November 2018 - Ffmpeg-devel-irc

ffmpeg-devel.log.20181129
by burek 30 Nov '18

30 Nov '18

[00:19:47 CET] <cone-041> ffmpeg 03Felicia Lim 07master:4cd6f08d2005: avcodec/libopusenc: Fix warning when encoding ambisonics with channel mapping 2 [00:51:10 CET] <atomnuker> michaelni: I didn't ack the opus ambisonics patch [00:51:17 CET] <atomnuker> I'd like for it to be reverted please [00:51:40 CET] <atomnuker> its a hack to improperly support ambisonics encoding for their use case only [00:52:11 CET] <atomnuker> I did express disapproval and none of the issues regarding it were fixed at all [00:52:36 CET] <atomnuker> in fact the person just hoped we'd forgotten about it [01:04:11 CET] <atomnuker> I don't want a repeat of august, if you thought that no one cares about this patch then you won't care if I revert it myself [01:04:23 CET] <cone-041> ffmpeg 03Rostislav Pehlivanov 07master:cf283f2dfbbf: Revert "avcodec/libopusenc: Fix warning when encoding ambisonics with channel mapping 2" [01:04:24 CET] <cone-041> ffmpeg 03Rostislav Pehlivanov 07master:ffec9d32feaf: Revert "libopus: Add channel mapping 2 support in libopusdec" [01:25:47 CET] <michaelni> atomnuker, iam not sure i somehow missed mails in that thread but to me it looked like Felicia was asking about and pinging this patch in august with no reply from anyone and november again. So it very much looks like noone cared since 4 months. Iam very happy if you take this over and np about the revert [01:26:35 CET] <iive> atomnuker, what were the issues you had with the patches ? [01:29:26 CET] <atomnuker> I did explain this in my reply but this is essentially relying on a non-specified mapping between ambisonics and a channel layout [01:29:39 CET] <atomnuker> and asking the encoder to interpret those specifically as ambisonics [01:32:35 CET] <michaelni> atomnuker, i think Felicia did not understand that there where issues remaining in the patch otherwise there would have been no ping i guess [01:32:36 CET] <atomnuker> we need to specify a way to signal ambisonics and we need to specify how the channels in such streams would map to avframe->data [01:33:41 CET] <atomnuker> yeah, I'll take over reviewing it, I'll cook up a patch to do what I proposed now [03:10:20 CET] <cone-041> ffmpeg 03James Almer 07master:53ca505acc53: avcodec/libdav1d: remove init cleanup internal codec cap [03:10:21 CET] <cone-041> ffmpeg 03James Almer 07master:3cd275bcde5c: avcodec/libdav1d: use constants defined in the public API to limit thread count [03:10:22 CET] <cone-041> ffmpeg 03James Almer 07master:0fca2f60dac7: avcodec/libdav1d: read profile from the sequence header referenced by the ouput picture [03:10:23 CET] <cone-041> ffmpeg 03James Almer 07master:e695b0beba4a: avcodec/libdav1d: add an option to toggle Film Grain [15:17:02 CET] <atomnuker> why is there an opusenc patch on the ML? [15:17:27 CET] <atomnuker> is someone seriously using it? [15:18:07 CET] <atomnuker> durandal_1707: there's an ambisonics channel layout RFC on the ml [15:18:18 CET] <atomnuker> you know a lot more about ambisonics, could you take a look at it? [15:36:57 CET] <CoreX> i think somebody came in last week and spoke about opusenc patch can't remember when exactly [18:48:16 CET] <durandal_1707> atomnuker: i know nuting, ask koda [21:17:40 CET] <durandal_1707> michaelni: stop spamming me via linkedin, thanks [21:46:55 CET] <BradleyS> durandal_1707: i think you mean linkedin, stop spamming me under false pretenses [21:48:32 CET] <BBB> true, linkedin will claim someone did something to you whereas in reality it's just advertising spam [21:49:01 CET] <durandal_1707> spam with your name in it? [21:49:08 CET] <BradleyS> yep [21:49:26 CET] <BradleyS> "your invitation from _____ is waiting" is a lie [21:49:27 CET] <atomnuker> isn't linkedin the most boring social network in existance? [21:49:45 CET] <BBB> I think that's why people use it [21:49:47 CET] <BBB> boring = trusted [21:50:01 CET] <BradleyS> people use it because everyone uses it [21:50:14 CET] <BBB> haven't you noticed that ICQ and MSN lookalikes are amongst the most valuable unicorns of this year? [21:50:36 CET] <BBB> not to mention IRC ripoffs with ICQ smileys [21:50:44 CET] <BBB> literally billions of dollars [21:50:54 CET] <BBB> for an IRC ripoff with smileys [21:51:08 CET] <BBB> so, yes, boring = good [21:51:10 CET] <atomnuker> I have no idea what the irc ripoffs are trendy now [21:51:41 CET] <michaelni> durandal_1707, thanks for notifying me, i was partly aware of this, but not that this still happens, just now went over my account and killed all invites i hope this stops all spam it sends in my name, but i dont really know [21:52:51 CET] <durandal_1707> good work! [21:52:54 CET] <kurosu> linkedin being such a facebook harvester of data is the primary reason for me not joining it [21:52:56 CET] <j-b> atomnuker: Slack. [21:53:03 CET] <kurosu> *facebook-like [00:00:00 CET] --- Fri Nov 30 2018

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ffmpeg.log.20181129
by burek 30 Nov '18

30 Nov '18

[00:39:57 CET] <degenerate> how do i turn on voicemail saving to disk [00:40:03 CET] <degenerate> right now it just emails but i want to save them [03:17:18 CET] <rjeli> Im encoding camera data in real time and noticed the sensor is very noisy. Will preprocessing to reduce noise improve encoder quality? [03:17:48 CET] <rjeli> I assume its encoding a ton of high frequency data thats not important [03:18:01 CET] <furq> it's worth a try [03:18:46 CET] <rjeli> K thanks [03:19:12 CET] <furq> i mean it will definitely save bitrate [03:19:45 CET] <furq> hqdn3d should do a decent job anyway [04:16:40 CET] <Hello71> I think x265 has built in noise reduction [04:16:49 CET] <Hello71> apparently av1 has film grain synthesis [04:20:06 CET] <furq> x265's noise reduction is ported from x264 [04:20:17 CET] <furq> neither of them are particularly effective [06:29:01 CET] <zanijwa> how do I remove the chapter metadata from a video? I found this on google but there was no replies: https://lists.ffmpeg.org/pipermail/ffmpeg-user/2011-September/002479.html [09:29:53 CET] <pagios> hello, i am looking at testing some OTT solutions on which i can add my own content and monetize, do you recommend any OTT project that would be stable enough to use as a building block in my project? Thank you [16:34:23 CET] <Adcock> Pardon me. [16:34:24 CET] <Adcock> https://bitbucket.org/ekolis/freee/overview [17:36:14 CET] <Shezi> A quick question. Some times while I am streaming to RTMP server, I get End of file error. The file is ok and runs many times perfectly ok but randomly fails with this glitch. Any ideas? thanks [17:36:59 CET] <kepstin> Shezi: the 'end of file' error probably comes from the network connection to the rtmp server being interrupted (possibly, closed by the server) [17:37:33 CET] <Shezi> Ok. So it has nothing to do with the media file I am streaming? [17:39:04 CET] <pink_mist> you're streaming a local file -to- an RTMP server? [17:39:44 CET] <Shezi> Yes [17:40:08 CET] <Shezi> Can it be that remote server has some glitch etc? [17:40:24 CET] <kepstin> note that many CDN services and streaming sites will disconnect you if your stream falls outside of permitted bitrate, codec, resolution, speed, etc. [17:41:07 CET] <Shezi> Yes that's correct but if I stream it 10 times, it fails randomly max 2 times [17:41:20 CET] <Shezi> Not even sequentially. [18:58:02 CET] <mixfix41> ffplay struggles with ssh/pulse/audio forwarding but with firefox or mplayer using the same port the sound is proper [18:59:51 CET] <furq> you're not really supposed to use ffplay for anything other than debugging [19:00:13 CET] <furq> if ffmpeg has the same issue then that's probably more of a cause for concern [19:17:41 CET] <mixfix41> ok [22:23:48 CET] <Yoann_> hello [22:25:19 CET] <Yoann_> I wonder why libaom (AV1) is still marked as experimental. Bitstream is frozen and libaom is released (1.0.0). [00:00:00 CET] --- Fri Nov 30 2018

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ffmpeg-devel.log.20181128
by burek 29 Nov '18

29 Nov '18

[01:13:17 CET] <cone-105> ffmpeg 03Mark Thompson 07master:b9aff7a53d59: hwcontext_opencl: Only release command queue if it exists [02:48:58 CET] <vincent42> hi all, I've some MKV produced from live DVB recording (tvheadend), when trying to remux them (ffmpeg in codec copy mode) I often get wrong DTS / PTS warnings and the result play with sluttering. [02:50:11 CET] <vincent42> However if I transcode them, there's no such problem, so the input stream is correct, it just seems that ffmpeg in copy mode (H264) does not extract all timing informations it could, is there an option to do that (to look deeper in the H264 ES, even though we don't want to re-encode) [04:12:36 CET] <BradleyS> vincent42: sounds like a question for #ffmpeg [04:12:55 CET] <BradleyS> probably want to include your ffmpeg --version [04:13:14 CET] <BradleyS> also, lots of people are sleeping at this hour, so be patient for replies [04:42:18 CET] <vincent42> BradleyS: noted, thanks [10:26:39 CET] <cone-155> ffmpeg 03Paul B Mahol 07master:18aea7bdd96b: avcodec/opus: set skip_samples [12:20:14 CET] <cone-155> ffmpeg 03Peter Ross 07master:533ba0b6b5a5: configure: detect djgpp libc [12:20:15 CET] <cone-155> ffmpeg 03Peter Ross 07master:dea1224754fd: additional math.h functions for djgpp [12:20:16 CET] <cone-155> ffmpeg 03Peter Ross 07master:8b7a1c22274e: configure: memalign is broken on djgpp 2.05 [12:20:17 CET] <cone-155> ffmpeg 03Peter Ross 07master:331715534a5b: avpriv_tempfile: add djgpp fallback [18:57:48 CET] <devinheitmueller> Good afternoon. Anyone interested in doing some consulting to address some bugs in libavcodecs HEVC decoder? [19:04:09 CET] <JEEB> durandal_1707: beep ^ [19:04:20 CET] <JEEB> I thought you were interested in work WRT FFmpeg [19:07:03 CET] <j-b> 'morning [19:14:43 CET] <kurosu> devinheitmueller, is that for a feature or a bug? For the later, having it on the bugtracker would be nice [19:15:01 CET] <kurosu> (bug or bugs :) [19:15:27 CET] <devinheitmueller> It relates to properly decoding interlaced streams, so I guess its a bug, or a feature if you consider supporting interlaced video a feature. :-) [19:15:39 CET] <nevcairiel> more like an anti-feature [19:15:42 CET] <nevcairiel> :) [19:15:54 CET] <kurosu> ah, that rings a bell, where the thing went nowhere [19:15:54 CET] <devinheitmueller> Hey, if I could get rid of all interlaced content from the world, I would. :-) [19:16:13 CET] <nevcairiel> It would help to not make new content in new codecs :-D [19:16:46 CET] <devinheitmueller> I wouldnt have objected if the HEVC people said deinterlace your damn content before encoding it. [19:17:08 CET] <nevcairiel> we almost made it to that [19:17:18 CET] <nevcairiel> but last minute someone put the damn interlaced metadata in [19:17:29 CET] <nevcairiel> at least there is no special interlaced coding [19:17:51 CET] <devinheitmueller> True. [19:18:05 CET] <devinheitmueller> OTOH, I have yet to see good open source deinterlacing that doesnt crush the CPU. [19:18:30 CET] <philipl> There's good open source deinterlacingthat crushes the GPU too. [19:18:43 CET] <devinheitmueller> yadif 2x does a decent job, but wow does it take CPU cycles. [19:19:48 CET] <devinheitmueller> All you have to do it tune to a channel with a news ticker at the bottom to see how terrible they are. [19:21:23 CET] <JEEB> I still haven't checked if the mix-mash deinterlacer that mixes yadif and w3f is any good [19:21:33 CET] <philipl> bwdif? [19:21:36 CET] <JEEB> yea [19:21:42 CET] <philipl> It seems to be better than yadif. [19:22:10 CET] <philipl> I've been trying to write a cuda version but I can't get it to pass a basical visual equivalence test. I'm doing something wrong... [19:23:13 CET] <devinheitmueller> Anybody have any idea if the vaapi deinterlacing is any good? [19:23:51 CET] <philipl> Seemed reasonable when I've briefly looked at it in the past. [19:25:49 CET] <devinheitmueller> Im sure Ill get to poke at it soon enouhg. [20:05:16 CET] <nevcairiel> durandal_1707: your opus change broke things [20:10:28 CET] <durandal_1707> fix it [20:10:56 CET] <JEEB> durandal_1707: also when you get the time read the back-log [20:26:58 CET] <durandal_1707> nevcairiel: i not gonna fix others bugs, one of those issues happen outside of ffmpeg [20:27:15 CET] <nevcairiel> you should run fate before pushing, if you dont, then you broke it =p [20:29:58 CET] <cone-041> ffmpeg 03Paul B Mahol 07master:68f289dacddc: avcodec/opus: check if internal is available [22:51:57 CET] <cone-041> ffmpeg 03Carl Eugen Hoyos 07master:271a00043ef3: lavf/rawdec: Do not mark streams from raw subtitle demuxers as data streams. [00:00:00 CET] --- Thu Nov 29 2018

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ffmpeg.log.20181128
by burek 29 Nov '18

