <thardin> it has open registrations? couldn't find such a form [09:37:28 CET] <feliwir> In the top right i was able to register [09:38:30 CET] <JEEB> do tell if you also have issues getting the e-mail. someone else on #ffmpeg noted that their e-mail server never got the confirmation e-mail [09:43:48 CET] <feliwir> I'd be suprised if someone answers actually [09:44:05 CET] <feliwir> I saw that this was written our for GSoC sometime, but seems like noone took it [09:46:09 CET] <JEEB> ah [09:46:21 CET] <JEEB> does it at least note who wrote that originally? [09:48:43 CET] <feliwir> ami_stuff [09:48:54 CET] <feliwir> Which is a user that doesn't really exist [13:33:31 CET] <BtbN> perflab1@nvidia does not sound like a good author [13:39:22 CET] <JEEB> nope [13:39:27 CET] <cone-034> ffmpeg 03Timo Rothenpieler 07master:15c639013909: avutil/cuda_check: avoid pointlessly exporting same symbol from two libraries [13:39:28 CET] <JEEB> sounds like missing git configuration [13:39:28 CET] <cone-034> ffmpeg 03Roman Arzumanyan 07master:f1f66df6a204: avcodec/nvenc: add b_as_ref support for HEVC [00:00:00 CET] --- Fri Feb 15 2019

1 0

ffmpeg.log.20190214
by burek 15 Feb '19

15 Feb '19

[00:03:28 CET] <mfolivas> Want to get better performance with the encoding/decoding. I want to try with hardware encoding were image quality and file size but on the decoding side neither of those matter. [00:03:38 CET] <mfolivas> looking into this: http://trac.ffmpeg.org/wiki/HWAccelIntro [00:18:23 CET] <mfolivas> is NVENC only supports GPU? [00:18:58 CET] <BtbN> nvenc, as the name suggests, only works on Nvidia GPUs [00:19:15 CET] <BtbN> Though it's not run by the GPU, but by dedicated silicon [00:19:29 CET] <mfolivas> BtbN as always thanks!! [00:26:10 CET] <mfolivas> any idea how I can get the vaapi_device to play in Quicktime? [00:26:16 CET] <mfolivas> time ffmpeg -vaapi_device /dev/dri/renderD128 -i two-minute-video.mp4 -vf 'hwupload,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi -level 4.1 -preset fast -compression_level 5 vvapi_two-minute-video.mp4 [00:26:27 CET] <mfolivas> I am not able to play it in Quicktime [00:29:11 CET] <mfolivas> BtbN I'm trying to see if I can get a quick gain with hw. Right now my videos are taking too long for my business case. [00:31:18 CET] <hans_> i guess the "retrying with new connection: Invalid argument"-part is a bug? http://paste.debian.net/plain/1067697 [00:31:31 CET] <hans_> anyway, haven't updated my ffmpeg installation in months [00:32:11 CET] <tmm1> is there a way to select frames *without* some metadata key set [00:53:14 CET] <brandonkal> Hello, I am trying to generate thumbnails and scale them down. But the output images are the same size as the source. Here is the command: [00:53:19 CET] <brandonkal> `ffmpeg -hide_banner -y -i src.mp4 -vf scale="240:-1" -vf fps=1/5 stills/img%03d.jpg` [00:53:38 CET] <JEEB> you are defining vf twice [00:54:01 CET] <JEEB> -vf "fps=1/5,scale=240:-2" [00:54:08 CET] <JEEB> probably should do it :P [00:55:16 CET] <brandonkal> Yep. Thanks JEEB [02:54:29 CET] <mfolivas> anyone using libmfx? [02:54:38 CET] <mfolivas> I'm trying to install it in Linux/Debian to be more exact [05:00:56 CET] <brandonkal> Is it possible to only transcode if the source codec is not the desired output codec? Or should I script ffprobe and then build out the command? [05:01:39 CET] <brandonkal> Same question for resize. i.e. don't resize to 1080 if source is only 720 [05:30:39 CET] <brandonkal> Another question: ffmpeg is saving my encoding settings to file metadata. That is cool but how do I disable this or save it to a sidecar file? [07:16:16 CET] <rnmhdn_> ffprobe gives three streams for a file [07:16:40 CET] <rnmhdn_> I run this to remove the third stream but it doesn't get removed [07:16:43 CET] <rnmhdn_> ffmpeg -i 8.mp4 -map 0 [07:16:50 CET] <rnmhdn_> / ffmpeg -i 8.mp4 -map 0:0 -map 0:1 -acodec copy -vcodec copy video.mp4 [07:21:03 CET] <rnmhdn_> Here are my streams: [07:21:20 CET] <rnmhdn_> Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661) [07:21:49 CET] <rnmhdn_> Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo [07:22:03 CET] <rnmhdn_> Stream #0:2(eng): Data: none (tmcd / 0x64636D74) [11:09:08 CET] <Ariyasu> im using -ss before -i when encoding to x264 but it's not frame accurate [11:09:21 CET] <Ariyasu> it's still jumping to the proceding i-frame [11:09:43 CET] <Ariyasu> i thought that only happened when doing a stream copy [11:25:10 CET] <Ariyasu> Input #0, mpegts, from 'test.ts': [11:25:10 CET] <Ariyasu> Duration: 00:38:59.46, start: 94117.334222, bitrate: 6236 kb/s [11:25:24 CET] <Ariyasu> what is that "start: 94117.334222" value? [11:38:25 CET] <JEEB> Ariyasu: start timestamp [11:38:33 CET] <JEEB> mpeg-ts streams almost never start from zero and go forwards [11:38:42 CET] <JEEB> ffmpeg.c can (and does by default) adjust it to start from zero [11:48:35 CET] <Ariyasu> what format is the timestamp jeeb? [11:50:22 CET] <Ariyasu> i've been issuing "ffmpeg -ss 00:04:55.000 -i" "ffmpeg -ss 00:04:55.040 -i" "ffmpeg -ss 00:04:55.080 -i" "ffmpeg -ss 00:04:55.120 -i" ect... and it always starts on the preceding iframe instead of the frame im specifying [11:50:56 CET] <Ariyasu> https://pastebin.com/raw/UJKVcEhN these are my test lines [12:39:50 CET] <th3_v0ice> Ariyasu, iirc -ss in front of the -i is a fast search, and it will go to the nearest I frame. If you specify -ss after -i then it will decode frames and will give you frame level accuracy you are looking for. This method is obviously slower. [12:40:21 CET] <Ariyasu> thank you th3_v0ice [12:40:27 CET] <Ariyasu> i will do some testing [12:40:45 CET] <JEEB> basically if you're not on an IRAP (intra random access picture), it will pick the previous one [12:40:56 CET] <JEEB> also if it's MPEG-TS the seek is not exact [12:41:10 CET] <JEEB> you will want to remux to mp4 or so if you want exact'ish cutting [12:41:26 CET] <JEEB> (or use something that does indexing, like ffms2 through vapoursynth/avisynth) [12:41:56 CET] <Ariyasu> im trying to save time by not cutting my input file but just using marked cut points from video redo [12:42:17 CET] <Ariyasu> if i need to remux or index, then im not gonna save any time [12:42:20 CET] <Ariyasu> but thanks for the info [12:42:44 CET] <Ariyasu> videoredo has frame acurate cutting though and uses ffmpeg iirc to make it's cuts [12:42:52 CET] <Ariyasu> so i wonder how it deals with the issue [12:43:11 CET] <JEEB> sounds like an editor so it decodes, and omst likely either decodes or uses indices like ffms2 [12:43:20 CET] <JEEB> and ffms2 also is a wrapper on top of FFmpeg's libraries [12:43:27 CET] <JEEB> it just uses the APIs to create an index [12:43:37 CET] <JEEB> and then you can f.ex. do exact seeks even if the input has no index of its own [12:44:39 CET] <JEEB> so it depends on if you want to do re-encoding or just cutting of pieces [12:44:59 CET] <JEEB> (Also if you index + use the APIs nicely you could even do a mix) [12:45:08 CET] <JEEB> so you re-encode the part where you can't make an exact cut [12:45:15 CET] <JEEB> and otherwise copy the input stream [12:46:28 CET] <Ariyasu> jeep i want to encode the entire file [12:46:44 CET] <Ariyasu> from an input frame i specify [12:46:53 CET] <Ariyasu> not do cutting or stream copy at all [12:47:16 CET] <Ariyasu> i will do some testing with the info you have given me [12:47:18 CET] <Ariyasu> thanks [12:47:19 CET] <JEEB> then it sounds like you could generate a filter chain [12:47:31 CET] <JEEB> you just decode the whole shebang, but only pass some frames to the encoder [12:47:37 CET] <JEEB> instead of -ss -t [17:30:13 CET] <mfolivas> besides `-preset` what else can affect the performance of rendering? [17:30:22 CET] <mfolivas> oh, and hardware of course [17:30:43 CET] <mfolivas> I'm looking for settings that I can tuned on the ffmpeg [17:30:58 CET] <kepstin> mfolivas: video content, resolution, framerate; filters [17:31:20 CET] <kepstin> in some cases using a hardware decoder can give a slight speedup to transcodes [17:32:43 CET] <kepstin> but yeah, in most simple cases the bottleneck is the encoder. [17:33:21 CET] <kepstin> there may be something specific to the command that you're running where we could give suggestions, but there's no magic options. [17:33:48 CET] <mfolivas> kepstin thanks! I'm looking to analyze some of the commands like -preset [17:34:43 CET] <kepstin> "-preset" is an encoder specific option, used most commonly with x264; x264 has a ton of options but aside from specific requirements there's basically no reason to adjust them individually, just use a preset. [17:35:53 CET] <egrouse> i ended up using -preset in combination with -crf and -tune to get what i needed sort; seems to have knocked off all my previous issues with streams going down [17:36:17 CET] <egrouse> which is gooood as i was about to start trying to log timestamps so i could resume when it broke but hey [17:36:19 CET] <egrouse> now it doesnt break [17:37:29 CET] <kepstin> crf is tricky with live streams, because often you're required to fit in specific bitrate limits - you might want to use vbv controls (the "-maxrate" and "-bufsize" options) to ensure it doesn't exceed the required rates. [17:37:50 CET] <kepstin> or even a full vbr mode [17:38:24 CET] <mfolivas> egrouse I need it form video and I need performance [17:38:35 CET] <mfolivas> not sure if I could use the "fastdecode" in tune [17:39:07 CET] <egrouse> kepstin, i must admit i currently have both -crf and the maxrate/bufsize options; is it likely only one of them is working therefore? [17:39:19 CET] <kepstin> egrouse: no, they work in combination [17:39:35 CET] <egrouse> i did the annoying thing where i made 2 changes at once and im not 100% sure which one actually fixed it [17:39:48 CET] <egrouse> and ok cool [17:39:52 CET] <kepstin> "fastdecode" option is generally not useful, only needed if the computer trying to play the video is too slow to decode it [17:39:52 CET] <egrouse> all good then :) [17:40:28 CET] <kepstin> although fastdecode will often speed up encoding as well, because it disables some features - but it's better to use a faster preset instead [17:40:46 CET] <egrouse> currently using 'faster' preset and tune 'zerolatency' with crf 16 and that seems to nail it [17:41:36 CET] <kepstin> yeah, just check your cpu usage and make sure it's sufficiently below 100% that there's room for it to spike a bit if you get a burst of complex content [17:42:08 CET] <egrouse> yeah its like 50% [17:42:26 CET] <egrouse> i had big issues before where it was at 100% all the time and kept cutting out [17:42:42 CET] <egrouse> but just dropped to 720 instead of 1080 and now its a lot better [17:42:52 CET] <egrouse> facebook only takes 720p anyway so, shrug [17:42:59 CET] <kepstin> yeah, if your cpu is pegged at 100% then that means the encoder can't keep up with the realtime data [17:43:15 CET] <kepstin> (you'll also see the "speed" in the ffmpeg cli status bar go below 1.0x) [17:43:47 CET] <furq> egrouse: do you actually need sub-second latency [17:43:52 CET] <furq> because otherwise zerolatency is terrible [17:44:04 CET] <furq> if you're going to facebook then i'm assuming you don't want it [17:44:36 CET] <egrouse> im not sure; that was one of the two changes i made at which point it was 'fixed' [17:44:49 CET] <egrouse> we are pushing to restream which then pushes to fb, twitch, yt - dont know if that makes any difference [17:45:21 CET] <furq> was it you who needed baseline profile for some reason [17:45:31 CET] <furq> if so then zerolatency forces baseline iirc [17:45:36 CET] <egrouse> i dont think so [17:45:41 CET] <kepstin> yeah, if you're doing something like that then latency obviously isn't a big deal, so i wouldn't use that tune [17:45:45 CET] <furq> i must be thinking of someone else [17:45:51 CET] <furq> but yeah zerolatency is almost always bad [17:45:57 CET] <egrouse> ok cool, i will review settings - cheers for the info [17:46:08 CET] <egrouse> stream kept getting dropped i think due to delay in processing or not being able to keep up [17:46:54 CET] <furq> well zerolatency also turns a bunch of stuff off [17:47:05 CET] <furq> although it also turns frame threading off which will slow things down [17:47:11 CET] <kepstin> egrouse: if your cpu was at 100% before, then it was probably dropped because your computer wasn't sending video fast enough. solution is to use a faster preset [17:47:14 CET] <furq> you'd be better off using -preset veryfast or something [17:47:25 CET] <egrouse> i must say that this here channel is probably the most informative and useful irc channel i've ever been in, ha; learnt so much just by even reading other peoples queries/answers [17:47:28 CET] <egrouse> so, thanks for that :D [17:50:31 CET] <egrouse> changed settings will see what happens next time it kicks in ~ [18:12:07 CET] <mfolivas> according to the documentation crf should be between 17-28 [18:12:13 CET] <mfolivas> is that correct? [18:12:27 CET] <mfolivas> or should I play with that as well to see if that affect performance? [18:12:43 CET] <furq> aren't you using h264_vaapi [18:13:19 CET] <Blacker47> mfolivas, at first crf will affect quality and that is with what you should start. cut a short clip of video (10-60 sec) and test with different settings to get what you want. [18:16:49 CET] <kepstin> note that the preset values and crf values aren't completely separate - changing crf can change encoding speed slightly (particularly lower crf values - resulting in higher bitrates - can be slower depending on other options), and if you change the preset, then the meaning of the crf value changes (quality won't be quite the same, although it's usually pretty close) [18:17:27 CET] <JEEB> which means that you have your 2500 or so frame sample, and you run different presets over it to check speed differences [18:17:34 CET] <JEEB> you should end up with the slowest fast enough preset [18:17:42 CET] <JEEB> then you start with crf 23 and if it looks good you go up [18:17:46 CET] <JEEB> if you it looks bad you go down [18:18:17 CET] <JEEB> and you should after that have the slowest preset that is fast enough for you, as well as the highest crf for that type of content (resolution, frame rate, contents) [18:18:32 CET] <kepstin> yeah, I usually work it by picking a crf from the "reasonable" range that looks ok on medium preset, adjust the preset to get the speed I want, then fine-tune the crf value to set quality if needed. [18:19:39 CET] <JEEB> but yea, many people who have already encoded some types of content have some sort of default range for *their* eyes and use case [18:20:01 CET] <JEEB> for example I would know which value I would start with for 720p24 animation if encoding with 10bit x264 :P [18:20:50 CET] <JEEB> but what you need to underline here is that different content is different, and that different people as well as viewing styles all affect your requirements [18:20:58 CET] <JEEB> (as far as parameters go) [18:22:03 CET] <JEEB> also generally it's much easier to go towards something that looks "good enough" from something that might not look exactly good enough, and thus I generally start with the default CRF with x264 (which is aimed a bit on the 'compress more' side) [18:22:43 CET] <kepstin> i wish figuring