29 Nov '18

[13:12:49 CET] <skident> Hi everyone, is it possible to use a filter loudnorm linear without adding measured_* options? [14:02:10 CET] <dfcnvt> Hi guys -- Can you tell me which of those compression and/or format is the lightest size or has the fastest rate of Camera Output as shown in this limited list: https://i.imgur.com/sODQ3GZ.png [14:03:22 CET] <pink_mist> afaik only H264 has even a chance of doing compression [14:03:36 CET] <pink_mist> the others I don't believe do any compression at all [14:07:41 CET] <pink_mist> I'm a complete noob at tht stuff though, so I may be completely wrong [14:07:44 CET] <pink_mist> *that [14:10:44 CET] <dfcnvt> pink_mist: Thanks -- I kinda thought so. I think it's the format. I guess I need to do the research by finding the size of the file and compare them (benchmark). [14:15:03 CET] <dfcnvt> Seem there's 3 different "TYPE"s: http://wiki.oz9aec.net/index.php/Pixel_formats [14:15:16 CET] <dfcnvt> packed, compressed, planar [14:32:42 CET] <iive> pink_mist, you are correct [14:36:52 CET] <iive> the pixel formats does take different space. e.g. rgb and bgr are using 3 bytes per pixel. aka 24 bits per pixel. YUYV 4:2:2 are using 16 bits per pixel, (8*4+8*2+8*2)/4; YV12 is YUV420 and uses 12 bits per pixel. [15:06:47 CET] <dfcnvt> iive: Thanks, I assume YUV12 would be the best to use -- with the interest of lightweight load and speed. [15:07:15 CET] <dfcnvt> Whoops, I meant YV12 [15:07:25 CET] <dfcnvt> U & V looks the same. [15:08:07 CET] <skident> Guys, is there somebody who built ffmpeg for Windows using MSVC2017 (release and debug versions)? [15:09:01 CET] <iive> dfcnvt, have you tried h264, it has lossless mode.it might be using it. record a sample and check it with ffprobe. [15:09:36 CET] <iive> otherwise yes, if the camera is not using interlace, use yv12, if it does yuyv 422 . [15:09:54 CET] <iive> interlace should have died with crt tv. [15:10:50 CET] <dfcnvt> Forgive me but I'm not quite catch up on the understanding of what's the "interlace" is. [15:11:38 CET] <iive> well, i guess you've seen 720p and 1080p, there is 720i and 1080i. [15:11:56 CET] <dfcnvt> nevermind, I checked the example of interlace -- http://www.infognition.com/tutorials/rip_guides/ntscfilmi.gif [15:12:08 CET] <iive> interlace is when you separate the picture in odd and even lines and you combine fields from different time. [15:12:39 CET] <iive> it kind of creates a comb effect on motion. [15:12:52 CET] <dfcnvt> Gotcha, like in the example. [15:14:19 CET] <dfcnvt> skident: I'm guessing you'd want to check the "Previous Builds" section here -- https://ffmpeg.zeranoe.com/builds/ [15:15:07 CET] <dfcnvt> Note, I'm just browsing out for you -- I have no knowledge whether this is reliable nor does it meet to your asking. [15:15:41 CET] <iive> the example looks like telecine. it is when you want to increase fps, but instead of repeating frames, you repeat fields. this way you turn nice and progressive video into mixed mess of progressive and interlaced frames. [15:15:46 CET] <skident> dfcnvt: no, I want to build ffmpeg libraries from sources on Windows and for Windows [15:18:05 CET] <skident> I tried to follow instructions (http://ffmpeg.org/platform.html#Microsoft-Visual-C_002b_002b-or-Intel-C_002…) but unfortunately I faced with problems like msys2 doesn't know where is cl.exe located [15:21:32 CET] <skident> or for example here is another manual https://trac.ffmpeg.org/wiki/CompilationGuide/WinRT, but it is only for Visual Studio 2015 [15:31:27 CET] <dfcnvt> skident: I see, this is out of my depth. [18:32:21 CET] <devinheitmueller> Anyone interested in doing some consulting to address some bugs in libavcodecs HEVC decoder? [18:32:31 CET] <jya> how can I get ffplay to properly downmix a video with a 5.1 audio when playing on stereo? [18:32:36 CET] <jya> right now it just drops the channel [18:34:13 CET] <JEEB> devinheitmueller: is this about making it output both fields into a single AVFrame? [18:34:17 CET] <devinheitmueller> jya: I cannot speak to ffplay, but at least with ffmpeg it will downmix to stereo if you specify -ac 2 [18:34:22 CET] <devinheitmueller> JEEB: Wow, thats exactly the issue. [18:34:38 CET] <jya> devinheitmueller: indeed, just tested that. [18:34:46 CET] <jya> so just converted with ffmpeg first [18:34:47 CET] <devinheitmueller> Ive got streams from a couple of different sources which exercise modes 1/2 and 9/10/11/12. [18:35:02 CET] <devinheitmueller> Pardon, I mean pic_struct 1/2 and 9/10/11/12. [18:35:59 CET] <JEEB> I've been thinking of taking a look at it since like none of the overall stuff actually handles separated fields automagically [18:36:10 CET] <devinheitmueller> I cooked up a quick video filter to merge the two fields into a single AVFrame for certain known broadcast resolutions (i.e. 1920x540, 720x240, etc), but thats obviously a hack and it really should be fixed in the decoder itself. [18:36:13 CET] <JEEB> buut, priorities are somewhere else [18:36:35 CET] <JEEB> (and I don't deal in interlaced HEVC too much yet) [18:37:04 CET] <devinheitmueller> My initial hope was that the code was close enough to the H.264 decoder that I could port the business logic. Then I started looking at the actual code. :-) [18:37:12 CET] <JEEB> yea :D [18:37:21 CET] <JEEB> thaat was my first guess/hope as well [18:38:36 CET] <JEEB> also I wonder how much my next MPEG-TS patch will raise eyebrows. :P (drop non-trustable [read: subtitles/teletext] packets if PCR hasn't been received yet) [18:38:59 CET] <devinheitmueller> Of course at least in my case Im hindered by my general lack of knowledge in how the standard operates (e.g. I had to look up what CTB stood for) [18:39:28 CET] <devinheitmueller> Oh, Ive got a fun MPEG TS patch in the works as well, to properly handle streams > 26.5 hours. [18:39:49 CET] <JEEB> there's actually lavf logic for that, but somehow that logic only triggers *once* [18:40:13 CET] <JEEB> but IIRC people were OK with an option that does that within the demuxer [18:40:38 CET] <devinheitmueller> Yeah, aside from the general improper handling of wraparound, in the cases where I do see it attempting it, it works exactly once. [18:40:48 CET] <JEEB> yup [18:41:08 CET] <devinheitmueller> I assume youre referring to the stuff in libavformat/utils.c. [18:41:10 CET] <JEEB> you get it once from lavf (expected) and then ffmpeg.c's discontinuity logic catches it [18:41:13 CET] <JEEB> yup [18:41:24 CET] <JEEB> and the ffmpeg.c code in theory should be dead weight by now :P [18:41:34 CET] <JEEB> yet it isn't, which is unexpected/boog in the lavf handling [18:41:53 CET] <JEEB> I also have a patch from Japan for the wrap-around handling which I still don't understand myself 100% [18:42:03 CET] <JEEB> but it seems to help with at least one sample I have [18:42:13 CET] <devinheitmueller> Also bear in mind that the clock gets re-based in ffmpeg.c. Hence even if you do wrapping at (1<<33) at the TS level, you still have to deal with the rebased clock wrapping around. [18:42:34 CET] <JEEB> yes [18:42:36 CET] <devinheitmueller> Hmm, if youve got a patch, send it along and I can compare it to what Ive done. [18:43:21 CET] <JEEB> https://github.com/jeeb/ffmpeg/commit/dfeeeb226094a8b00f7319b89e4d3b3660326… [18:43:36 CET] <JEEB> I don't fully habla it so I haven't posted it on the ML yet [18:44:14 CET] <JEEB> and that came to me without any comments etc :P [18:44:25 CET] <JEEB> https://kuroko.fushizen.eu/screenshots/ffmpeg/blue_vanilla-orange_patched.p… [18:44:33 CET] <JEEB> this is the result with one of my samples [18:44:52 CET] <JEEB> so it clearly fixes *something* [18:46:17 CET] <JEEB> also I think durandal_1707 was looking for things to do, so you might want to poke him on #ffmpeg-devel regarding the HEVC thing [18:46:21 CET] <devinheitmueller> I dont think that would work for streams that have B-frames. [18:47:18 CET] <devinheitmueller> Doh, I didnt even realize there was an #ffmpeg-devel channel. [00:00:00 CET] --- Thu Nov 29 2018

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ffmpeg-devel.log.20181127
by burek 28 Nov '18

28 Nov '18

[00:15:03 CET] <cone-786> ffmpeg 03Kyle Swanson 07master:fd2d6f376d84: libavf/libvmaf: update docs [01:16:49 CET] <cone-786> ffmpeg 03Carl Eugen Hoyos 07master:5a7617136ab6: lavc/jrevdct: Avoid an aliasing violation. [02:57:43 CET] <cone-786> ffmpeg 03Michael Niedermayer 07master:a036c2596949: avcodec/tiff: Fix integer overflows in left shift in init_image() [03:47:22 CET] <cone-786> ffmpeg 03Jun Zhao 07master:e002ec332b75: lavfi/buffersrc: Indent the code. [03:47:23 CET] <cone-786> ffmpeg 03Jun Zhao 07master:7608809ae424: fftools/ffprobe: Indent the code. [03:47:24 CET] <cone-786> ffmpeg 03Jun Zhao 07master:1ffac23885ee: fftools/ffmpeg: delete the unused code. [06:57:51 CET] <cone-015> ffmpeg 03Andrey Semashev 07master:1035206102e2: lavf/dashdec: Add webm to the list of allowed extensions. [12:10:02 CET] <BtbN> What's it with all people going crazy over nvenc/cuda stuff lately? Just got a very upset E-Mail asking me to undo the nvenc unbundling, because it fucked streamers. [12:10:58 CET] <durandal_1707> ? [12:11:06 CET] <nevcairiel> the header unbundling? [12:11:09 CET] <BtbN> yes [12:11:21 CET] <nevcairiel> if they cant figure out how to get a second repository, they shouldnt be building code :p [12:11:29 CET] <BtbN> It caused some distros to stop offering nvenc enabled ffmpeg builds, because it was auto-enabled before [12:12:49 CET] <BtbN> But really, go complain to those distros, or just build it yourself. [12:13:03 CET] <BtbN> All the more fancy stuff is non-free anyway [12:18:46 CET] <thardin> I talked to a local streaming shop recently that does a lot of cuda stuff [12:19:04 CET] <thardin> that and ~AI~ may be the reason [12:28:53 CET] <nevcairiel> the yadif_cuda filter doesn't really need the cuda SDK, does it? Not at runtime? Only for compiling the kernel at build time? [12:29:23 CET] <nevcairiel> hm i guess not all functions are in the compat header either [13:02:26 CET] <durandal_1707> atomnuker: there is opus patch on ml that needs resolving [13:10:51 CET] <BtbN> nevcairiel, the only thing making it non-free is it needing nvcc to build, yeah [13:11:13 CET] <nevcairiel> but that doesnt really influence the binaries non-free status, does it [13:11:20 CET] <BtbN> nope [13:11:31 CET] <BtbN> (L)GPL demands that you are able to reproduce the build [13:11:42 CET] <BtbN> which you clearly aren't without accepting Nvidias EULA [13:11:49 CET] <nevcairiel> would be nice if a portable build without linked cuda sdk would be possible with th ose filters [13:12:01 CET] <BtbN> It is [13:12:05 CET] <BtbN> It's really only nvcc [13:12:12 CET] <BtbN> to compile the .cu to .ptx [13:12:25 CET] <nevcairiel> its not currently the way its setup, not without changes anyway [13:12:44 CET] <BtbN> philipl added functions to the headers specifically for that though [14:24:07 CET] <durandal_1707> atomnuker: what is status with opus decoding and gapless? [14:39:08 CET] <cone-882> ffmpeg 03Michael Niedermayer 07master:28c96c2ce278: avcodec/diracdec: Check component quant [15:24:40 CET] <durandal_1707> this is outrageous, all development power is sucked into dav1d [15:25:35 CET] <j-b> durandal_1707: AV1 is the new kid in town. In a couple of months, it won't. [15:25:48 CET] <j-b> and seeing the mess that AV1 is... [15:45:35 CET] <kierank> durandal_1707: nonsense [16:37:36 CET] <atomnuker> durandal_1707: should work, if the packet duration and length are correct [18:38:37 CET] <cone-882> ffmpeg 03Carl Eugen Hoyos 07master:1893c7208602: lavfi/fps: Avoid duplicating Closed Captions when increasing frame rate. [18:38:38 CET] <cone-882> ffmpeg 03Carl Eugen Hoyos 07master:698574ac5d68: ffmpeg: Avoid duplicating Closed Captions when increasing frame rate. [18:49:29 CET] <cone-882> ffmpeg 03Carl Eugen Hoyos 07master:8cc5107705e9: lavf/Makefile: Fix demuxer dependencies: dvbsub, dvbtxt [20:45:49 CET] <j-b> 'morning [20:53:00 CET] <iive> where in the world is morning? [20:53:18 CET] <iive> australia, japan? [20:56:07 CET] <gnafu> iive: It is always morning when someone arrives on IRC, no? [20:57:14 CET] <iive> that's strange notion. [20:57:55 CET] <iive> I would prefer to think that j-b is enjoying the summer in Australia atm. [21:40:54 CET] <durandal_1707> atomnuker: seen my opus patch? [21:41:25 CET] <atomnuker> yeah, I'm not sure what it does [21:42:36 CET] <atomnuker> I was sure gapless worked [21:43:26 CET] <atomnuker> I guess if the packet doesn't have skip samples data decode.c uses avctx->internal->skip_samples [21:43:36 CET] <atomnuker> so it should be okay [21:44:40 CET] <atomnuker> but what is avctx->internal->skip_samples_multiplier [21:45:42 CET] <atomnuker> apparently its only used if there's side packet data, but if there is what's the point of multiplying the value when you can directly set it and modify it [21:48:18 CET] <atomnuker> ugh, its an aac he hack [21:48:47 CET] <JEEB> sounded like it since IIRC it uses half the sample rate and the decoder then makes u for it or so? [21:49:06 CET] <JEEB> and even in some containers the half-rates ended up in [21:49:25 CET] <atomnuker> durandal_1707: anyway, looks fine, can you make a fate test for gapless opus? its missing from tests/fate/gapless [22:36:15 CET] <j-b> iive: Australia, or NZ? [22:37:36 CET] <iive> j-b, yeh, New Zealand is next to Australia. [22:37:57 CET] <j-b> I would prefer imagining NZ then [22:49:16 CET] <iive> deal. I'll imagine that one too. [23:16:42 CET] <j-b> cool [00:00:00 CET] --- Wed Nov 28 2018

1 0

ffmpeg.log.20181127
by burek 28 Nov '18

28 Nov '18

[08:56:21 CET] <skident> hi, does anybody have build an ffmpeg library for windows 10 using MSVC 2017? [09:01:38 CET] <skident> I tried to use this guide https://trac.ffmpeg.org/wiki/CompilationGuide/WinRT, but paths are different from VS2015 [11:33:31 CET] <cslcm> Hi everyone - I have a big memory leak in this program: https://pastebin.com/raw/3X5h9YiK centered around av_bitstream_filter_filter - if i remove this call there is no leak. Could someone take a look and tell me if i'm handling resources incorrectly? [12:07:34 CET] <skident> \cslcm have you checked a return value from av_bitstream_filter_filter (int a = )? [12:10:36 CET] <skident> cslcm: have you checked a return value from av_bitstream_filter_filter (int a = )? [12:16:18 CET] <cslcm> the return value is always 1 [13:20:51 CET] <skident> how does this memleak looks like? It eats memory on each call of av_bitstream_filter_filter and doesn't free memory at all& [13:20:52 CET] <skident> ? [13:24:20 CET] <cslcm> right, it quickly fills up to gigabytes of allocations [13:24:48 CET] <cslcm> I posted a question on stackoverflow including valgrind output https://stackoverflow.com/questions/53498403/av-bitstream-filter-filter-mem… [13:25:21 CET] <cslcm> i suspect the leak may be internal to ffmpeg [13:25:37 CET] <BtbN> Unlikely, that functions is used a lot. [13:25:49 CET] <cslcm> with h264_mp4toannexb? [13:25:52 CET] <BtbN> yes [13:25:58 CET] <BtbN> it's auto-inserted whenever it's needed [13:26:51 CET] <cslcm> i could try with ffmpeg itself, is there a command line option to tell it to keep looping the input indefinitely [13:26:59 CET] <BtbN> -loop [13:27:05 CET] <cslcm> hah [13:27:07 CET] <cslcm> fancy that [13:28:23 CET] <BtbN> But I see your leek [13:28:26 CET] <BtbN> https://github.com/FFmpeg/FFmpeg/blob/0c14b73e435cf3373df7bca149e665d4e6b80… [13:28:38 CET] <BtbN> av_free_packet only frees buffers, not raw data on the packet [13:28:48 CET] <BtbN> *leak [13:29:45 CET] <BtbN> so your pkt2 is never freed [13:31:47 CET] <cslcm> but it's reused [13:31:59 CET] <BtbN> doesn't matter [13:32:00 CET] <cslcm> so it will leak once [13:32:17 CET] <BtbN> each call to av_bitstream_filter_filter calls malloc and assigns the result to pkt2.data [13:32:29 CET] <BtbN> And you just overwrite it with NULL again, never freeing it [13:32:40 CET] <cslcm> ah right, how do i free that data? [13:32:47 CET] <cslcm> literally just free? [13:32:51 CET] <BtbN> av_free [13:32:54 CET] <cslcm> ok [13:32:57 CET] <cslcm> will try that, thanks [13:33:08 CET] <BtbN> av_freep(&pkt.data) ideally [13:33:45 CET] <BtbN> You probably shouldn't use a packet there [13:34:05 CET] <BtbN> Or take a look at how the legacy compat code handles it: https://github.com/FFmpeg/FFmpeg/blob/f72b9904fefa79d799d0f6ecc8bd97ce52658… [13:36:33 CET] <BtbN> av_free_packet is also deprecated and should not be used. Use unref instead [13:37:50 CET] <cslcm> should i be freeing the data and the packet on each iteration? [13:37:53 CET] <cslcm> or just the data? [13:40:27 CET] <BtbN> The whole API you are using is deprecated btw. [13:40:45 CET] <BtbN> You should be using the av_bsf_* API instead [13:40:50 CET] <cslcm> I'm only following the examples on the ffmpeg website heh [13:40:53 CET] <cslcm> this is new to me [13:41:26 CET] <BtbN> https://www.ffmpeg.org/doxygen/trunk/group__lavc__misc.html#ga511d9ec0593e7… [13:41:48 CET] <cslcm> is there an example anywhere showing how to do it the new way? [13:42:37 CET] <BtbN> plenty all over the ffmpeg codebase. [13:42:44 CET] <BtbN> Search for av_bsf_ for example in http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/cuviddec.c [13:44:02 CET] <cslcm> looks like it's av_freep i needed and not av_free. and if i also try to av_free_packet it complains of a double-free. [13:44:15 CET] <cslcm> but the leak is fixed, thanks for your help [13:44:21 CET] <cslcm> i'll look into the new api [13:44:22 CET] <BtbN> av_free and av_freep do the same, but freep sets the pointer to NULL [13:44:41 CET] <BtbN> av_free(v); v = NULL; is equivalent to av_freep(&v); [13:45:45 CET] <cslcm> hm, that's very weird, because I am setting data to null after ever free... but free results in "munmap_chunk(): invalid pointer" whereas freep does not [13:46:18 CET] <BtbN> You're doing something wrong then [13:50:42 CET] <BtbN> The new API is way less prone to this btw. [13:50:54 CET] <cslcm> if av_free_packet doesn't free the packet data then why does it result in a double free when i try to free both [13:51:17 CET] <BtbN> You are freeing something you are not supposed to free I guess [13:51:29 CET] <BtbN> but really, use the new API, it makes a whole lot more sense [13:51:42 CET] <BtbN> And avoid that issue entirely [13:51:56 CET] <cslcm> i'm trying to find some actual examples [13:52:49 CET] <cslcm> anyway, thanks for your help :) I appreciate it [13:55:46 CET] <BtbN> The new API is fairly self-explanatory [13:55:58 CET] <BtbN> Why do you even want to use it directly? Usually the encoders that need it do it themselves [13:58:02 CET] <cslcm> openmaxil on the raspberry pi requires h264 data in this format [13:58:55 CET] <cslcm> interestingly none of the examples i could find on the web which use this filter free the frame data with av_free [13:59:00 CET] <cslcm> so they all leak i guess [14:16:00 CET] <cslcm> BtbN: So it appears that av_free is only necessary on the output from av_bitstream_filter_filter.. i guess av_read_frame does it for us [14:16:17 CET] <cslcm> that's why i was getting the double free issue [14:17:09 CET] <BtbN> The old API is really bad for that. It was replaced for a reason [14:17:26 CET] <cslcm> i get it, i know. I will look into it. Just have a deadline for this phase. [14:18:46 CET] <cslcm> ==44787== LEAK SUMMARY: ==44787== definitely lost: 0 bytes in 0 blocks ==44787== indirectly lost: 0 bytes in 0 blocks ==44787== possibly lost: 0 bytes in 0 blocks [14:18:48 CET] <cslcm> success :D [14:18:49 CET] <cslcm> cheers [14:19:04 CET] <cslcm> ls [14:19:07 CET] <cslcm> whoops [17:31:27 CET] <mooniac> I can't believe I can't find this with a simple web search. I want to use ffmpeg to record mike audio into a file. No video needed. Should be pretty simple I thought. [17:41:00 CET] <CoreX> mooniac maybe spell mic correctly [17:41:01 CET] <CoreX> http://free-tutorials.org/how-to-record-sound-from-microphone-with-ffmpeg/ [17:41:30 CET] <CoreX> or [17:41:31 CET] <CoreX> http://www.pc-freak.net/blog/record-microphone-input-sound-good-ffmpeg/ [17:44:24 CET] <mooniac> thx [17:54:52 CET] <rainey> when using libavresample to convert to a lesser amount of audio channels (eg stereo to mono), is it possible to specify which channel is taken? [18:06:48 CET] <mooniac> none of them worked for me. Oh well, I'll just use OBS then, that has a pulseaudio option (audio only, not screen capture). Works great. [18:12:22 CET] <rainey> I guess converting to mono I can just convert to stereo planar and get a pointer to whichever channel I need [21:28:55 CET] <TheAMM> How do I get RGBA from the lafvi color input filter? [21:29:08 CET] <TheAMM> ffmpeg -f lavfi -i "color=color=0xFFFFFF00:size=50x50" gives me yuv420p [21:29:35 CET] <TheAMM> vf format=rgba is technically rgba, but it's just a white square, because the input doesn't have the alpha [21:33:37 CET] <furq> TheAMM: color=...,format=yuva420p [21:34:26 CET] <TheAMM> Thanks [00:00:00 CET] --- Wed Nov 28 2018

1 0

ffmpeg-devel.log.20181126
by burek 27 Nov '18

27 Nov '18

[01:43:02 CET] <cone-588> ffmpeg 03James Almer 07master:cc25529420e3: avcodec/libdav1d: update the API usage after upstream changes [03:09:03 CET] <cone-588> ffmpeg 03Lauri Kasanen 07master:46c5693ea3a9: swscale/output: Altivec-optimize yuv2plane1_8 [03:09:04 CET] <cone-588> ffmpeg 03Michael Niedermayer 07master:90ac0e5f29ba: avcodec/tiff: Limit filtering to decoded data [11:17:19 CET] <BtbN> Does lavf automatically re-configure filters when the input parameters, like size, change? [11:24:19 CET] <nevcairiel> no, dynamic changes are not supported [11:24:44 CET] <nevcairiel> it would usually insert a new scale filter to restore the original size, if necessary [11:25:23 CET] <BtbN> So the answer to that guy on trac is just a mere "That's not supported." [11:26:39 CET] <nevcairiel> i have no idea whats that about, but its safe to say that ffmpeg filtering with a video that changes resolution mid-playback is often not going to do what someone might want [11:26:53 CET] <nevcairiel> it usually works, but only by scaling stuff [11:26:59 CET] <BtbN> He has scale,hwupload,hwfilters,hwdownload [11:27:07 CET] <BtbN> and then at runtime changes parameters on the first scale [11:27:24 CET] <nevcairiel> yeah thats probably not really supported [11:28:25 CET] <nevcairiel> the majority of filters are not build for changing input parameters [11:28:34 CET] <BtbN> It should be fairly straight forward to add support to hwupload/hwdownload [11:29:02 CET] <BtbN> hwdownload would be trivial, and hwupload needs to monitor the frame size and pix_fmt and re-run config_props when they change [11:29:18 CET] <nevcairiel> or maybe not, because you need a new frame context and tell all following hw filters abouut the new frame context [11:29:30 CET] <nevcairiel> which is probably not something we can do right now [11:30:25 CET] <BtbN> true. All the filters would need the same kind of logic [11:30:45 CET] <BtbN> Makes me wonder if this couldn't be written once globally outside of the filters [11:34:03 CET] <BtbN> hwupload_cuda can probably be removed by now btw. [12:44:48 CET] <BtbN> I give up on this guy. Either he sends patches, or I don't care anymore. [12:54:26 CET] <durandal_1707> lol [12:57:17 CET] <BtbN> Also not sure if that empty paste is a track issue, a mistake on his end, or some kind of joke. [15:24:59 CET] <durandal_1707> BradleyS: is something with pcmdec patches wrong? [15:25:52 CET] <BradleyS> nope, just curious about any connected issues [15:26:35 CET] <BradleyS> aside, it would be good to cherry-pick https://git.ffmpeg.org/gitweb/ffmpeg.git/commitdiff/027f032bbce9bdf7bbec406… [15:26:40 CET] <BradleyS> onto release/4.1 [15:26:56 CET] <BradleyS> handbrake needs that one, we'll patch for now [15:39:00 CET] <cone-468> ffmpeg 03Paul B Mahol 07release/4.1:fcffed470a52: avformat/movenc: fix size calculation in mov_write_eac3_tag() [15:39:00 CET] <cone-468> ffmpeg 03Paul B Mahol 07release/4.1:59e30c05d74e: avformat/movenc: get number of written bytes from bitstream writer [15:40:14 CET] <BradleyS> ty [18:38:54 CET] <durandal_1707> what was the reason we do not have pcm-dvd encoder? [19:10:51 CET] <kierank> jya: what was the link to reporting a page that doesn't work in firefox [19:49:29 CET] <philipl> BtbN: credit where credit is due, he did write a patch. [19:51:13 CET] <BtbN> I'm not sure about that patch though. Scaling to frames which have a way over-allocated buffer? Plus, with a hw_frames_ctx that reports a size that doesn't match the frames? [19:51:25 CET] <BtbN> That's asking for trouble [19:55:24 CET] <BtbN> Also, nvenc absolutely expects CUDA frames to be in a continous buffer [19:55:29 CET] <BtbN> this would 100% break it [20:51:19 CET] <durandal_1707> good luck with that guy, he will take you all exp [20:52:58 CET] <philipl> He's something special. [20:53:24 CET] <philipl> Sounds like he has it all sorted out. Maybe we have a new project leader. [20:54:28 CET] <durandal_1707> why mpeg muxer is full of hacks? [20:55:06 CET] <durandal_1707> for example it muxes pcm as pcm_dvd this is gross hack [20:55:41 CET] <durandal_1707> muxer should not do transcoding, who wrote this mess [20:55:57 CET] <philipl> Does it date back to pre-ffmpeg days... [20:58:40 CET] <durandal_1707> philipl: its very old, from 2003 [21:00:28 CET] <durandal_1707> i will remove direct PCM encoding, and it will use pcm-dvd encoder instead, if nobody is against [21:00:47 CET] <kierank> sure [21:00:48 CET] <kierank> remove hacks [21:00:59 CET] <philipl> Is there a command line compatibility issue? [21:01:17 CET] <durandal_1707> yes, but who cares [21:01:38 CET] <durandal_1707> basically pcm direct muxing will no longer work [21:02:18 CET] <philipl> I'm not personally affected, but doesn't command line behaviour fall under our major/minor compatibility contract? [21:02:45 CET] <philipl> (I guess it's also programmatic usage too) [21:03:31 CET] <philipl> I love getting rid of hacks too, so just wanted to point it out. [21:03:49 CET] <durandal_1707> i could leave pcm_s16be path but it looks like hack, so i rather not [21:04:56 CET] <atomnuker> what's wrong with the pcm_xxx encoders? [21:05:48 CET] <durandal_1707> atomnuker: mpeg muxers does not support directly pcm s16be [21:05:55 CET] <durandal_1707> they need header [21:08:33 CET] <atomnuker> can't it insert the header itself? [21:08:40 CET] <philipl> haha. [21:09:03 CET] <philipl> The circle is complete. [21:09:59 CET] <BtbN> I'm just gonna ignore him, no matter how valid some of the points might be, cba dealing with that shit. [21:11:15 CET] <philipl> Yeah. That attitude isn't productive. [21:12:44 CET] <BtbN> He's also wrong. His patch still uses data[0] and data[1], and never adjusts them. So sure, it might work in his special case, but it produces invalid frames. [21:18:11 CET] <philipl> I'm pretty sure he said he's not wrong, so that means he can't be wrong. [21:18:35 CET] <BtbN> true [23:09:28 CET] <cone-786> ffmpeg 03James Almer 07master:2ddaaaf59570: avcodec/libaomenc: increase the default bitrate [23:51:17 CET] <cone-786> ffmpeg 03Mark Harris 07master:01dc152a92d3: avformat/vivo: Don't log null value [23:51:18 CET] <cone-786> ffmpeg 03Mark Harris 07master:8108064043bf: avfilter/vf_chromashift: Fix mixed declaration and code [23:51:19 CET] <cone-786> ffmpeg 03Mark Harris 07master:4361293fcf59: avutil/mem: Fix invalid use of av_alloc_size [00:00:00 CET] --- Tue Nov 27 2018

1 0

ffmpeg.log.20181126
by burek 27 Nov '18

27 Nov '18

[01:48:29 CET] <iio7> I have a h265 mkv file with several audio tracks I want to remove, so I basically did this: "ffmpeg -i foo.mkv -map 0:0 -vcodec copy -map 0:1 -acodec copy -sn -threads 8 bar.mkv", I suspected it would use all threads, but it's only using 1. I also suspected, since it's just a copy, it would be rather quick, but it takes several hours on a 3Ghz i7. Am I missing something? [01:53:01 CET] <Hello71> is your storage device very slow [01:53:18 CET] <iio7> No. [04:00:07 CET] <Alish> how to add blur to a some screen area for some seconds? [04:00:32 CET] <Alish> !info blur [04:04:59 CET] <Alish> and can this be reversed if I make a videe, blur it and upload to youtube? [04:05:21 CET] <Alish> then anyone who downloaded can undo th blur? [04:12:02 CET] <Alish> how to add blur to a some screen area for some seconds? [04:35:29 CET] <Alish> Input #0, matroska,webm, from 'mobi.mkv': [04:35:31 CET] <Alish> Metadata: [04:35:33 CET] <Alish> COM.ANDROID.VERSION: 7.0 [04:35:35 CET] <Alish> MAJOR_BRAND : mp42 [04:35:37 CET] <Alish> MINOR_VERSION : 0 [04:35:39 CET] <Alish> COMPATIBLE_BRANDS: isommp42 [04:35:41 CET] <Alish> ENCODER : Lavf58.20.100 [04:35:43 CET] <Alish> Duration: 00:04:58.98, start: 0.000000, bitrate: 924 kb/s [04:35:45 CET] <Alish> Stream #0:0(eng): Video: h264 (High), yuv420p(progressive), 1080x1920 [SAR 1:1 DAR 9:16], 20.59 fps, 20.59 tbr, 1k tbn, 41.17 tbc (default) [04:35:47 CET] <Alish> Metadata: [04:35:49 CET] <Alish> HANDLER_NAME : VideoHandle [04:35:51 CET] <Alish> ENCODER : Lavc58.35.100 libx264 [04:35:53 CET] <Alish> DURATION : 00:04:58.982000000 [04:35:58 CET] <Alish> here is the ffprobe for the video. I just wanna hide th keyboard for some seconds [05:48:53 CET] <Fenrirthviti> Alish: In the future, you'll want to use something like pastebin for output like that, spamming the channel is pretty well frowned upon [08:09:29 CET] <idlus> Hi, Id like to use the documented "tpad" filter, but it doesnt seem to be included in my ffmpeg (arch package), my ffmpeg-filters manpage doesnt even mention it. [11:10:03 CET] <relaxed> idlus: you need a more recent version. Arch probably has a option to install ffmpeg git master [13:26:14 CET] <idlus> relaxed: I just upgraded 4.0.2 to 4.1, still without tpad, so it means the documentations includes features not yet released? Ill try the git version. [13:31:59 CET] <BtbN> tpad was added 2018-11-02, 4.1 was branched the same day, so it very narrowly missed the cutoff [13:34:58 CET] <idlus> BtbN: ok, I only looked at the release summaries to see if it was a new feature, didnt think it would be documented otherwise. Good to know. [13:35:18 CET] <BtbN> The releases pretty much only exist to make distributions happy [17:11:57 CET] <kepstin> ah, tpad filter looks handy. I was thinking of adding something to the fps filter to allow extending a video by repeating the last frame, but this more general filter is probably a better idea. [17:56:54 CET] <utack_> where was the discussion about "pict_type=?" for AV1 in ffprobe again? I think JEEB said it was arleady discussed somewhere to fix it [21:08:43 CET] <lavaflow> I've go a file where stream 0:0 is audio and 0:1 is video, and I'd like two swap them around without reencoding the whole deal, how can I do that? [21:09:46 CET] <pink_mist> mkvmerge? [21:10:10 CET] <JEEB> or just remux? [21:10:13 CET] <klaxa> ffmpeg -i input.mkv -map 0:v -map 0:a -c copy output.mkv ? [21:10:19 CET] <JEEB> by mapping first the one you want first [21:10:24 CET] <klaxa> or rather -map 0:1 -map 0:0 [21:10:55 CET] <JEEB> (when you add -map into ffmpeg.c cli, it will stop any automagic stream selection) [21:25:23 CET] <Alina-malina> ffmpeg -i D:\VIDZ\episode40.mp4 -i D:\VIDZ\episode40_ready.mp3 -vf "fade=out:221:29" -c:v copy -map 0:v:0 -map 1:a:0 D:\VIDZ\episode40_SOUNDED.mp4 i am trying to add fade out at the end but i get an error like: Filtergraph 'fade=out:91:29, fade=out:221:29' was defined for video output stream 0:0 but codec copy was selected. Filtering and streamcopy cannot be used together. [21:38:57 CET] <lavaflow> thank you all [21:52:18 CET] <JEEB> &25 [00:00:00 CET] --- Tue Nov 27 2018

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ffmpeg-devel.log.20181125
by burek 26 Nov '18

26 Nov '18

[00:27:44 CET] <cone-040> ffmpeg 03Igor Derzhavin 07master:a80530a627de: avformat/pcmdec: mime-type should be case insensitive (audio/L16) [00:27:45 CET] <cone-040> ffmpeg 03Igor Derzhavin 07master:b058617fae20: avformat/pcmdec: audio/L16 should be in network byte order by default (rfc 2586) [00:27:47 CET] <cone-040> ffmpeg 03Igor Derzhavin 07master:f95c928f80f1: avformat/pcmdec: endianness for audio/L16 mime type [00:38:47 CET] <cone-040> ffmpeg 03Carl Eugen Hoyos 07master:63c8558667db: configure: Disable direct stripping on sunos. [00:39:27 CET] <cone-040> ffmpeg 03Carl Eugen Hoyos 07master:737b5f586925: configure: Add -Wno-char-subscripts. [01:34:40 CET] <BradleyS> i don't see any discussion on the mailing list regarding the pcmdec patches, is there some public discussion or ticket referencing them? [01:51:51 CET] <nevcairiel> it was on the ML, but there wasnt that much discussion [02:16:15 CET] <BradleyS> i must be blind, link please nevcairiel? [07:02:23 CET] <cone-757> ffmpeg 03Limin Wang 07master:370b8bd8470a: libavcodec/libdavs2.c: Fix for the wrong line size is used [11:36:33 CET] <cone-568> ffmpeg 03Steven Liu 07master:f9ea4931479b: avformat/hlsenc: remove the cannot reach result [13:03:19 CET] <cone-568> ffmpeg 03Paul B Mahol 07release/4.1:9efc591cb72b: avfilter/vf_overlay: fix crash with negative y [20:44:58 CET] <durandal_1707> which bug should be fixed ASAP? [20:57:53 CET] <durandal_1707> looking for SVAC chinese codec samples? [21:02:06 CET] <Compn> never heard of it [21:02:12 CET] <Compn> why china make new codecs :( [21:05:11 CET] <thardin> mrrr, getting a ton of linker errors when cross-compiling for mingw32 [21:05:37 CET] <JEEB> are you using mingw-w64 or ye olde mingw? [21:06:06 CET] <thardin> ./configure --arch=x86 --target-os=mingw32 --cross-prefix=i686-w64-mingw32- [21:06:10 CET] <thardin> straight off the wiki [21:06:14 CET] <JEEB> ok [21:06:17 CET] <JEEB> so mingw-w64 [21:06:41 CET] <thardin> doing make clean && make. I did a compile before this too [21:13:16 CET] <durandal_1707> Compn: its chinese surveillance codec [21:44:26 CET] <cone-588> ffmpeg 03Paul B Mahol 07master:7488d58436c2: avcodec/eac3dec: remove obsolete comment [22:04:14 CET] <thardin> that did the trick [22:21:48 CET] <cone-588> ffmpeg 03Martin Vignali 07master:e488d7b4f507: fate/prores-metadata : make output bit exact [00:00:00 CET] --- Mon Nov 26 2018