out quality/speed settings for vp9 was as easy as with x264 [18:22:56 CET] <kepstin> maybe that's something that'll be better in av1? probably not, but who knows :/ [18:23:11 CET] <JEEB> everyone's favourite - libvpx [18:23:12 CET] <JEEB> :) [18:23:22 CET] <kepstin> (x264 was *really* terrible to use until they added the preset stuff, btw) [18:23:49 CET] <kepstin> before the presets system, you'd just copy/paste bad option sets off wikis and hope they did ok :/ [18:23:50 CET] <JEEB> yup [18:24:15 CET] <furq> i can't imagine people would put bad video encoding settings on a wiki [18:24:30 CET] <furq> next you'll be telling me they're doing it in forum posts [18:25:38 CET] <egrouse> :D [18:26:42 CET] <kepstin> did they eventually end up removing that x264 code that printed a warning if it detected options that looked like the bad defaults from old ffmpeg versions? [18:26:49 CET] <JEEB> nope [18:26:50 CET] <kepstin> must have, i can't seem to find it any more. [18:26:53 CET] <JEEB> oh [18:26:56 CET] <kepstin> hmm, then i just can't find it [18:27:06 CET] <JEEB> let me just boot my dev VM [18:27:45 CET] <kepstin> a sufficiently old ffmpeg where that would be an issue wouldn't compile against current x264 anyways [18:27:59 CET] <JEEB> true [18:28:41 CET] <JEEB> encoder/encoder.c: x264_log( h, X264_LOG_ERROR, "broken ffmpeg default settings detected\n" ); [18:28:59 CET] <JEEB> 545de2ffec6ae9a80738de1b2c8cf820249a2530 [18:29:03 CET] <JEEB> so at that point it was still there [20:51:33 CET] <tmm1> how can i render a video frame's planes to separate images? [20:52:43 CET] <DHE> https://ffmpeg.org/ffmpeg-filters.html#extractplanes sounds like what you want [20:53:02 CET] <tmm1> thanks! [20:55:48 CET] <tmm1> is there a way to get ffplay to start timestamps at 0 like ffmpeg's output does [20:59:17 CET] <kepstin> the timestamps starting at 0 is something the ffmpeg cli tool has special code to do on the inputs [20:59:47 CET] <kepstin> if you just want to reset timestamps in ffplay, you could do something like setpts=PTS-STARTPTS, but note that it'll break a/v sync. [21:05:12 CET] <tmm1> ok cheers [21:14:24 CET] <tmm1> are there any docs on the various modes in vf_blend [21:14:58 CET] <durandal_1707> tmm1: there is documentation for all filters [21:15:44 CET] <tmm1> yes i see the vf_blend docs but i'm curious what modes like dodge, freeze, linearlight etc do [21:17:30 CET] <durandal_1707> look at source code [21:49:20 CET] <tmm1> are there any examples of how to use the hysteresis filter [21:51:00 CET] <JEEB> seems like it takes two inputs [21:53:20 CET] <JEEB> yea, seems to be -filter_complex "[0:v][1:v]hysteresis[outcome]" -map "[outcome]" [21:53:36 CET] <JEEB> so two video inputs, filter them and you should get one output? [21:54:09 CET] <TheWild> hello [21:55:28 CET] <TheWild> got mpv and vtt out of the internets, converted vtt to ass, burned on top of mp4. But the two-line subtitles don't sync quite right. How I can get just single line? [21:55:40 CET] <TheWild> *got mp4 and vtt [21:56:28 CET] <JEEB> you probably need some sort of merging of lines that are during the same time or so. not sure if there's anything automagic for that [21:57:59 CET] <TheWild> :/ [21:58:17 CET] <TheWild> the original vtt has correct, single line subtitles. Just ffmpeg messes it up. [21:59:07 CET] <JEEB> huh, it shouldn't convert that sort of stuff [21:59:18 CET] <JEEB> if there's a single line that's a single AVSubtitle [21:59:32 CET] <TheWild> I'm getting first subtitle displayed on lower row, next on upper row and lower doesn't disappear [21:59:34 CET] <JEEB> unless the webvtt "decoder" makes multiple rectangles [21:59:51 CET] <JEEB> oh [21:59:58 CET] <JEEB> also timings shouldn't be touched [22:00:11 CET] <JEEB> so if it has overlapping timings then -fix_sub_duration is probably for you [22:00:18 CET] <JEEB> if they shouldn't overlap [22:00:31 CET] <TheWild> lower is changed when next subtitle comes into play, but because of the alternating lines, the sentences are just mangled. [22:00:32 CET] <JEEB> but then again that's just a workaround for broken timings in that case [22:00:44 CET] <TheWild> overlapping timings? Oh, I didn't check this [22:01:13 CET] <TheWild> oh gosh [22:01:47 CET] <TheWild> point for you, and ffmpeg didn't mess up anyting. Just the subtitles are in fact overlapping :/ [22:02:07 CET] <JEEB> yea, for better or worse the decoders and encoders generally don't poke at the values too much [22:02:30 CET] <JEEB> but yes, you either fix the overlaps yourself, or if you're OK with "show until next one" then -fix_sub_duration does that [22:06:59 CET] <TheWild> "show until next one" is fine, but -fix_sub_duration doesn't seem to make a thing. Does it only work when converting subtitles and not burning them in? [22:07:41 CET] <JEEB> it should work since it's adjusting by the decoded AVSubtitles (unless that thing is somehow skipped with the lavfi flow that you're using) [22:07:55 CET] <JEEB> right, it probably is since you're using the subtitle filter most likely [22:11:36 CET] <TheWild> converted vtt to ass again with -fix_sub_duration, the resulting file seems okay. [22:11:41 CET] <TheWild> Yes! Yes! Yes! Yes! [22:11:49 CET] <TheWild> this worked! [22:11:51 CET] <JEEB> :) [22:11:51 CET] <TheWild> thank you JEEB [22:34:44 CET] <tmm1> is there a filter i can use to turn all pixels below a certain threshold to black [22:35:15 CET] <JEEB> probably, I just can't name it [22:35:28 CET] <JEEB> (as seems to be the answer to most questions regarding "is there a thing for X" [22:35:30 CET] <tmm1> geq might work [22:37:45 CET] <furq> geq will work but it's slow [22:38:04 CET] <furq> you might be able to coerce chromakey/colorkey into doing it but you'll need some extra work to figure out the right values [22:42:55 CET] <tmm1> i basically want a black/white image based on threshold [22:49:26 CET] <furq> if you mean literally two colours then you could do a format conversion to monob/monow and back again [22:49:36 CET] <furq> you wouldn't have any direct control over the threshold though [22:51:49 CET] <furq> like i said, geq will do this, so probably try that first and see if it's too slow [22:51:58 CET] <tmm1> awesome, thank you [22:52:16 CET] <tmm1> yea geq gives me good control over the threshold, but format=monob gets pretty close to what i want too [22:52:37 CET] <furq> you could mess around with the brightness etc to tune it [22:53:15 CET] <furq> there's probably some precise way to get the right threshold but i'm not enough of a lavfi guy to tell you [00:00:00 CET] --- Fri Feb 15 2019

1 0

ffmpeg-devel.log.20190213
by burek 14 Feb '19

14 Feb '19

[03:17:51 CET] <blue_misfit> hey-o! wondering if there's any particular reason some of the recent prores improvements didn't go out with 4.1.1 e.g. https://github.com/FFmpeg/FFmpeg/commit/e4788ae31b2e9af45d11f4bf4498c075dcc… [09:56:34 CET] <j-b> good moronings! [13:07:36 CET] <cone-491> ffmpeg 03Jun Zhao 07master:ff5ed7abd6ee: avfilter/tests/integral: Correct the comment [13:59:42 CET] <Shred00> i don't seem to be getting e-mail confirmation tokens trying to create an account on trac. can anyone poke it and see if it's still working? [14:00:27 CET] <JEEB> does anyone remember who's the admin for trac? [14:18:11 CET] <cone-491> ffmpeg 03Reto Kromer 07master:bf78aa9ee9c4: doc/muxers: fix typo [17:39:28 CET] <philipl_> BtbN: I've taken a look at the new video sdk headers. Does what the release notes says it does. I'll put together an update at some point this week [17:44:44 CET] <BtbN> philipl_, nvidia has already sent them to the list [17:51:34 CET] <philipl_> I see it now. How kind of them. [17:54:05 CET] <philipl> BtbN: His cuviddec patch seems wrong. I had to introduce the new pixfmts to handle the 444P10/P12 content and he's trying to reuse the existing ones. I assume garbage on the screen as a result. [17:55:04 CET] <philipl> or at least garbage if he does anything with software filters or encoders. [17:58:40 CET] <philipl> Do you think I should resend without the new pixfmts and just treat everything as 444P16? And avoid that rabbit hole... [19:16:47 CET] <blue_misfit> hey-o! wondering if there's any particular reason some of the recent prores improvements didn't go out with 4.1.1 e.g. https://github.com/FFmpeg/FFmpeg/commit/e4788ae31b2e9af45d11f4bf4498c075dcc… [19:17:43 CET] <JEEB> if it's a bug fix then it sounds like it could. most likely nobody just took notice of it [19:18:21 CET] <blue_misfit> there were a bunch of improvements in the last couple months, one of the biggest of which is 12 bit decoding for 4444 / 4444 XQ [19:18:32 CET] <JEEB> improvements are different from bug fixes [19:18:42 CET] <blue_misfit> indeed [19:18:46 CET] <JEEB> I wouldn't be surprised if improvements wouldn't go out into a frozen branch [19:18:53 CET] <JEEB> but bug fixes probably should [19:18:57 CET] <blue_misfit> I see [19:19:19 CET] <blue_misfit> any sense of when the next release will be? [19:20:06 CET] <BradleyS> that one does seem like a bug fix, michaelni ^ https://github.com/FFmpeg/FFmpeg/commit/e4788ae31b2e9af45d11f4bf4498c075dcc… [19:20:26 CET] <BradleyS> (consider for 4.1.x) [19:20:53 CET] <BtbN> philipl, imagine if they made cuvid output what ffmpeg expects. The chaos. [19:21:07 CET] <JEEB> :D [19:23:24 CET] <blue_misfit> TBH this is what I really care about: https://github.com/FFmpeg/FFmpeg/commit/6a583261ea631e6da5593c4477e36fde9bc… [19:23:36 CET] <blue_misfit> plus any related commits [19:23:46 CET] <blue_misfit> thx again for these, kierank :) [19:27:52 CET] <blue_misfit> just curious if there's known issues with 12 bit prores decoding or if it's generally considered safe to use [19:31:59 CET] <michaelni> BradleyS, backported e4788ae31b2e9af45d11f4bf4498c075dcc25a6c to release/4.1 locally [19:32:32 CET] <michaelni> blue_misfit, backports are for bugfixes not new features [19:33:04 CET] <BradleyS> ty [19:33:12 CET] <blue_misfit> understood, not expecting this to get backported :) [19:34:28 CET] <blue_misfit> just curious whether anyone is aware of any known issues with this new feature since I'll probably be pinning my CI/CD to use the most recent commit [19:34:41 CET] <BtbN> philipl, the