1 0

ffmpeg.log.20181125
by burek 26 Nov '18

26 Nov '18

[02:30:14 CET] <markweston> what is the smallest OS (in total lines of code) that has ffplay ported to and can play x254 video @ 720p? [06:46:23 CET] <Woodpecker> I need to take a directory full of about 3000 audio files, with wildly different volumes with many peaks and valleys, and adjust the audio so it is all within a very uniform range. That means, no peaks or valleys are allowed to exist outside the range. I dont know what the term for this process is, or where to start. [07:26:40 CET] <sinclair> Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument [07:28:59 CET] <sinclair> sorry Woodpecker [07:29:24 CET] <Woodpecker> sinclair: I think I found it, loudnorm [07:30:01 CET] <sinclair> Woodpecker: I have a bit of a issue issue with ffmpeg myself, im trying to pipe images in over stdin, and pipe h264 (mp4) video out over stdout. [07:30:31 CET] <Woodpecker> sinclair: I actually have a bit of experience with that. [07:30:36 CET] <Woodpecker> sinclair: are you doing a slideshow? [07:31:08 CET] <sinclair> Woodpecker: well, i have a bunch of rendered (ray traced frames), these are computed at 24fps, and i have thousands of them. [07:31:20 CET] <sinclair> Woodpecker: am using image2pipe, but im not really sure what i should be doing [07:31:32 CET] <sinclair> Woodpecker: there is no 'audio' in my case. [07:32:54 CET] <Woodpecker> sinclair: I dont use quite so many frames. My videos might have 100-200 frames for an hour long video. For concatenating them... mmm let me check what I was using. [07:32:55 CET] <sinclair> Woodpecker: the idea is to be able to feed these frames into ffmpeg, have it encode them as mp4, then stream them back out over stdout (which get persisted on a web server), the whole thing is distributed, and i can't rely on sourcing images from disk using patterns, its all got to be done over stdio. [07:33:13 CET] <sinclair> Woodpecker: any examples would be great [07:35:56 CET] <Woodpecker> sinclair: unfortunately my code has been lost in a tragic wipe, but if I recall, I was using imagemagick to create the images, and for the timing, I used an ffmpeg script file [07:36:50 CET] <Woodpecker> and now I got to go [07:36:52 CET] <Woodpecker> sorry [07:37:13 CET] <sinclair> Woodpecker: alright, well, thanks anyway [08:01:46 CET] <sinclair> is anyone able to help me with this command [08:01:48 CET] <sinclair> 'ffmpeg -f image2pipe -vcodec png -framerate 24 -f mp4 -i - -r 24 -s 1024x768 -f mp4 -crf 15 -pix_fmt yuv420p -preset veryslow pipe:0' [08:01:58 CET] <sinclair> i have no idea what im doing \o/ [08:18:36 CET] <sinclair> if im piping images in over stdin, how do i tell ffmpeg that its received the last image and should process and exit? [08:18:51 CET] <sinclair> i think there are likely images still left in the codec buffer [08:22:21 CET] <sinclair> is there a good source of information for using ffmpeg? [08:25:34 CET] <sinclair> hello? [13:13:03 CET] <ProN00b> is there a way to generate HLS/DASH segments on demand? (like if someone wants to skip to the last 10 seconds of a video, that only the segments around that get transcoded?) [13:41:55 CET] <JEEB> ProN00b: make an API client and generate the segments on demand. for live the most usual design is to have an archive which is being fed by a live encoder from one side, and then you on-demand generate what clients request [13:42:21 CET] <JEEB> for VOD if your content is already properly encoded then you already have your archive [14:29:51 CET] <FurretUber> Hi, I'm trying to encode video using vp8_vaapi but the encoding is failing: http://termbin.com/75k7 The vainfo output can be seem at: http://termbin.com/8i1g [14:31:00 CET] <FurretUber> The CPU is an i3-6100U and the GPU is an Intel HD Graphics 520. With hevc_vaapi a very similar error happens while with h264_vaapi it works [14:40:28 CET] <jkqxz> FurretUber: Is that dynamic linker path override making it pick a different driver, perhaps? [14:40:54 CET] <jkqxz> The log with -v debug will show what driver it actually loaded. [14:47:23 CET] <FurretUber> Looks like it found the driver in the right location: http://termbin.com/50tm I'm not replacing libraries related to VAAPI, only software encoders (libopus, libvpx, libx264) [14:48:30 CET] <jkqxz> Ah. There isn't a low-power encode entrypoint for VP8. Remove the -low_power option. [14:52:19 CET] <FurretUber> Now it works. Thank you. As it was showing in -h encoder=vp8_vaapi I thought it was working for it too [14:55:09 CET] <jkqxz> It depends on the driver. A driver could offer that, but from vainfo your version only does for H.264. (Some Intel drivers also offer it for H.265, and maybe more in future.) [14:56:04 CET] <FurretUber> From my tests now, only H264 and MPEG2 works with low-power option [17:58:52 CET] <preyalone> anyone else notice ffmpeg -i running more slowly when overwriting an existing output file, compared to creating a new output file? [20:02:36 CET] <pong> that sounds OS-specific [23:28:18 CET] <OnkelTem> Oh hi [23:28:28 CET] <OnkelTem> Glad to see this channel alive [23:29:12 CET] <OnkelTem> I'm on Ubuntu 18.04 and trying to convert some MP2 audiotracks to AAC. And I'm curious how can I do it using libfdk-aac cuz it's available. [23:29:18 CET] <OnkelTem> it's NOT available [23:29:27 CET] <OnkelTem> So... what options guys do I have now? [23:30:35 CET] <FurretUber> There is the aac encoder. Not sure about how good the aac encoder from the FFmpeg version Ubuntu ships is [23:30:53 CET] <OnkelTem> you mean -a:c aac? [23:30:59 CET] <FurretUber> Yes [23:30:59 CET] <OnkelTem> -c:a* [23:31:44 CET] <OnkelTem> Good, I'm using exactly that now. Just tried to follow steps from here https://superuser.com/a/370637/122672 [23:31:58 CET] <OnkelTem> For some reason libfdc_aac was mentioned first [23:32:11 CET] <OnkelTem> and it said something about possible issues with source VBR [23:32:47 CET] <OnkelTem> I haven't faced any though :) [00:00:00 CET] --- Mon Nov 26 2018

1 0

ffmpeg-devel.log.20181124
by burek 25 Nov '18

25 Nov '18

[12:35:41 CET] <durandal_1707> Compn: find me more .dav files [12:52:38 CET] <durandal_1707> always glad to find unstripped binaries in wild [13:08:17 CET] <j-b> http://streams.videolan.org/misc/DNR114S_ch1_main_20161123072428_2016112307… [13:08:34 CET] <j-b> http://streams.videolan.org/MPlayer/incoming/HCVR_ch1_main_20160114112651_2… [13:09:30 CET] <durandal_1707> thank you Sir! [13:11:07 CET] <j-b> durandal_1707: and 3 here: https://streams.videolan.org/misc/dav/ [13:11:34 CET] <j-b> durandal_1707: I have _no_ idea, if they are actually DAV files or just random files named .dav. [13:11:41 CET] <j-b> but that is what I found. [13:13:56 CET] <JEEB> durandal_1707: I agree 100%, unstripped binaries are <3 [13:16:21 CET] <BtbN> the filter chain building logic does number filters from left to right, does it? [13:19:02 CET] <JEEB> there's a nice function that builds you a visual representation of your filter connections in a filter chain. not sure why ffmpeg.c doesn't utilize it [13:19:43 CET] <JEEB> char *filter_graph_dump = avfilter_graph_dump(filter_graph, NULL); [13:20:00 CET] <JEEB> that after linking and config [13:20:20 CET] <BtbN> I mean in that ticket. He claims that hwdownload does not react to the size change. But I fail to see why it would need to, or even be involved in it. [14:37:29 CET] <durandal_1707> lol, thilo mail flagged by google as potential phishing [14:48:47 CET] <atomnuker> I did tell him it was textbook spamming [15:16:18 CET] <BtbN> My server is also blacklisted by Microsoft for ages, and I recently learned the reason why they ignore my requests to fix that: http://blog.fefe.de/?ts=a50aba98 [15:16:56 CET] <BtbN> So I have to send more non-spam to them, so it can learn I'm good, which I can't, because it learned that I'm bad already, from whatever data. [15:17:18 CET] <cone-233> ffmpeg 03Paul B Mahol 07master:b80003fe286e: avformat: add DHAV demuxer [15:18:08 CET] <BtbN> What's with all the patches to fix DOS support lately? Why? [15:18:45 CET] <durandal_1707> pross developement machine is MS DOS 6 [16:25:51 CET] <cone-233> ffmpeg 03Martin Vignali 07master:80b0624099da: avutil : add YUVA444P12 and YUVA422P12 [16:25:52 CET] <cone-233> ffmpeg 03Martin Vignali 07master:86e6f0dbc7fd: swscale : add support for YUVA444P12 and YUVA422P12 [18:03:40 CET] <j-b> durandal_1707: I found no other sample, on our servers. [18:28:44 CET] <durandal_1707> j-b: ok [18:38:41 CET] <durandal_1707> j-b: what is policy to adding demuxing of dav to vlc? libavformat have some support now [18:38:57 CET] <j-b> durandal_1707: then, it is done. [18:39:26 CET] <j-b> durandal_1707: If people want/care, you can add the .dav to open-panel and register the extension on Windows. But basically, there is nothing to do. [19:53:58 CET] <durandal_1707> do we have helper to convert time/date to pts? [20:01:18 CET] <JEEB> time yes [20:01:50 CET] <JEEB> av_parse_time [20:02:11 CET] <JEEB> that gets converted into int64_t [20:02:53 CET] <JEEB> and I think it returns the value in AV_TIME_BASE_Q [20:39:59 CET] <Compn> someone with an @ffmpeg.org email , i need some halp... [20:41:03 CET] <durandal_1707> ? [20:42:06 CET] <Compn> you got one paul ? [20:43:01 CET] <durandal_1707> no [20:43:24 CET] <durandal_1707> better ask on ml [20:56:48 CET] <Compn> oh thats what happened lol [20:57:02 CET] <Compn> some guy spoofed the sender to my email, and my email forwarded it to my gmail [20:57:14 CET] <Compn> so the mail headers said this was a legit email from ffmpeg :D [20:57:22 CET] <Compn> i'm just going to sit in the corner now and look stupid [20:59:14 CET] <BradleyS> sounds like a smart analysis to me [21:01:22 CET] <Compn> eventually i found the bangladeshi ip address within all of the nonsense mail headers. [21:55:22 CET] <durandal_1707> if i want to use generic index for seeking but need to do some stuff after seek what should i do? [22:13:25 CET] <cone-040> ffmpeg 03Michael Niedermayer 07master:2ae39d795613: avcodec/truemotion2: fix integer overflows in tm2_low_chroma() [22:27:36 CET] <cone-040> ffmpeg 03Paul B Mahol 07master:24386b170753: avformat/dhav: improve A/V sync [22:52:40 CET] <durandal_1707> what i need to fill in this survey? [00:00:00 CET] --- Sun Nov 25 2018