patches are also not checking the version. I'd like to keep compat for older drivers. [19:43:39 CET] <philipl> Yeah. So I'm going to refresh my old patches with the official SDK and resend. [20:03:11 CET] <BtbN> philipl, the SDK update they sent is also in itself incomplete. That's so weird. [20:03:30 CET] <BtbN> I copied over the headers from the official download, and it has more stuff [20:03:43 CET] <JEEB> funky [20:21:04 CET] <philipl> BtbN: yes. I've been comparing too. [20:21:21 CET] <philipl> I would recreate from scratch [20:23:24 CET] <BtbN> Nah, it was fine. Patch and ammendments are already merged. [20:28:30 CET] <philipl> cheers [23:08:21 CET] <blue_misfit> were there any breaking / non backwards compatible changes with 4.0 vs 3.4? [23:08:44 CET] <JEEB> did the 0 != EOF change come up with that? [23:08:55 CET] <JEEB> I think that's the largest change in the last year+ [23:09:05 CET] <JEEB> which changes how the API should be utilized [23:09:20 CET] <blue_misfit> okay, but if I'm just using the CLI I should be okay, right? [23:09:41 CET] <blue_misfit> (I will of course test my stuff, just wondering if there's known stuff to be aware of) [23:10:04 CET] <JEEB> not that I know, cli in general hasn't changed too much unless someone accidentally broke something [23:10:17 CET] <blue_misfit> I figured as much, thx! [23:10:34 CET] <JEEB> because generally removing options needs to first warn, and only then remove [23:10:41 CET] <blue_misfit> I'm sure you guys hear this a lot but this is an _AMAZING_ project. Thank you for so much hard work [23:11:35 CET] Action: JEEB wonders on which level MMTP should be implemented [23:12:01 CET] <JEEB> because the "files" that recorders do are literally the data that comes off of the receiver, which is just IP over broadcast [23:12:10 CET] <JEEB> so suddenly your multimedia application has to be able to do IP [23:20:13 CET] <BtbN> philipl, it looks like the nvdec hevc decoder also needs some work now. There are a bunch of new fields it needs to fill. [23:34:09 CET] <philipl> BtbN: I have that done locally [23:34:57 CET] <BtbN> It's also interesting that they introduced a version check macro for nvdec. Inside the header. So the macro can't be used, because it's not in the older headers. [23:35:36 CET] <philipl> #ifndef? :-P [23:48:40 CET] <J_Darnley> I need to stop reading the user list. All it does is make me angry. [23:49:13 CET] <J_Darnley> On a side note I cant believe it is nearly 9 years since I first proposed colored log messages. [23:49:23 CET] <JEEB> :D [23:53:13 CET] <feliwir_> Don't we have colored log messages? [23:54:19 CET] <cone-532> ffmpeg 03Marton Balint 07master:dd6dd49f1d5e: avformat/mpegts: fix charset of type 0x11 [23:54:19 CET] <cone-532> ffmpeg 03Marton Balint 07master:b35843e3913d: avformat/mpegts: also convert strings without a specified encoding to UTF-8 [23:54:19 CET] <cone-532> ffmpeg 03Marton Balint 07master:dbfd042983ee: avformat/utils: parse some stream specifiers recursively [23:54:45 CET] <J_Darnley> Yes but users still ignore them [23:56:19 CET] <J_Darnley> Personally I think that is sometimes because too much is colored [23:57:03 CET] <J_Darnley> Other times if you laser engraved it onto a bat and started hitting them it still wouldnt leave an impression [23:58:21 CET] <BtbN> It'll probably be something like #if defined(CHECK_MACRO) && CHECK_MACRO(9, 0) [23:58:33 CET] <BtbN> Normal if semantics do apply there, right? [00:00:00 CET] --- Thu Feb 14 2019

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ffmpeg.log.20190213
by burek 14 Feb '19

14 Feb '19

[02:35:36 CET] <wfbarksdale> Hey folks, was just wondering if it is possible to remux an mp4, placing the entire moov atom at the end of the file (i.e. incurring no seek operations, on the aviocontext) [02:37:14 CET] <wfbarksdale> I am basically trying to trim an mp4 file without transcoding, and send the output directly out over http via chuncked response, without producing an intermediate file [02:37:39 CET] <wfbarksdale> but on writing the trailer, it looks like quite a few seek operations occur [02:48:31 CET] <wfbarksdale> is it even possible :( [02:48:38 CET] <Hello71> wfm with -movflags empty_moov [02:48:57 CET] <Hello71> also, browsers strongly prefer to know the length at the start [02:49:10 CET] <Hello71> why don't you just use matroska [02:50:22 CET] <wfbarksdale> I am sending this over network to mobile devices, which don't like mkv [02:50:52 CET] <wfbarksdale> i'm using the C api. Wondering how I can specify that option, persumably with avformat_alloc_output_context? [02:51:45 CET] <Hello71> dunno [02:51:48 CET] <wfbarksdale> or avformat_write_header perhaps [02:51:49 CET] <Hello71> that doesn't sound right though [02:55:54 CET] <wfbarksdale> worked, you are a genius Hello72 [02:58:47 CET] <wfbarksdale> literally just saved me [03:01:17 CET] <wfbarksdale> actually, may have spoken too soon [03:14:34 CET] <wfbarksdale> nope totally works, thank you! [03:31:14 CET] <keegans> how can I read the per-sample IV data using ffmpeg [03:31:38 CET] <keegans> i think it is