1 0

ffmpeg.log.20181124
by burek 25 Nov '18

25 Nov '18

[02:22:09 CET] <wheelzi> How would I achieve this? i tried ffmpeg -i in *.mkv; do ffmpeg -i "$i" -c:a libfdk_aac -b:a 384k -c:v copy "${i%.mkv}.mp4"; done i get bash: syntax error near unexpected token do' any advice? [02:22:22 CET] <wheelzi> I have a folder of mkvs with ac3 that I want to batch convert to mp4 with aac 5.1 leaving the video untouched. [02:22:34 CET] <wheelzi> How would I achieve this? i tried ffmpeg -i in *.mkv; do ffmpeg -i "$i" -c:a libfdk_aac -b:a 384k -c:v copy "${i%.mkv}.mp4"; done i get bash: syntax error near unexpected token do' any advice? [09:47:36 CET] <Daisae> Whiskey: That is a bash question rather than an FFMPEG one, so another channel would perhaps be better able to help you. [09:47:50 CET] <Daisae> oops, that was meant for wheelzi, who is gone. [09:48:35 CET] <linux_gnu> Hi , anyone could suggest command to use filter for altering voice in any cool way???thanks [10:11:42 CET] <linux_gnu> anyone knows any ffmpeg commands to alter voice??? I don tknow anything about filters options... [11:17:14 CET] <durandal_1707> linux_gnu: see tremolo and vibrato filter docs [11:41:59 CET] <linux_gnu> anyone has advice for disquising voice recordings with ffmpeg???? [11:44:24 CET] <durandal_1707> linux_gnu: told you already [11:47:26 CET] <linux_gnu> durandal ,if today I got disconnected [11:48:44 CET] <linux_gnu> durandal ,you gave me 2 commands 2 months ago ,only one is working ,I wish I had more options.. [11:49:26 CET] <linux_gnu> this one doesn t work ffmpeg -i voice.wav -f lavfi -i sine=400 -lavfi amultiply out.wav [11:50:26 CET] <linux_gnu> the one -af vibrato=f=37 is nice! [11:52:19 CET] <durandal_1707> "doesn't" work means nothing to me, probably you ffmpeg is too old [11:58:30 CET] <linux_gnu> durandal ffmpeg -i a.mp3 -f lavfi -i sine=400 -lavfi amultiply out.mp3 [11:58:45 CET] <linux_gnu> gives No such filter: 'amultiply' [11:58:45 CET] <linux_gnu> Error initializing complex filters. [11:58:45 CET] <linux_gnu> Invalid argument [11:59:31 CET] <linux_gnu> durandal ubuntu 18.10 up to date [12:00:01 CET] <furq> you need 4.1 or newer for amultiply [12:01:45 CET] <linux_gnu> durandal: 7:4.0.2-2 is my syntax wrong? [12:03:05 CET] <pink_mist> what is 7:4.0.2-2 meant to be? [12:03:12 CET] <pink_mist> some random number an punctuation? [12:03:19 CET] <pink_mist> *numbers and [12:03:40 CET] <linux_gnu> it s result of dpkg -s ffmpeg | grep "Version:" [12:04:09 CET] <pink_mist> well that doesn't look anything like 4.1 [12:05:08 CET] <linux_gnu> I get it , sorry ): it means 4.0.. [12:06:19 CET] <linux_gnu> durandal: maybe an example with tremolo?? [12:10:40 CET] <durandal_1707> linux_gnu: you could alter more with afftfilt in frequency domain, but i dunno any simple command right now [12:11:17 CET] <durandal_1707> and to have more options you really need at least 4.1 version [12:11:51 CET] <durandal_1707> alternatively use aeval filter, but that too needs complex expessions [12:16:24 CET] <linux_gnu> durandal thanks! (first time I m impeded by a sowtware being too old in ubuntu...learning arch at the moment so will play with it on it [12:19:16 CET] <linux_gnu> durandal tremolo I did : ffmpeg -i a.mp3 -af tremolo=f=5 out2.mp3 [12:19:34 CET] <linux_gnu> and I don t notice any alteration ,any tip? [12:34:53 CET] <durandal_1707> linux_gnu: you need to more experiment [12:35:12 CET] <durandal_1707> bigger number, also change other options [12:35:44 CET] <linux_gnu> durandal: thanks ,I will ,and look into all you pointed [12:36:07 CET] <linux_gnu> durandal what options? you mean adding several filters? [12:39:39 CET] <durandal_1707> no, see: ffmpeg -h filter=tremolo [12:39:56 CET] <durandal_1707> there is 2nd option: depth/d [12:43:04 CET] <linux_gnu> durandal: thanks!!! ,(I m really new to filters..) [12:51:11 CET] <linux_gnu> durandal: ok tremolo is not altering much ,but I ll dig into all the others! [17:39:06 CET] <Daisae> How do I get make FFMPEG produce VBR video and adjust the quality? I tried # ffmpeg -i Input.mkv -codec:v libvpx-vp9 -crf 60 Output.webm [17:39:06 CET] <Daisae> and the result is low visual quality. I suspect that it is constraining the bit rate. [17:39:47 CET] <JEEB> I think you should look at the libvpx specific documentation on vp9's CRF range (or actually, what it maps against and what its range is) [17:39:50 CET] <JEEB> among other things [17:44:13 CET] <Daisae> I take your comment and the parenthesis in the FFMPEG doc to mean that it enables CQ mode. That would answer why the throughput is constrained. A WWW site I read said that it would produce VBR output but apparently that was wrong. [17:44:49 CET] <Daisae> It still leaves a mystery of how to enable VBR output. I've read in https://www.webmproject.org/docs/encoder-parameters/ [17:45:17 CET] <Daisae> and the vpenc manual [17:45:34 CET] <Daisae> vpxenc [17:45:47 CET] <JEEB> and then the community wiki @ trac has these https://trac.ffmpeg.org/wiki/Encode/VP9 [17:45:58 CET] <JEEB> I've not used VP9 myself at all, though [17:48:03 CET] <Daisae> That could be on the right trac. Thanks, I will investigate it. [17:49:10 CET] <Daisae> VBR section is empty. I guess they didn't find information either. [17:49:12 CET] <furq> Daisae: short answer is you need to set -b:v 0 to use crf with vp9 [17:49:59 CET] <furq> recommended settings for vpx are still some kind of elusive mystery though [17:56:52 CET] <Daisae> That -b:v 0 did cause crf to take effect for video. [17:56:52 CET] <Daisae> However, with -b:a 0 , libopus still reports : No bit rate set. Defaulting to 96000 bps. [17:57:34 CET] <BtbN> I don't think opus supports vbr? [17:58:17 CET] <JEEB> also generally vbr in audio is setting the quantizer [17:58:27 CET] <JEEB> aka "don't expect all encoders do the same thing" [17:58:35 CET] <Daisae> opusenc manual: -vbr Use variable bitrate encoding (default) [17:58:47 CET] <JEEB> and libvpx is the more special case [18:00:00 CET] <JEEB> if you had tried to set the quantizer [18:00:00 CET] <JEEB> "Quality-based encoding not supported, " [18:00:01 CET] <JEEB> "please specify a bitrate and VBR setting.\n") [18:00:17 CET] <JEEB> so it needs a bit rate and the VBR setting (which is probably default looking at the AVOptions) [18:00:33 CET] <furq> opus is vbr by default [18:00:49 CET] <furq> -b:a sets the nominal bitrate but it's basically the same as using -q for vorbis or -V for lame [18:01:04 CET] <furq> there's also an abr mode and a cbr mode if you really want to use them [18:01:07 CET] <JEEB> Daisae: welcome to all encoders are different :P [19:45:49 CET] <kerio> why am i getting a tbn of 10000k when remuxing an avi to mp4? :\ [19:45:52 CET] <kerio> quicktime is complaining [19:47:45 CET] <JEEB> 10k I would understand but 10,000k is quite high of a time base [19:47:48 CET] <JEEB> or wait [19:47:51 CET] <JEEB> that's one of the time base things [19:47:56 CET] <JEEB> although it still sounds high [19:48:16 CET] <JEEB> I know movenc.c had a thing that artificially heightened the video time base for some reason [19:48:25 CET] <JEEB> I even tried to remove that at one point [19:49:48 CET] <JEEB> http://ffmpeg.org/pipermail/ffmpeg-devel/2018-March/226366.html [19:50:01 CET] <JEEB> or not remove, but I brought up its weirdness on the mailing list [19:51:30 CET] <kerio> do i just reencode this shit [19:51:44 CET] <JEEB> you can set the output time base [19:51:51 CET] <JEEB> IIRC ffmpeg.c had an option for that [19:51:57 CET] <JEEB> although movenc will then still make it larger than 10k [19:52:10 CET] <JEEB> just so it can position audio "correctly" (without edit lists) it seems [19:52:31 CET] <JEEB> although there of course is an AVOption to override that behavior which makes it all more hilarious [19:52:57 CET] <JEEB> I would be interested in what makes it think it needs a big enough time base for that, though [19:53:00 CET] <JEEB> like 10,000k [19:54:46 CET] <kerio> well apparently the audio is also an issue here [19:54:51 CET] <kerio> quicktime refuses to play this mp3 [19:54:54 CET] <wheelzi> . [19:54:55 CET] <kerio> who made this shit [19:55:39 CET] <wheelzi> I got disconnected last night anybody know if anyone answered my batch convert question? [19:56:21 CET] <kerio> oh hold ON [19:56:29 CET] <kerio> this is XVID [19:56:35 CET] <kerio> no wonder quicktime doesn't like it [20:00:20 CET] <JEEB> lol [20:03:19 CET] <ChocolateArmpits> wheelzi, there's an archive of this channel available, you may look into it [20:03:58 CET] <wheelzi> where is it thanks [20:05:02 CET] <ChocolateArmpits> somewhere, I forgot where [20:06:37 CET] <ChocolateArmpits> can't find it :/ hmmm [20:06:52 CET] <wheelzi> I'll re ask lol. I have a folder of mkvs with ac3 that I want to batch convert to mp4 with aac 5.1 leaving the video untouched. [20:06:52 CET] <wheelzi> How would I achieve this? [20:07:17 CET] <ChocolateArmpits> wheelzi, what OS are you using? [20:07:54 CET] <wheelzi> linux mint. I've tried ffmpeg -i in *.mkv; do ffmpeg -i "$i" -c:a libfdk_aac -b:a 384k -c:v copy "${i%.mkv}.mp4"; done [20:07:54 CET] <wheelzi> i get bash: syntax error near unexpected token do' [20:08:21 CET] <ChocolateArmpits> sorry not familiar with bash [20:09:55 CET] <wheelzi> I had 1 for Windows but it doesn't work for linux [20:11:15 CET] <RemedyTomm> Well I just came into this channel to ask for help with a linking issue, but then i accidentally squirted coca cola up my nose and somehow i realised what I was doing wrong [20:11:48 CET] <RemedyTomm> forgot extern C >.< [20:12:05 CET] <ChocolateArmpits> >squirted coca cola up my nose [20:12:11 CET] <ChocolateArmpits> Thanks, I'll try that next time my brain isn't working [20:13:15 CET] <RemedyTomm> wrong channel anyway, i guess.. I probably wanted #ffmpeg-devel [20:13:33 CET] <RemedyTomm> oh, or not [20:13:40 CET] <RemedyTomm> #confuse [20:52:27 CET] <jin> ffmpeg -i in.mp4 -filter:v "crop=in_w:in_h-100" -c:a copy out.mp4 [20:52:31 CET] <jin> this command crops 100 from top and bottom [20:52:40 CET] <jin> what if I wanted to shave some pixels just off the top? [21:15:47 CET] <misterunknown> Hi guys. I try to make a video like the amazon fire tv screensaver (i.e. an image slideshow with different zoompan filters). I already figured out how to achieve the zoompan effect and created four configurations (different directions, zoom in + out). My actual question: Is it possible to apply these four filters in round-robin order to an undefined amount of input images? I hope you understand what I mean. [21:16:02 CET] <misterunknown> My command so far: [21:16:31 CET] <misterunknown> ffmpeg -framerate 1/10 -r 30 -pattern_type glob -i '*.jpg' -c:v libx264 -filter_complex "scale=8000x4000,zoompan=d=300:z='min(zoom+0.00035,1.5)':x='if(gte(zoom,1.5),x,x+3/a)':y='if(gte(zoom,1.5),y,y+3)" slideshow.mp4 [21:16:45 CET] <misterunknown> I read already about the -map parameter but only found explanations for 1:1 mapping. [23:59:26 CET] <djk> I'm trying to stream to Facebook at 720p but not sure where that is set? It is working but only 480 [00:00:00 CET] --- Sun Nov 25 2018

1 0

ffmpeg-devel.log.20181123
by burek 24 Nov '18

24 Nov '18

[00:24:08 CET] <cone-664> ffmpeg 03Andreas Rheinhardt 07master:e895b800fe27: cbs_h2645: Avoid memcpy when splitting fragment [05:46:36 CET] <sinclair> hi, im looking for information on piping video frames (encoded as png's) into FFMPEG, and having the resulting video piped to out. [05:47:28 CET] <sinclair> I have a fixed (known) number of frames, and what im sorta not sure about is having ffmpeg signal its finished encoding after processing the last frame. [05:47:51 CET] <sinclair> also, not sure about delimiting frames as they are piped into ffmpeg. [05:48:13 CET] <sinclair> is anyone able to provide assistance here? [05:58:41 CET] <jamrial> sinclair: see https://trac.ffmpeg.org/wiki/Slideshow [05:58:52 CET] <jamrial> also, this is a question for #ffmpeg, not this channel [05:59:12 CET] <sinclair> jamrial: i can't seem to talk in ffmpeg [05:59:52 CET] <sinclair> jamrial: but thanks for the link, is slideshow really the thing i need to use, each PNG is 1 frame in a 24fps video [06:10:19 CET] <sinclair> so yeah, it looks like slideshow is probably what i want, but is it possible to stream images in over STDIN? [06:22:17 CET] <Compn> ffmpeg might try to read it as individual files not a stream/video [08:43:02 CET] <j-b> morning [09:01:56 CET] <BradleyS> o/ [09:46:08 CET] <cone-045> ffmpeg 03Andrey Semashev 07master:fa08345e882c: lavf/dashenc: Fix segment duration overflow on fine time bases. [10:01:41 CET] <LigH> Hi. [10:02:40 CET] <LigH> Compiling ffmpeg in MSYS2/MinGW fails for configuring tesseract with undefined references to ZSTD_ symbols. [10:04:31 CET] <LigH> in libtiff.a [10:08:22 CET] <durandal_1707> we do no use libtiff [10:09:33 CET] <LigH> tesseract may? [10:09:51 CET] <LigH> But on its own, tesseract compiles. [10:11:12 CET] <LigH> I'll remove existing tesseract libraries and package conf files, force a recompile... [10:20:47 CET] <Gag_H> DC? [10:23:24 CET] <LigH> Nope. [10:23:36 CET] <LigH> Forced recompiling of tesseract: passes. [10:23:51 CET] <LigH> Configuring ffmpeg incl. libtesseract: Fails. [10:25:07 CET] <LigH> Brief head and tail of config.log: https://pastebin.com/HMEgbU3y (exceeds 512 KB, max. size of pastebin) [10:25:31 CET] <LigH> I could offer a ZIP archive of several log files... [10:26:55 CET] <durandal_1707> could you compile tesseract without tiff support? [10:30:55 CET] <LigH> I don't know how that may impact the compilation of everything else in media-autobuild_suite. So I will propose that in this project, to give their maintainers a chance to inspect it. [10:31:49 CET] <LigH> OK, I see it has been reported already. [10:31:52 CET] <LigH> https://github.com/jb-alvarado/media-autobuild_suite/issues/981 [10:31:58 CET] <LigH> So I guess it will be fixed. [10:32:09 CET] <LigH> Thanks, bye for now. [10:51:01 CET] <LigH> durandal_1707, success: --extra-libs=-lzstd [11:56:26 CET] <thardin> is there some reliable way to count the number of samples in a stream in a muxer? st->duration doesn't seem to come out right [11:56:43 CET] <thardin> or, number of frames I guess [11:56:57 CET] <nevcairiel> before muxing, or afterwards? [11:57:20 CET] <nevcairiel> duration is of course timestamp-bound, so gaps would offset it [11:59:00 CET] <thardin> hum [12:03:49 CET] <thardin> ah I need to convert between sample_rate and st->time_base [12:06:07 CET] <thardin> avpriv_set_pts_info() to the rescue [12:12:05 CET] <thardin> is there a flag for saying "extradata will take a little while, don't write headers immediately" from encoder to muxer? [12:12:39 CET] <nevcairiel> no, thats not really a generic feature we can do [12:12:45 CET] <thardin> I see side data at least [12:13:05 CET] <nevcairiel> its very specific to some codecs and muxers [12:16:22 CET] <thardin> I guess I'll signal AV_PKT_DATA_NEW_EXTRADATA and just always rewrite extradata on the end in the muxer [12:25:04 CET] <thardin> it seems the extract_extradata bsf is related [12:29:16 CET] <nevcairiel> thats for demuxing [12:29:27 CET] <nevcairiel> to get extradata from a stream without global headers [12:30:08 CET] <thardin> actually, libavcodec/pngenc.c uses this [12:30:49 CET] <nevcairiel> oh i meant the bsf [12:32:02 CET] <thardin> right [12:32:44 CET] <thardin> ok, so that side data gets picked up in apngenc.c and is not automagically put into codecpar as part of the parameter probing shenanigans [12:33:07 CET] <thardin> not so strange for apng since it is only available on the end. oh well [12:42:58 CET] <thardin> it werks [15:52:06 CET] <Holter> Hi [15:53:45 CET] <Holter> I think I found a small bug in mpegtsenc.c. [16:02:01 CET] <Holter> Changing the type will fix it. [16:02:34 CET] <Holter> Can anybody fix it? [16:06:19 CET] <jamrial> Holter: send a patch to the mailing list with the fix, or open a ticket on https://trac.ffmpeg.org/ with details about the bug [16:07:32 CET] <Holter> ok, thanks. [16:46:32 CET] <thardin> is it possible to tell whether a stream's duration is an estimate or not? [16:50:51 CET] <thardin> it's kind of crappy that it doesn't get filled in correctly when transcodinge [17:02:22 CET] <atomnuker> BBB: videolan's shitlab instance logged me out after 3 hours [17:03:07 CET] <atomnuker> and I stopped charging my phone after I found out it would stop beeping at every piece of mail I got if it had no battery [17:07:27 CET] <atomnuker> also could you voice some sanity regarding what unlord's suggesting [17:07:48 CET] <atomnuker> he's adding back the hell that msac.c used to be with the anti-optimization which was LOTS_OF_BITS [17:08:05 CET] <atomnuker> and snake oil placebo tier likely/unlikely optimizations [17:15:47 CET] <BBB> ... [17:16:48 CET] <BBB> look, for each of these optimizations, I think we can look at them individually if you want to go that way [17:17:00 CET] <BBB> the way I look at it is that if it gets faster, I'm ok [17:17:14 CET] <BBB> don't say shitlab, just tune your preferences to not send you email [17:17:56 CET] <atomnuker> yeah but I can't disable dual factor [17:17:58 CET] <BBB> and if you think some optimization is not right ("snake oil"), let's measure the improvement of it in isolation [17:20:54 CET] <jamrial> BBB: clean, maintainable no spaghetti ifdeffed code should be a priority as well [17:21:01 CET] <jamrial> speed is great, but when you focus too much on it and let it go wild you end up with swscale and swresample :p [17:21:23 CET] <BBB> fair enough [18:28:48 CET] <BtbN> philipl, I just had a really messy idea that I'm not sure I'd want merged. A hidden flag somewhere on CUDA frames that contain nvdec mapped surfaces that have been deinterlaced. And then a filter that separates them into two (still mapped) frames. [19:17:32 CET] <philipl> BtbN: so two deinterlaced frames carried by one avframe? [19:17:51 CET] <philipl> and then the filter splits them. [19:18:35 CET] <philipl> That doesn't seem terrible to me, especially if the frame still works for ignorant filters or outputs. You'd get half rate without the extra filter. [19:19:08 CET] <BtbN> Yeah, something like that. Just can't come up with a clean idea to have both frames in one in a somewhat clean way. [19:19:43 CET] <BtbN> It crosses library boundaries, so can't use our private storage space. [19:20:11 CET] <philipl> Well, storing both frames can be fudged. for semi-planar you can use plane 2 and 3 directly. [19:20:22 CET] <philipl> and any real world content will be semi-planar [19:20:41 CET] <philipl> Then it's a question of where the flag goes. [19:21:45 CET] <philipl> I wonder if the cuvid deinterlacer even works for 444 interlaced content. [19:21:51 CET] <philipl> i never bothered trying. [19:22:10 CET] <philipl> if it doesn't, then it's moot [19:22:57 CET] <philipl> we'd need something uglier for that. [19:24:25 CET] <philipl> You'd have to put the fuller buffer in plane 3 and then rely on the fact that the first frame's layout is the same and recover how to split the second frame's planes. [19:24:40 CET] <philipl> s/fuller/full second/ [19:29:16 CET] <BtbN> I'm not even sure if both planes can be mapped at the same time. [19:30:37 CET] <BtbN> Well, they probably can [19:30:42 CET] <BtbN> would be pretty useless otherwise [19:37:16 CET] <philipl> speak of the devil... [19:38:19 CET] <nevcairiel> decoder and encoder sharing a surface pool is always a bit iffy [19:38:25 CET] <nevcairiel> at least if your pool has to be fixed size [19:47:04 CET] <philipl> BtbN implemented a special pool that directly uses nvdec mapped planes but that has to be fixed. We put offer a filter that just copies frames using a dynamic pool to remove the restriction. [19:47:53 CET] <philipl> This would give you equivalent functionality to what we saw before he added the mapped frame pool [19:49:11 CET] <BtbN> The b frame depth should be known beforehand, so the pool size should (and I think already is?) adjusted accordingly. [19:50:18 CET] <BtbN> I don't see how being interlaced makes a difference there really [19:50:36 CET] <BtbN> unless it's paff vs. mbaff stuff. [19:53:15 CET] <philipl> could be that [20:20:27 CET] <orbea> Hi, not sure if anyone recalls, but little over a week ago I asked about a crash with alephone and ffmpeg. The alephone developer had time to follow through on the advice given here and now its fixed, thanks. :) https://github.com/Aleph-One-Marathon/alephone/issues/121 [20:25:24 CET] <philipl> BtbN: malakudi is the guy from devtalk with all the problems. [20:27:55 CET] <BtbN> Well, he's probably not wrong. Just using very unusual and untested combinations of stuff. [20:28:15 CET] <JEEB> 732 [21:32:57 CET] <BtbN> Bugreports ending in "Fix this bug!" are surely motivation to invest time into the issue. [21:33:07 CET] <nevcairiel> yeah as a baseline i ignore those [21:34:10 CET] <BtbN> Well, they all seem like legitimate issue, but the perceived level of entitlement is through the roof. [21:34:49 CET] <durandal_1707> gonna push truehd_core_bsf soon! [21:39:20 CET] <jamrial> BtbN: eh, "Please, fix this bug!" is not a demand [21:39:30 CET] <nevcairiel> ! makes it a demand [21:39:34 CET] <nevcairiel> in my book [21:44:55 CET] <durandal_1707> kickban bug creator for life, he spammer [22:05:18 CET] <cone-065> ffmpeg 03Paul B Mahol 07master:0279cb4f6985: avcodec: add truehd_core bitstream filter [22:26:42 CET] <BtbN> I'm confused about this hwdownload scale issue. Somehow I doubt his conclusion that hwdownload is at fault is correct. [00:00:00 CET] --- Sat Nov 24 2018