the senc header [04:12:10 CET] <kandinski> hi, can someone give me pointers on how to encode a stream of static images, coming at irregular intervals, into a video stream? [04:26:20 CET] <wfbarksdale> can anyone explain how `bits_per_coded_sample` is derived in the AVCodec parameters? [04:38:03 CET] <keegans> yeah ffmpeg is only giving me side data with IV information for 130 of ~5000 samples [04:38:11 CET] <keegans> all the IV data is in the senc header but it isn't getting read [04:38:18 CET] <keegans> so idk [04:38:28 CET] <keegans> https://github.com/FFmpeg/FFmpeg/blob/0c3333faf6b7bfd45ce666ae697759d9239ba… [04:38:33 CET] <keegans> the sample_count must be wrong [07:19:06 CET] <xj> I am using ffmpeg to decode x264 files in a c++ app. Everything almost works. I am decoding 18 files at once. the cpu is about 10% the gpu is about 30% and the hdd is about 5%, but I am getting audio errors and stuttering occasionally. could the motherboard be the problem? any ideas? [07:20:16 CET] <xj> the files i am decoding are fine. each one has been checked in vlc. i was wondering if the machine just doesn't have the bandwidth. oh, RAM usage is at 40%. [08:51:25 CET] <th3_v0ice> Why is it that neither AV_PKT_FLAG_CORRUPT (in AVPacket) nor AV_FRAME_FLAG_CORRUPT nor FF_DECODE_ERROR_INVALID_BITSTREAM nor FF_DECODE_ERROR_MISSING_REFERENCE (in AVFrame) are set but mpegts demuxer shows errors when I set ffmpeg log level of 40? The stream has perceivable errors in it. [13:00:22 CET] <Shred00> the trac bug reporting tool doesn't seem to be sending account verification token e-mails. i have requested a token several times and am getting nothing. it's not in a spam folder or anything like that. the e-mails are not even being received by my server. [13:33:59 CET] <JEEB> Shred00: you could poke -devel [13:37:01 CET] <AhirPK> hi [13:37:01 CET] <AhirPK> i am having some issues in complex_filter application development in C lang [13:37:01 CET] <AhirPK> I want to overlay showvolume output to a small scaled video of TS using c program [13:37:01 CET] <AhirPK> but I am getting 2 different errors [13:37:01 CET] <AhirPK> Error: 1) av_buffersink_get_frame fails with return value AVERROR(EAGAIN) (this is not error but this error stays forever) [13:37:02 CET] <AhirPK> Error: 2) Pushing decoded frame to filters [13:37:04 CET] <AhirPK> [Parsed_showvolume_0 @ 0x1c54600] Format change is not supported [13:37:06 CET] <AhirPK> Error while feeding the filtergraph [13:37:08 CET] <AhirPK> here is my code link: https://www.dropbox.com/s/eojzuw78s7zemzt/main.c?dl=0 [13:37:10 CET] <AhirPK> can anyone please help me to find issue? [13:37:12 CET] <AhirPK> Thank you [13:37:15 CET] <Shred00> what's the most common/useful way to share a video snippet when posting to the mailing list? [13:37:49 CET] <JEEB> there's a videolan stream sample pages (streams.vidoelan.org or something?) and then there's the no-frills 0x0.st [13:37:59 CET] <JEEB> for 0x0.st you have to remember how to upload a file through curl :P [13:38:08 CET] <JEEB> (also it's not eternal) [13:38:45 CET] <JEEB> AhirPK: if your (input) formats change, put a filter before that which converts them to something normalized [13:38:55 CET] <JEEB> Shred00: also this was mostly to poke IRC [13:39:11 CET] <JEEB> to post that the trac doesn't seem to be working :P [13:39:14 CET] <JEEB> (fully correctly) [13:39:20 CET] <JEEB> not sure who admins it currently [13:41:50 CET] <Shred00> hrm. uploading to streams.videolan.org doesn't give me the url to the uploaded file [13:46:30 CET] <JEEB> I've gotten urls to them from #elsewhere so most likely people have access to that [13:46:37 CET] <JEEB> just mention where you have uploaded it to there [13:46:49 CET] <Shred00> i'll just use google drive i guess [13:47:10 CET] <JEEB> but as I said, bug reports should go to trac, and you should poke -devel IRC channel or so regarding if the trac is borked registration wise [13:50:01 CET] <AhirPK> i am new in filter programming, can you please be more specific? [13:50:01 CET] <AhirPK> and right now i am make filter_graph like this: [13:50:02 CET] <AhirPK> input1 => abuffer => showvolume => \/ [13:50:02 CET] <AhirPK> => overlay => \/ [13:50:02 CET] <AhirPK> input2 => buffer => scale => /\ [13:50:02 CET] <AhirPK> => overlay => buffersink [13:50:04 CET] <AhirPK> input3 => nullsrc => /\ [13:51:44 CET] <AhirPK> JEEB: [13:51:58 CET] <JEEB> not going to be able to utilize more time on you right now, sorry :P [13:52:04 CET] <JEEB> but my recommendation still stands [13:52:14 CET] <JEEB> normalize the audio/video format if the filter requires it to be static [13:52:47 CET] <JEEB> (not audio normalization, but making sure that before it's fed to a filter it's X) [13:52:56 CET] <JEEB> or you re-initialize the filter chain if that happens :P [13:52:59 CET] <JEEB> anyways, afk [13:53:17 CET] <durandal_1707> AhirPK: what changes? channels/sample_rate/depth or? [13:54:50 CET] <AhirPK> durandal_1707: i have no idea, because i dont see any specific error msg [13:56:00 CET] <AhirPK> and i am using same filter disc as i am using in command line [13:56:25 CET] <AhirPK> but CLI works fine while programming isnt [13:57:36 CET] <durandal_1707> AhirPK: perhaps you need to insert aformat filter before showvolume? [13:58:25 CET] <durandal_1707> ffmpeg auto inserts such filters when they are needed [14:00:31 CET] <AhirPK> well, i tried that while ago, but let me try to add again and try [14:13:51 CET] <AhirPK> durandal_1707: still getting same error [14:13:52 CET] <AhirPK> [auto_resampler_0 @ 0x2647900] Format change is not supported [14:13:53 CET] <AhirPK> Error while feeding the filtergraph [14:19:23 CET] <durandal_1707> AhirPK: what you did? show the code! [14:20:35 CET] <AhirPK> //aformat filter [14:20:35 CET] <AhirPK> snprintf(strbuf, sizeof(strbuf), [14:20:35 CET] <AhirPK> "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64, [14:20:35 CET] <AhirPK> av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100, [14:20:35 CET] <AhirPK> (uint64_t)AV_CH_LAYOUT_STEREO); [14:20:36 CET] <AhirPK> fprintf(stderr, "aformat: %s\n", strbuf); [14:20:37 CET] <AhirPK> ret = avfilter_graph_create_filter(&(screenInfo->aformat_ctx), aformat, "aformat", [14:20:39 CET] <AhirPK> strbuf, NULL, filter_graph); [14:20:41 CET] <AhirPK> if (ret < 0) { [14:20:47 CET] <AhirPK> av_log(NULL, AV_LOG_ERROR, "error initializing abuffer filter\n"); [14:20:49 CET] <AhirPK> return ret; [14:20:51 CET] <AhirPK> } [14:21:30 CET] <AhirPK> and then added aformat filter between abuffer and showvolume [14:21:31 CET] <AhirPK> //connecting abuffer to aformat [14:21:32 CET] <AhirPK> ret = avfilter_link(screenInfo->abuffersrc_ctx, 0, screenInfo->aformat_ctx, 0); [14:21:32 CET] <AhirPK> if(ret < 0){ [14:21:32 CET] <AhirPK> av_log(NULL, AV_LOG_FATAL, "failed to link abuffer filter to aformat filter\n"); [14:21:32 CET] <AhirPK> break; [14:21:33 CET] <AhirPK> } [14:21:36 CET] <AhirPK> //connecting aformat to showvolume filter [14:21:37 CET] <AhirPK> ret = avfilter_link(screenInfo->aformat_ctx, 0, screenInfo->showvolume_ctx, 0); [14:21:40 CET] <AhirPK> if(ret < 0){ [14:21:41 CET] <AhirPK> av_log(NULL, AV_LOG_FATAL, "failed to link aformat filter to showvolume filter\n"); [14:21:48 CET] <AhirPK> break; [14:21:50 CET] <AhirPK> } [14:21:54 CET] <AhirPK> durandal_1707 [14:21:56 CET] <durandal_1707> AhirPK: use pastebin PLEASE! [14:21:57 CET] <AhirPK> durandal_1707: [14:23:10 CET] <durandal_1707> AhirPK: the showvolume filter needs FLTP sample format [14:23:27 CET] <durandal_1707> not S16 [14:33:24 CET] <AhirPK> durandal_1707: still not working [14:35:58 CET] <durandal_1707> AhirPK: look at others code, like mpv one [14:37:14 CET] <JEEB> hint: there's a function to dump your filter chain [14:37:16 CET] <JEEB> (graphically) [14:37:21 CET] <JEEB> so you can see wtf you ended up doing [14:37:29 CET] <JEEB> s/doing/configuring/ [14:38:10 CET] <AhirPK> JEEB: i know there is a function called avfilter_graph_dump [14:38:31 CET] <AhirPK> but its crashing while i call this function [14:38:43 CET] <AhirPK> but app is crashing while i call this function [14:38:59 CET] <JEEB> then you are calling it before the filter graph is configured [14:39:20 CET] <JEEB> (I know because I have done that too) [14:41:06 CET] <AhirPK> https://pastebin.com/4iSdnwHe [14:41:17 CET] <AhirPK> JEEB: here is filter graph [18:50:26 CET] <Mista_D> off topic question: can capture ts with FFmpeg ".. -i udp://${mcast}:${port}?sources=${src}..." but can't seem to make tcpdump write the file to a disk off a nic. [18:55:49 CET] <DHE> depending on the type of switch involved, it may be necessary for something to explicitly request the multicast stream before it will arrive at the NIC [18:55:58 CET] <DHE> otherwise the network itself will filter the traffic out [19:08:12 CET] <Mista_D> DHE ffmpeg capture seem to work ok, but tcpdump "udp port $port and host $mcast and src $src" doesnt work... [19:10:37 CET] <DHE> Mista_D: try running tcpdump while ffmpeg is running [19:27:27 CET] <Ariyasu> i want to encode a segment of video between the timestamps 00:05:09.160 and 00:35:55.920 [19:28:00 CET] <Ariyasu> setting the start time with "-ss 00:05:09.160" works fine but setting "-to 00:35:55.920" fails [19:28:46 CET] <Ariyasu> i think because after it sets the start time, it becomes 00:00:00.000 ? and i would need to recalculate my end time? [19:29:05 CET] <Ariyasu> whats the best solution for this? [19:35:04 CET] <Ariyasu> like 00:35:55.920 in my input file will actually be 00:30:46.760 in my output file if i start the encode job from 00:05:09.160 [19:35:19 CET] <Ariyasu> is there someway to specify the -to to a timestamp in the input? [19:35:47 CET] <Ariyasu> or an easy way to deduct 1 timestamp from another to calculate the duration to feed to -to [19:45:40 CET] <DHE> Mista_D: ...? [23:14:53 CET] <utack> is there way to find out which version of an encoder was statically linked? "-h encoder=*" does not show that [23:17:08 CET] <furq> nothing smarter than ldd [23:17:40 CET] <JEEB> or looking at possibly exported symbols if those are left [23:21:48 CET] <utack> okay thanks [00:00:00 CET] --- Thu Feb 14 2019

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