1 0

ffmpeg.log.20181123
by burek 24 Nov '18

24 Nov '18

[05:11:37 CET] <vincent42> hi all, I see two option of the libx264 encoder : keyint and min-keyint, the later is documented, but to make sure, the former defines the maximim gop length ? [06:18:47 CET] <tth> $ mplayer https://ve.media.tumblr.com/tumblr_pi5lhd3K7l1qigfjt.mp4 [06:18:58 CET] <tth> oups, wrong channel :) [06:23:12 CET] <pink_mist> lol [06:23:30 CET] <pink_mist> lost it when the butter melted [07:32:01 CET] <TheAMM> That was amazing [07:32:18 CET] <TheAMM> I've grown tired of rube goldberg machines but that had a fresh take on it [08:32:54 CET] <ejr> hi. i am using devuan ascii and previously used standarf ffmpeg from the repository. yesterday i installed the latest ffmpeg from source because i needed one of the newer functions (-to option working for cutting properly). but now, when i try to convert mp4s to mp3s, i get the following error when I run "ffmpeg -n -i "$1" -f mp3 "${1%.mp4}.mp3" [08:32:59 CET] <ejr> Automatic encoder selection failed for output stream #0:0. Default encoder for format mp3 (codec mp3) is probably disabled. Please choose an encoder manually. [08:33:02 CET] <ejr> Error selecting an encoder for stream 0:0 [08:33:08 CET] <ejr> any suggestions on how to fix this? [08:39:24 CET] <JEEB> ejr: you prpbably meant to copy the stream so -c copy :p by default ffmprg.c re-encodes with whatever is marked as default for a given format [08:39:50 CET] <JEEB> and you need LAME to actually encode mpeg-1 layer 3 aka mp3 [08:40:36 CET] <JEEB> which is why your build has no encoder for it most likely :p [08:40:40 CET] <pink_mist> well if he built from source, it might just be that he forgot to explicitly enable lame [08:40:58 CET] <pink_mist> (but copy would probably be smarter) [08:42:44 CET] <ejr> ok, i tried ffmpeg -n -i file.mp4 -c copy -f mp3 out.mp3 but that does not work either. pink_mist: right, i did only run ./configure, make and sudo make install. maybe i should just recompile it, but how can i enable lame in that process? [08:55:04 CET] <pink_mist> assuming you have lame and its headers installed, you can just pass --enable-libmp3lame [08:55:09 CET] <pink_mist> to ./configure [08:55:29 CET] <pink_mist> you should probably look through the ./configure --help output too to see what else you might want to enable [08:59:01 CET] <ejr> will do, thanks so far! [09:01:47 CET] <ejr> wow, i don't know 99 percent of the codecs and stuff listed there that can be enabled. what are the typical things i should enable for the occasional conversion to standard formats? or can one just enable everything? [09:03:11 CET] <JEEB> you generally only need to care about what you're encoding; decoders are generally internal [09:03:25 CET] <JEEB> so if you need to encode mp3, LAME. aac has an internal encoder [09:03:56 CET] <JEEB> basically you can check the list of encoders enabled with just ./configure and figure out what you'd need more, since you generally know what you need to encode to [09:04:34 CET] <ejr> ok. what about avi to mp4 and mkv to mp4/avi? does that need any extra decoders enabled? (sorry, i don't know much about decoders/encoders) [11:00:20 CET] <w1kl4s> mby question unrelated to ffmpeg, but someone here will prolly know [11:00:35 CET] <w1kl4s> wd purples or seagate purples for torrent server [11:06:18 CET] <pink_mist> never seagate [11:13:38 CET] <Alina-malina> ffmpeg -loop 1 -i kuck4_crop.jpg -i kuck4_crop.mp3 -c:v libx264 -c:a aac -strict experimental -b:a 192k -shortest -vf scale=-2 :720 kuckready.mp4 please someone help, this is taking so creaping long and using all resources, what i am doing wrong? [13:47:46 CET] <skident> Hey guys [13:49:20 CET] <skident> Does anybody know how to create an AVFrame (audio frame) manually in case I have more than 8 channels? Should I put all data into extended_data buffer or only data from channels greater than 8? [13:55:20 CET] <JEEB> https://ffmpeg.org/doxygen/trunk/structAVFrame.html#afca04d808393822625e09b… [14:02:48 CET] <skident> Yeah I have read this "documentation" and it is not clear for me [14:03:21 CET] <skident> for example I have an audio stream with 16 channels [14:03:38 CET] <skident> I see few options: [14:04:03 CET] <skident> 1. put first 8 channels into frame.data[] [14:04:15 CET] <skident> and other 8 channels into extended_data [14:04:31 CET] <skident> 2. put all 16 channels into frame.extended_data[] [14:04:37 CET] <skident> which one is correct? [16:20:05 CET] <JC_Yang> do any build configure options ensure us that the deprecated features are disabled in the public API? for example, the AVPacket.convergence_duration [16:21:13 CET] <c_14> Not that I know of, only thing you can do is build everything with -Werror=deprecated [16:21:29 CET] <c_14> which will abort compilation if it tries using a deprecated feature [16:23:04 CET] <JC_Yang> I mean, if I want to mute the warning, I have to undef FF_API_CONVERGENCE_DURATION, or maybe something that's upper layer which disable all of them for me, but how can I make sure that the build libraries(be it static or dynamic) do work with the the deprecation-free client codes? [16:24:16 CET] <c_14> you want to make sure that your client doesn't use deprecated functions? [16:44:56 CET] <JC_Yang> no, I mean the compatibility between my deprecation-free client codes and the pre-built libavcodec.a/libavformat.a. for example, if I undef FF_API_CONVERGENCE_DURATION, my client codes will utilize a AVPacket struct which is one field less, how can I make sure the prebuilt libavcodec.a use the same version of AVPacket? This is just one example, even it might not lead to real world problem, I doubt any other features incompatible will lead to hard to debug [16:44:56 CET] <JC_Yang> errors... I wonder whether my concerns are valid. iiuc, AVPacket is a public API struct and never malloc/free() inside the libraries so it won't lead to any problem, but does it mean there's no other areas we should pay attention to? [16:54:45 CET] <c_14> If you're linking statically you don't have to worry about that since the version you link against and the version you're running against are always the same, if you link dynamically you have to check the MAJOR/MINOR/MICRO versions. As long as the Major version of the library you're running against is the same as the major version you linked against the ABI should be compatible afaik [16:57:04 CET] <c_14> the version's are listed in libav*/version.h with defines which you use in your code and functions you compare them to at runtime [17:04:33 CET] <JC_Yang> okay, get it. [18:32:06 CET] <the_gamer> is there an easier way than this to just concatenate two videos? https://trac.ffmpeg.org/wiki/Concatenate#differentcodec i don't get it. just [0:v:0][0:a:0][1:v:0][1:a:0][2:v:0][2:a:0]concat=n=3:v=1:a=1[outv][outa] looks really complicated [18:33:46 CET] <kepstin> the_gamer: if you are only doing concat and have all inputs listed with -i on the ffmpeg command line, you can shorten that to "-filter_complex concat=n=3:v=1:a=1" (if omitted, it's just grab inputs and outputs in order from what's available) [18:34:06 CET] <furq> n=2 if you have two videos [18:34:13 CET] <the_gamer> is n=3 the number of the videos? [18:34:15 CET] <the_gamer> ok [18:34:16 CET] <furq> yes [18:34:30 CET] <furq> and v and a are the number of video and audio streams [18:34:48 CET] <the_gamer> where i would have thought 1 would be standard if omitted? [18:35:01 CET] <furq> iirc it defaults to v=1 a=0 [18:35:15 CET] <the_gamer> no audio is standard? [18:35:25 CET] <furq> [0:v:0] etc selects which streams you want to pass to the filter, but generally speaking they'll be in the right order by default [18:36:05 CET] <the_gamer> got it thank you :) [18:36:36 CET] <kepstin> if you have videos with different numbers of video or audio tracks (e.g. one video is dual audio), you'll probably have to explicitly specify the input tracks to use since it might pull the wrong ones otherwise. [18:36:52 CET] <kepstin> but if the inputs all match, then it's usually fine. [18:37:52 CET] <the_gamer> all 1 audio only, it is still rendering& [22:27:31 CET] <zinger> hi [22:27:43 CET] <zinger> I sometimes use a combination of youtube-dl and FFmpeg to download live streams. Is it somehow possible to download a live stream using only FFmpeg without using youtube-dl ?? [22:29:09 CET] <durandal_1707> no [22:29:31 CET] <durandal_1707> you need to get url [22:38:06 CET] <furq> zinger: youtube-dl -g will get the url and then ffmpeg may or may not be able to download it [22:38:16 CET] <furq> if you just want to pass custom options to ffmpeg or something [22:40:08 CET] <zinger> ok thanks [00:00:00 CET] --- Sat Nov 24 2018

1 0

ffmpeg-devel.log.20181122
by burek 23 Nov '18

23 Nov '18

[02:40:50 CET] <cone-184> ffmpeg 03James Almer 07master:3913d6f73484: x86/scene_sad: fix link errors when HAVE_X86ASM is not defined [07:37:01 CET] <cone-648> ffmpeg 03kjeyapal(a)akamai.com 07master:4903662a1da9: avformat/dashenc : Handled error from ff_http_do_new_request() cleanly [07:37:01 CET] <cone-648> ffmpeg 03kjeyapal(a)akamai.com 07master:ce0a75363405: avformat/http : Added check for valid URL context before calling shutdown [10:02:04 CET] <durandal_1707> nobody reviewed my patches? :( time to push them! [10:10:34 CET] <JEEB> w/25 [10:17:24 CET] <cone-648> ffmpeg 03Paul B Mahol 07master:7d70f0930784: avutil/avsscanf: do not use unsupported p suffix for hex variable [10:20:30 CET] <cone-648> ffmpeg 03Paul B Mahol 07master:027f032bbce9: avformat/movenc: fix size calculation in mov_write_eac3_tag() [13:06:21 CET] <j-b> 'morning [13:11:46 CET] <pross> libavlibc any day now [13:24:40 CET] <durandal_1707> pross: libavc [15:31:59 CET] <jya> j-b: hi "Sir" :) [17:53:19 CET] <nevcairiel> BBB: that blog says dav1d is "feature complete" in 8/10bit, does that include 422/444, or only profile 0? [17:55:05 CET] <BtbN> I still wonder how hwaccel for AV1 should work [17:56:10 CET] <BBB> nevcairiel: 422/444/400 work also [17:56:19 CET] <nevcairiel> BBB: cool [17:56:32 CET] <jamrial> BtbN: seeing hardware people are waiting for the new spec revision which will apparently be incompatible with the current one, it's not something we should worry about for the next two years :p [17:56:47 CET] <JEEB> lol [17:56:55 CET] <nevcairiel> they are breaking the bitstream already? [17:57:01 CET] <jamrial> looks like [17:57:03 CET] <atomnuker> yeah [17:57:13 CET] <nevcairiel> way to fail early [17:58:19 CET] <jamrial> it's not ideal, but better do it now before hardware is out in the wild than after that [18:11:10 CET] <BtbN> philipl, made a new round of coded-header releases. Now watch a new patch come up on the ML within a few hours :D [18:13:25 CET] <nevcairiel> personally I wouldnt even add every function some random person requests, if its not in use by ffmpeg =p [18:34:05 CET] <philipl> BtbN: [18:34:10 CET] <philipl> thanks [18:36:34 CET] <BBB> nevcairiel: it does everything except 12bit and ext-tile [18:36:45 CET] <BBB> nevcairiel: I don't know what ext-tile is or how to make samples using it [18:36:48 CET] <BBB> and 12-bit will be done [18:37:13 CET] <BBB> more important is to get 10bit avx2 and 8bit earlier optimizations, and finish the tiny pieces of avx2 missing (cfl_ac, z1/2/3, some super_res/scalable_mc stuff) [18:37:32 CET] <jamrial> isn't that what the tile list OBU is for? [18:37:40 CET] <nevcairiel> 12-bit is probably a bit niche, as always [18:38:40 CET] <jamrial> afaik, tile list obu stuff is not required for conformance [18:44:48 CET] <BBB> nevcairiel: well, I'd like to support it at some point [18:44:57 CET] <BBB> but not immediately, yes [18:45:02 CET] <BBB> it's not the most important thing [21:05:51 CET] <durandal_1707> looks like copy-trac3074 needs to be updated [21:11:17 CET] <jamrial> durandal_1707: if the changes from 027f032bbc are correct, then update the test [21:21:11 CET] <durandal_1707> well, it should actually use number of written bits instead of hacks [21:33:30 CET] <cone-664> ffmpeg 03Paul B Mahol 07master:97d1ee437bbf: avformat/movenc: get number of written bytes from bitstream writer [21:40:25 CET] <durandal_1707> any more comments for truehd_core_bsf patch? [22:14:03 CET] <jamrial> durandal_1707: seems to work with the sample i tried. 29mb, 45 seconds long truehd+atmos sample into 20mb, and it decodes fine (same checksum since our decoder just ignores atmos anyway) [22:15:20 CET] <nevcairiel> i dont suppose you can test if hardware still likes it? maybe bitstreaming the resulting file to some receiver? [22:15:58 CET] <jamrial> i don't have that kind of hardware :( [22:19:05 CET] <atomnuker> why is thilo wanting to stir shit now? [22:20:35 CET] <durandal_1707> we need more bureaucracy [22:21:28 CET] <jamrial> atomnuker: what? [22:21:40 CET] <atomnuker> durandal_1707: lets bring the worst of the real world online [22:23:28 CET] <atomnuker> jamrial: he'll spam the email addresses of every subscriber pointlessly asking to participate in a survey [22:25:23 CET] <durandal_1707> hey, that is not spamming, its information gathering [22:34:47 CET] <atomnuker> >every email survey ever sent [00:00:00 CET] --- Fri Nov 23 2018

1 0

ffmpeg.log.20181122
by burek 23 Nov '18

23 Nov '18

[00:01:52 CET] <TheRock> can i compile ffmpeg-4.1 with openssl 1.0.2q? [00:02:22 CET] <TheRock> LINK : fatal error LNK1181: cannot open input file 'crypto.lib' [00:02:31 CET] <TheRock> this file doesn't exist at me pc [04:25:51 CET] <cocktail> Can I put mp3 in ogg container without transcoding? [05:45:42 CET] <cocktail> How can I know which codecs a muxer supports? [05:45:54 CET] <cocktail> Is there a command-line option for that? [05:55:06 CET] <cocktail> https://wiki.xiph.org/index.php/MIMETypesCodecs tells me that ogg doesn't support mp3. [06:20:11 CET] <Renari> Hey guys I'm having an issue with metadata being corrupted when converting from wav files to flac files. [06:21:12 CET] <Renari> I'm on ffmpeg N-88804, I originally was using a batch script in windows and read that it's likely command prompts fault. However using the script in powershell ended with the same results. [06:21:39 CET] <Renari> Rather running the command in powershell, not using the batch script. [06:25:28 CET] <Renari> e.g. before: €ôûô¡ëìüÌ, after: ýCýýýEýýý@ýýýýý[ýk; [06:26:05 CET] <Renari> The command I'm running is pretty basic: ffmpeg -i filename.wav -c:a flac "filename.flac" [13:28:52 CET] <tombb> hi all! was wondering if it's possible to use ffprobe to print a list of the audio levels in frames for every audio stream [13:29:00 CET] <tombb> what would be the best way to approach this? [13:29:57 CET] <remlap> Hi anyone successfully built ffmpeg with nvenc and cuda on Ubuntu 18.10? [16:14:50 CET] <TheRock> "avpriv_snprintf" not found in avutil-56.dll [16:14:56 CET] <TheRock> after I compiled ffmpeg [16:16:06 CET] <TheRock> -/configure --disable-everything --extra-version=TS --disable-debug --enable-shared --enable-runtime-cpudetect --toolchain=msvc --enable-openssl --enable-decoders --enable-demuxers --enable-parsers --enable-protocol='file,https,http,ftp,ftps' --enable-stripping --strip --extra-libs='advapi32.lib gdi32.lib user32.lib Ole32.lib' [16:34:36 CET] <TheRock> according to dependency walker avpriv_strtod, avpriv_snprintf is not exported [16:35:17 CET] <furq> isn't that only defined for mingw [16:37:00 CET] <TheRock> i'm building with MSYS [16:37:07 CET] <furq> yeah but you're building with msvc [16:37:16 CET] <TheRock> I've configured MSVC [16:37:18 CET] <furq> where are you actually getting that error [16:37:21 CET] <TheRock> already built successful [16:37:31 CET] <TheRock> but the those two exports are problematic [16:37:36 CET] <furq> problematic how [16:37:55 CET] <TheRock> they're not exported, but required on programm startup [16:38:20 CET] <furq> if you get that error when you run ffmpeg then that's maybe a bug [16:38:35 CET] <TheRock> i tried to run ffprobe, ffmpeg binaries they work perfect [16:38:59 CET] <TheRock> i'm actually not sure what causes it [16:39:10 CET] <furq> what causes this error message [16:40:22 CET] <furq> is this your own code that's linked against the ffmpeg libs [16:40:31 CET] <furq> and if so are you trying to call avpriv_snprintf directly [16:42:11 CET] <TheRock> It's not my own code, i'll check for references in the third-party library now (uses ffmpeg as backend) [16:42:40 CET] <TheRock> https://ffmpeg.org/pipermail/libav-user/2016-September/009616.html [16:42:52 CET] <TheRock> > -Dstrtod=avpriv_strtod -Dsnprintf=avpriv_snprintf [16:42:52 CET] <TheRock> > -D_snprintf=avpriv_snprintf -Dvsnprintf=avpriv_vsnprintf [16:43:20 CET] <TheRock> may that help? [16:43:29 CET] <furq> i doubt it [16:44:08 CET] <furq> as far as i can tell from configure, those symbols only exist if you build with mingw [16:45:17 CET] <TheRock> $ echo $PATH [16:45:17 CET] <TheRock> .:/usr/local/bin:/mingw/bin:/bin:/c/Program Files/Microsoft Visual Studio 14.0/ [16:45:19 CET] <furq> or if you build with msvc < 15 [16:45:22 CET] <TheRock> maybe that's an issue? [16:45:32 CET] <furq> honestly i've never touched msvc [16:46:17 CET] <TheRock> i use VS 15, but g++, gcc is available in msys [16:46:33 CET] <TheRock> eventually it mixes some mingw stuff in he build [16:47:35 CET] <furq> my guess is the thing you're linking against libavutil expects msvc 13 and so it's trying to call avpriv_snprintf directly [16:47:50 CET] <furq> which it shouldn't be doing [16:48:06 CET] <furq> but like i said i've never touched msvc [16:55:05 CET] <TheRock> yeah i'm gonna look into it [17:00:24 CET] <TheRock> The project itself that uses ffmpeg doesn't use avpriv_ functions [17:12:22 CET] <TheRock> i unset now everything related to mingw, gcc, g++, etc. giving it another try. [17:16:49 CET] <rmbeer> hello... [17:17:51 CET] <rmbeer> how to make a command for extract the video stream of the first video and apply a list of music in succession?... [17:18:37 CET] <rmbeer> i can't found in the internet... [17:30:03 CET] <rmbeer> The only possible way is to create a concatenated audio file before joining it to a video?... [17:31:00 CET] <TheRock> I figured out avresample.dll (ffmpeg) tries to call from avutil.dll (ffmpeg) "avpriv_snprintf" [17:31:08 CET] <TheRock> so, it's probably a bug [17:31:26 CET] <TheRock> avutil doesn't export those functions [17:32:52 CET] <TheRock> and ffprobe.exe & ffmpeg works since avresample.dll is not used [17:33:09 CET] <TheRock> so i'll have to fix it in the ffmpeg src :| [17:35:32 CET] <furq> uh [17:35:36 CET] <furq> that build shouldn't have avresample [17:36:14 CET] <furq> are you sure all these dlls came from the same place [17:36:40 CET] <TheRock> yep [17:36:59 CET] <furq> you shouldn't even have avresample without --enable-avresample [17:37:02 CET] <furq> and you shouldn't be using that [17:37:11 CET] <furq> that's just for compat with libav [17:37:56 CET] <TheRock> there is a avresample-4.dll after make install [17:38:07 CET] <TheRock> Libraries: [17:38:08 CET] <TheRock> avcodec avformat swresample [17:38:08 CET] <TheRock> avdevice avutil swscale [17:38:08 CET] <TheRock> avfilter [17:38:15 CET] <furq> yeah that's swresample [17:38:18 CET] <TheRock> it's not included in configure [17:38:19 CET] <furq> that's a different lib [17:38:22 CET] <TheRock> but it exists [17:38:26 CET] <furq> what version is this [17:38:32 CET] <TheRock> yes, i know swresample is another lib [17:38:37 CET] <TheRock> ffmpeg 4.1 [17:38:47 CET] <JEEB> avresample is still buildable if you enable it [17:38:52 CET] <JEEB> I don't think it was ever enabled by default [17:38:56 CET] <furq> yeah but his configure line doesn't enable it [17:39:01 CET] <furq> unless --disable-everything is screwing something up [17:40:03 CET] <TheRock> --disable-everything builds also d3d11va, dxva2, etc. [17:40:22 CET] <TheRock> probably because of --enable-decoders [17:40:26 CET] <JEEB> that's probably solvable with --disable-autodetect [17:40:40 CET] <JEEB> that way you need to enable stuff like zlib etc [17:40:45 CET] <JEEB> and d3d11va/dxva2 [17:41:20 CET] <rmbeer> is this correct for concatenate 4 musics? -lavfi "[0:a] [a1];[1:a] [a2];[2:a] [a3];[3:a] [a4];[a1][a2][a3][a4] concat=n=4;a=4[outa]" [17:41:40 CET] <furq> uh [17:42:03 CET] <furq> just [0:a][1:a][2:a][3:a] [17:42:06 CET] <JEEB> yup [17:42:15 CET] <furq> unless you're actually doing some filtering that you omitted [17:42:31 CET] <TheRock> i just looked in another configuration, avresample is built although not listed in the configration overview [17:42:36 CET] <TheRock> I never enabled it [17:42:51 CET] <TheRock> with the configuration line, which i wrote above [17:42:57 CET] <JEEB> check with current master, and file a bug on trac if it still does [17:43:28 CET] <JEEB> since I for example will not start debugging the shell script right now [17:43:47 CET] <TheRock> i'll wait until the current build finished [17:44:40 CET] <rmbeer> i use this but can't work: ffmpeg -i f1.mp3 -i f2.mp3 -i f3.mp3 -i f4.mp3 -lavfi "[0:a][1:a][2:a][3:a]" outmusic.mp3 [17:44:52 CET] <TheRock> avresample is okay for me, but it asks for avpriv_snprintf from avutil, which is not exported there [17:45:02 CET] <JEEB> rmbeer: you still need the concat [17:45:17 CET] <JEEB> it's just that you had that other part that renamed 0:a to a1 :P [17:45:18 CET] <JEEB> etc [17:45:43 CET] <JEEB> TheRock: yea but it still sounds fishy :P [17:46:13 CET] <rmbeer> "[0:a][1:a][2:a][3:a] concat=n=4[outa]" -> Stream specifier ':a' in filtergraph description [0:a][1:a][2:a][3:a] concat=n=4[outa] matches no streams. [17:48:07 CET] <JEEB> sounds like something is either lacking or there that makes ffmpeg.c or lavfi misparse it [17:48:49 CET] <sybariten> oh hai [17:49:03 CET] <JEEB> ohai [17:50:06 CET] <sybariten> i'm a bit confused by ffmpegs syntax. First i tried using an "alac" codec for some problematic mts files, just to get the audio, but that didnt work for my purpose. Syntax was ffmpeg -i ./00264.MTS -vn -acodec alac 00262.m4a ... now i want to try "Native FFmpeg AAC Encoder" as it's called on an ffmpeg wiki [17:50:42 CET] <sybariten> but the syntaxc there, in the example, is ffmpeg -i input.wav -c:a aac -b:a 160k output.m4a ... how come theres no "-acodec" used? Is there a shorthand? [17:50:42 CET] <JEEB> instead of acodec -c:a aac (same for alac, "acodec" is just the old way of defining it) [17:51:12 CET] <JEEB> so you have -c for all streams, -c:v for video, -c:a for audio etc [17:51:25 CET] <JEEB> and :TYPE can be used generally with options [17:51:30 CET] <JEEB> or other selectors [17:51:39 CET] <sybariten> okay so i could use either -c:a aac , -c:a alac , -acodec aac and -acodec alac ? [17:51:43 CET] <JEEB> yes [17:52:02 CET] <JEEB> so you can have -maxrate:v:3 which means that you want to set the 4th video track's maxrate :P [17:52:04 CET] <sybariten> OK cool i'll go with the older way then as it's more ... uh... elaborate or whatever the english word is :) [17:52:46 CET] <rmbeer> then this not have solution?... Dx [17:52:55 CET] <sybariten> And what is the correct container, by the way, if i'm "transcoding" .mts video to something using aac (just audio then i guess) [17:53:12 CET] <JEEB> generally you may want to use mp4/m4a [17:53:19 CET] <sybariten> aha. Thanks! [17:53:22 CET] <JEEB> but it depends on if you need support on some plastic boxes [17:53:31 CET] <JEEB> although most plastic boxes take exactly that in [17:53:34 CET] <JEEB> for audio only [17:53:41 CET] <sybariten> hardware? Nope. I'm having Adobe Premiere Pro problems [17:53:45 CET] <JEEB> ok [17:54:01 CET] <JEEB> if it's audio only might as well go for raw audio in mov or something :D [17:54:20 CET] <JEEB> if lossless audio codecs are not supported [17:54:22 CET] <sybariten> Well... it's 28 gb of video [17:54:38 CET] <JEEB> yea, but audio is not exactly the largest part of that most likely? [17:54:42 CET] <sybariten> So i was just trying to economize a bit [17:54:51 CET] <sybariten> nah, true that, audio is often a tenth or soemthing maximum [17:55:18 CET] <sybariten> So what was your idea with mov? put just audio into mov? [17:55:33 CET] <JEEB> yea, raw audio in mov sounds like something that those things can ingest [17:55:38 CET] <JEEB> not that I know :P [17:55:51 CET] <sybariten> ok [17:56:17 CET] <sybariten> I'd put raw audio in wav.. but i'm not sure if you can say that that is a container [17:56:25 CET] <JEEB> yup [17:56:30 CET] <JEEB> wav should work as well [17:59:01 CET] <TheRock> ok guys [17:59:13 CET] <TheRock> it didn't build it this time [18:01:46 CET] <rmbeer> i fixed it with 'concat' command... [18:08:58 CET] <TheRock> that avresample/swresample stuff is not important for video decoding, right? [18:14:17 CET] <furq> no [18:14:27 CET] <furq> i'm not sure why it's being built at all with that configure command [18:16:33 CET] <TheRock> it isn't anymore [18:16:39 CET] <TheRock> i have created a complete new build [20:02:55 CET] <giaco> is there any dash mpd expert here? I am manually editing the stream.mpd file but I'm not achieving what I need, but it should be easy. I have a 12 minutes dash stream chunked in 5 seconds files, the original mpd file is a 1 period, 1 segment template barebone audio descriptor. What I would like to do is edit it so that it would play, according to an index value, the Nth minute. So if index is 5 the generated stream.mpd would pla [20:02:55 CET] <giaco> y, starting from the same chunks, from minute 5:00 to 6:00 displaying a duration of 1 minute [22:44:19 CET] <TheRock> I see that ffmpeg links to bcrypt. is there a way to avoid it? [22:46:12 CET] <TheRock> http://www.ffmpeg.org/doxygen/4.1/random__seed_8c_source.html [22:46:37 CET] <TheRock> through ./configure? [22:46:58 CET] <TheRock> I build on WIN7 (which has bcrypt), but this doesn't run on XP [22:47:28 CET] <JEEB> I think XP support was at some point specifically dropped for that [22:48:10 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=9b121dfc32810250938021952a… [22:48:55 CET] <JEEB> basically the threading stuff was switched in december 2017 that only vista+ threading APIs would be utilized and thus xp was no longer supported [22:49:12 CET] <TheRock> ah, ok [22:49:27 CET] <TheRock> gonna drop xp support then [00:00:00 CET] --- Fri Nov 23 2018

1 0

ffmpeg-devel.log.20181121
by burek 22 Nov '18

22 Nov '18

[00:35:30 CET] <cone-577> ffmpeg 03Peter Ross 07master:7cda7d217cd0: api-h264-slice-test: use av_be2ne16 instead of ntohs [00:35:31 CET] <cone-577> ffmpeg 03Peter Ross 07master:3fc7b69496fd: fate-api-h264-slice: use the heap for nal buffer [00:35:32 CET] <cone-577> ffmpeg 03Michael Niedermayer 07master:77bf85515e59: avcodec/truemotion2: Check huffman code max bits [10:11:58 CET] <j-b> 'morning [12:00:46 CET] <durandal_1707> thardin: it is not rounding and nothing that put_bits can do, previous code allocated too small number for worst scenario [12:01:32 CET] <durandal_1707> and it would overwrite exactly 2 bits, causing flush bits to assert [12:12:40 CET] <kierank> kb fflogger [13:30:53 CET] <cone-604> ffmpeg 03Peter Ross 07master:1ee4b4006be6: avfilter/graphmonitor: use SIZE_SPECIFIER for size_t type [13:34:07 CET] <thardin> durandal_1707: k, mostly just a thought [13:41:53 CET] <cone-604> ffmpeg 03Andrey Semashev 07master:322e53f8aebb: lavf/dashenc: Add DASH segment type auto and make it the default [13:41:53 CET] <cone-604> ffmpeg 03Andrey Semashev 07master:281a21ed5084: lavf/dashenc: Add support for format-specific file extensions. [13:41:53 CET] <cone-604> ffmpeg 03Andrey Semashev 07master:d290bb0c5404: lavf/dashenc: Fix AVDictionary leaks in case of various init errors. [14:52:03 CET] <smbz> Looking at #5023, it seems like the problem is related to ffio_ensure_callback() [14:52:34 CET] <smbz> Anyone know if ffio_ensure_callback() is supposed to work mid-stream, or just at the beginning (for probing and headers)? [14:55:53 CET] <smbz> *ffio_ensure_seekback... [16:00:05 CET] <jamrial> jkqxz: https://people.xiph.org/~tterribe/av1/samples-all/ [16:17:21 CET] <durandal_1707> where i can read about dec3 atom, specifically for mp4/mpv? [16:24:42 CET] <nevcairiel> durandal_1707: https://www.etsi.org/deliver/etsi_ts/102300_102399/102366/01.03.01_60/ts_10… page 202 [16:28:39 CET] <durandal_1707> nevcairiel: thanks [16:54:17 CET] <durandal_1707> fflogger drinks too much [16:59:56 CET] <kierank> durandal_1707: learned from you [17:01:24 CET] <kierank> spamming the channel all the time [17:01:24 CET] <kierank> now needs to get twitter and troll there [17:13:36 CET] <durandal_1707> is there any open source tool that parses dec3 correctly? [17:14:24 CET] <nevcairiel> most of those audio boxes are just pointless and not required for decoding [17:14:24 CET] <nevcairiel> so maybe not [17:14:24 CET] <durandal_1707> but stupid quicktime depends on it [17:54:53 CET] <durandal_1707> does anybody with windows 10 can try to remux one file with ffmpeg to see what windows player reports for audio? [18:03:07 CET] <jamrial> sure [18:09:14 CET] <durandal_1707> remux https://cloudstore.zih.tu-dresden.de/index.php/s/PHoOt67flM8jep5 to mp4 with -c copy [18:29:21 CET] <jamrial> durandal_1707: it detects it as six channels [18:29:37 CET] <jamrial> but so does ffmpeg when i read the resulting file [18:30:01 CET] <durandal_1707> 5.1 ? and not 6.0 ? [18:30:26 CET] <jamrial> ffmpeg says 5.1(side) [18:30:47 CET] <durandal_1707> what says windows player? [18:30:51 CET] <jamrial> the windows "Movies & TV" app just says 6 channel [18:30:57 CET] <jamrial> with the mkv it says 8 channel [18:31:13 CET] <nevcairiel> dont expect too much detail from typical players [18:31:49 CET] <durandal_1707> eh, mov muxer needs to be fixed to correctly interpret channel maps from eac3 [18:33:11 CET] <durandal_1707> and demuxer needs fixing too [18:33:11 CET] <nevcairiel> hm some msvc builds a re also broken right now [18:33:11 CET] <nevcairiel> avsscanf.c(486): error C2059: syntax error: 'bad suffix on number' [18:34:45 CET] <nevcairiel> hex float syntax is apparently not supported everywhere [18:34:55 CET] <durandal_1707> its 0x1p [18:35:02 CET] <jamrial> mediainfo says this eac3 audio stream has atmos [18:35:44 CET] <durandal_1707> mediainfo uses eac3 bitstream to get it [18:42:17 CET] <durandal_1707> nevcairiel: so how to fix? just multiply it? [19:06:51 CET] <durandal_1707> using pow(2, DBL_MANT_DIG) fixed it [19:15:13 CET] <durandal_1707> who is movenc expert to review patch? [19:20:49 CET] <kierank> durandal_1707: derek [20:18:27 CET] <BBB> for those interested, first performance numbers of dav1d (av1 decoder) on j-b's website: http://www.jbkempf.com/blog/post/2018/dav1d-toward-the-first-release [20:32:44 CET] <durandal_1707> hmm i'm looking at extra stream from truehd that is supposed to be atmos, and i see no typical start of atmos header [20:58:38 CET] <atomnuker> finding new av1 samples on youtube is perilous [20:58:59 CET] <atomnuker> only new videos have them and only those popular enough [20:59:52 CET] <atomnuker> so you need to go into trending and start fishing [20:59:52 CET] <atomnuker> one look at that page is enough to make you vomit though [20:59:54 CET] <atomnuker> is that what normal people watch? its worse than what's on television [21:03:08 CET] <BBB> I think some john oliver videos have av1, that's not too bad [21:03:08 CET] <BBB> or maybe that was just the beta... [21:07:42 CET] <Easyfab> atomnuker here a list ( not sure it's updated recently ) with some AV1 videos mostly videos clips https://docs.google.com/spreadsheets/d/110F_knHDvfVwRR86LlGgMWf-DO27Oa0e1GE… [21:11:42 CET] <jamrial> i've found videos about fancy japanese cooking get av1 streams [21:30:11 CET] <durandal_1707> does anybody with windows 10 can confirm it can play atmos with all objects within truehd samples? [21:30:41 CET] <akravchenko188> jkqxz: hi. it is reminder. Did you have a chance to review amf patches from November 2? it would be really great to have them applied to continue development. [22:12:38 CET] <durandal_1707> JEEB: ok, there is atmos media foundation dll, but what it uses to understand truehd? [22:13:12 CET] <nevcairiel> i think it only does atmos in EAC3 [22:19:28 CET] <durandal_1707> me too, and it actually only bitstreams [22:19:57 CET] <durandal_1707> i do not have EAC3+ATMOS samples at all [22:20:08 CET] <nevcairiel> i think netflix streams that [22:20:26 CET] <durandal_1707> for free? [22:20:32 CET] <nevcairiel> of course not [22:31:05 CET] <durandal_1707> so no atmos for you, its black box [22:32:08 CET] <kierank> durandal_1707: there is tranformers [22:35:25 CET] <kierank> The sample [22:35:28 CET] <kierank> Eac3 I dunno [23:34:24 CET] <durandal_1707> only dolby media decoder have it, and that cost $$$$ [23:37:46 CET] <jamrial> <@durandal_1707> i do not have EAC3+ATMOS samples at all <--- the one you posted above doesn't? [00:00:00 CET] --- Thu Nov 22 2018

1 0

ffmpeg.log.20181121
by burek 22 Nov '18

22 Nov '18

[01:21:40 CET] <fling> Can't ffmpeg input from libarchive like mpv does? [01:43:07 CET] <JEEB> fling: no [01:43:13 CET] <JEEB> it has no module to use libarchive [03:53:22 CET] <twtduck> Hey people, I'm looking to transcode all the videos on my media server. What encoding and format would you recommend, to avoid losing any quality and make it easy for my server to decode the files? [03:53:54 CET] <twtduck> Storage is not a huge concern, but I do have a few hundred hours and only 8 terabytes of storage, so nothing excessive. [05:06:10 CET] <twtduck> bump [05:14:53 CET] <ariyasu> you will always loose some quality [05:15:10 CET] <ariyasu> and depends what sort of content it is [05:15:49 CET] <ariyasu> if it's scripted (movies) encoding them to x264 at a crf of 18 or 19 would be a good ish idea, and the audio to aac [05:16:27 CET] <ariyasu> you could also downscale them to like 720p or lower which would reduce the file sizes considerably [05:20:07 CET] <twtduck> Most of them are 1080p, which is already less than ideal for my 4k screens [05:20:46 CET] <twtduck> My server struggles when decoding 2-3 at a time, which is why I'm more concerned. [17:09:34 CET] <podman_> I'm having difficulty maintaining proper colors when extracting a jpg from a video using bt709 colorspace and the tv color range. Any thoughts on where to start? [17:12:04 CET] <podman_> it seems to be converting it to yuvj420p(pc) which i think is what's causing the problem [17:32:58 CET] <furq> podman_: use zscale if your build has it [17:34:16 CET] <furq> maybe just -vf format=yuv420p will work [17:34:46 CET] <furq> if you want to force everything then -vf zscale=rin=limited:r=limited:pin=709:tin=709:min=709,format=yuv420p [17:36:24 CET] <podman_> furq: thanks, i'll give that a try [17:56:57 CET] <podman_> furq: it looks like i don't have zscale, so i tired -vf scale=in_range=tv:in_color_matrix=bt709:out_range=limited:out_color_matrix=bt709,format=yuv420p. it doesn't seem to have any effect. the output always just seem to be Stream #0:0: Video: mjpeg, yuvj420p(pc) [17:58:06 CET] <furq> try out_range=full [17:58:26 CET] <furq> and then get rid of format=yuv420p [18:02:01 CET] <podman_> furq: sure thing [18:02:22 CET] <podman_> furq: no dice. [18:05:24 CET] <durandal_1707> podman_: mjpeg encoder supports only J, aka pc range [18:08:00 CET] <furq> limited to full range should look fine as long as the scaler is converting it properly [18:08:00 CET] <podman_> durandal_1707: interesting. is there any way i can shift the range then to make it match? [18:08:00 CET] <furq> that's what that last command should do [18:08:00 CET] <furq> it should also work by default [18:08:00 CET] <podman_> hmm, doesn't seem to be doing anything [18:08:46 CET] <podman_> to give you an idea of what i'm talking about: https://imgur.com/a/YIUs8O3 the top image is the jpg, and the bottom is a png. the png properly matches the input video [18:14:26 CET] <kepstin> hmm, that doesn't look like a color range issue (I'd expect it to either be dim/low contrast or have blown out highlights) [18:14:39 CET] <podman_> kepstin: yeah, seems like a color space issue maybe? [18:18:01 CET] <kepstin> Hmm. could be color matrix or primaries? Could be that it's mixed up between bt601/bt709 [18:18:49 CET] <kepstin> (how did you make that png?) [18:19:50 CET] <podman_> ffmpeg -ss 1 -i in_file.mp4 -vframes 1 out_file.png [18:22:18 CET] <ritsuka> kepstin: are you on a mac? [18:22:26 CET] <podman_> ritsuka: i am [18:22:47 CET] <ritsuka> when you play a video in chrome, does it have the same colors as in quicktime player? [18:24:38 CET] <podman_> yep, the issue is specifically with converting to jpeg though [18:24:38 CET] <kepstin> podman_: that's really weird, I'd expect that the png and jpeg both made by ffmpeg should have the same colours when viewed [18:24:38 CET] <podman_> the interesting thing is that if i convert the original video and don't specify the color primaries and color range, the output video displays the same color shift [18:24:45 CET] <podman_> so i have to force it to bt709 and tv color range [18:26:27 CET] <podman_> if i don't, the output looks like the jpg instead of the png [18:26:45 CET] <ritsuka> podman_: right, the chrome is color managed the same way as quicktime, that means that ffmpeg won't output the same image, because it uses a different transfer function [18:26:45 CET] <podman_> ritsuka: so why would the png look right? [18:27:56 CET] <ritsuka> good question, then chrome doesn't handle colors the same way as quicktime, and please ignore my message [18:28:56 CET] <podman_> ultimately the output jpg looks the same everywhere, it's just different than the input mp4. [18:46:10 CET] <podman_> with -vf scale=in_range=tv:in_color_matrix=bt709:out_range=tv:out_color_matrix=bt=709 the colors are the same, but the output is a bit dimmer [18:46:13 CET] <podman_> so, that's close [19:06:51 CET] <podman_> i guess maybe something is wrong with this specific video? [20:36:24 CET] <friendofafriend> I am compiling ffmpeg with --enable-static --disable-shared, but the resulting binary is still looking for shared object files. Am I misunderstanding what these flags do? [20:37:47 CET] <furq> those flags only affect the internal libs [20:38:09 CET] <furq> if you want a fully static build then you want something like --extra-ldflags="-static" [20:40:19 CET] <friendofafriend> Wonderful, I'll try compiling again. It only takes about a day on my Raspberry Pi. :P [20:40:55 CET] <friendofafriend> Thank you for your help. [20:41:08 CET] <furq> you can cross-compile from something less underpowered [20:42:33 CET] <pink_mist> like a 10 year old cellphone [20:42:48 CET] <pink_mist> (ok, that might not be the best thing to do it from :P) [20:43:33 CET] <Harzilein> pink_mist: i think i actually do have distcc on my lil'debi chroot :) [20:43:47 CET] <Harzilein> pink_mist: and it's >10y old [20:44:13 CET] <pink_mist> haha, nice :P [21:03:08 CET] <friendofafriend> That's very strange, when I add --extra-ldflags="-static", now configure complains that it cannot find mmal. [21:03:08 CET] <furq> i don't think you can static link mmal or omx [21:03:08 CET] <furq> they definitely don't provide static libs by default [21:07:39 CET] <foo> Strange, getting this AttributeError: module 'ffmpeg' has no attribute 'input' - with python-ffmpeg. [21:07:39 CET] <foo> Also this AttributeError: module 'ffmpeg' has no attribute 'Error' [21:07:39 CET] <foo> (Python errors) [21:21:05 CET] <safinaskar> how to remove black frames from stream? [21:22:02 CET] <safinaskar> they are absolutely black [21:23:54 CET] <ariyasu> crop [21:24:16 CET] <ariyasu> -vf crop=width:height:crop_from_left:crop_from_top [21:26:42 CET] <ariyasu> oh frames [21:26:51 CET] <safinaskar> ariyasu: yes [21:26:51 CET] <ariyasu> ignore me, i just woke up [21:29:56 CET] <safinaskar> so how?? the only way i am aware of is converting into raw video and then removing the frames using hand-written c++ program [21:30:05 CET] <safinaskar> so i probably have to do this [22:01:14 CET] <foo> disregard my error, looks like I get this error: https://paste.ofcode.org/pvSLTmbTigdxAuYTfJnpDD - with that code there... on a 64MB mp3 file. [22:20:10 CET] <giaco> hello [22:23:53 CET] <giaco> I have a general question about audio file formats. I have a folder containing a dash stream, so it has a stream.mpd, a init-stream0.m4s and many chunk-stream0-<number>.m4s. I am trying to create 5 minutes segments out of it. I already have a ffmpeg concat based solution, but I wonders if it could be done even faster. If I do "cat init-stream0.m4s chunk-stream0-00001.m4s chunk-stream0-00002.m4s ... > output.mp4" I can hear the [22:23:53 CET] <giaco> file, but to make 5 minute segments I need to start from the middle of the list such as "cat $(ls -p chunk-stream0-0002* | sort) > output.mp4" but this second version is not playable. Do you know why I face this difference? Thank you [22:24:43 CET] <BtbN> You forgot the init segment. [22:25:04 CET] <giaco> BtbN: is the init segment required even when catting starting from the middle? [22:25:13 CET] <BtbN> That's the whole point of it. [22:25:51 CET] <giaco> BtbN: this is really a good news! [22:26:06 CET] <giaco> But why? Which standard should I read to understand how it works? [22:26:35 CET] <BtbN> the dash one I guess [22:27:02 CET] <BtbN> and the mp4, mov, isobmff ones [22:29:58 CET] <giaco> BtbN: thank you, it works [22:30:31 CET] <giaco> oh dear, that was easy ... I feel so stupid it was obvious [00:00:00 CET] --- Thu Nov 22 2018

1 0

ffmpeg-devel.log.20181120
by burek 21 Nov '18

21 Nov '18

[08:26:22 CET] <pross> fun question: at what point should av_malloc be used instead of stack? how many kbytes? [17:40:32 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:56b188941fd3: avutil/pixfmt: Document chroma plane size for odd resolutions [17:40:33 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:83ffda78ddb1: swresample/swresample: Fix input channel count in resample_first computation [17:40:34 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:00c3f178a8e5: avcodec/diracdec: Prevent integer overflow in intermediate in global_mv() [17:40:35 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:1c05c6bd91f1: avcodec/diracdec: Change frame_number to 64bit as its a 32bit from the bitstream and we also have a -1 special case [17:40:36 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:b12e1618a39a: avcodec/qtrle: Check remaining bytestream in qtrle_decode_XYbpp() [17:40:37 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:a36d649ffce6: avcodec/mpegaudio_parser: Initialize poutbuf* [17:40:38 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:ed0f21a89a41: avcodec/shorten: Check verbatim length [17:40:39 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:bb070fc4684d: avcodec/shorten: Fix integer overflow in residual/LPC combination [17:40:40 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:b671ebfd61b6: avformat/mlvdec: read_string() received unsigned size, make the argument unsigned [17:40:41 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:898f83386354: avformat/nsvdec: Do not parse multiple NSVf [17:40:42 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:7ddcb02809f9: avcodec/snowdec: Fix integer overflow with motion vector residual [17:40:43 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:8717129d58e9: avcodec/vb: Check for end of bytestream before reading blocktype [17:40:44 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:db39404c1581: avcodec/hq_hqa: Check remaining input bits in hqa_decode_mb() [17:40:45 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:2d65c44f220c: avcodec/ra144: Fix undefined integer overflow in add_wav() [17:40:46 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:fb0cd972a704: avcodec/h264_refs: Document last if() in ff_h264_execute_ref_pic_marking() [17:40:47 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:6645783a3aa4: avcodec/dvdsubdec: Avoid branch in decode_run_8bit() [17:40:48 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:f2640f764894: avcodec/shorten: Fix bitstream end check in read_header() [17:40:49 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:edae5259c055: avcodec/zmbv: Update decomp_len in raw frames [17:40:50 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:79c018351e6e: avcodec/zmbv: Check that the decompressed data size is correct [17:40:51 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:49023147c7fc: avcodec/mpeg4videodec: Fix undefined shift in get_amv() [17:40:52 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:c2f02af69684: avcodec/dvdsubdec: Sanity check len in decode_rle() [17:40:53 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:604957c557df: avcodec/unary: Improve get_unary() docs [17:40:54 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:6ce3fee455d6: avformat/utils: Fix integer overflow in discontinuity check [17:40:55 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:dc13bac6deaa: avformat/utils: Never store negative values in last_IP_duration [17:40:56 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:e42f2eb165e7: avcodec/ra144: Fix integer overflow in add_wav() [17:40:57 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:89a8b69e9bf6: avcodec/h264_cavlc: Check mb_skip_run [17:40:58 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:d74cd4d69ab7: avcodec/mpeg4videodec: Fix typo in sprite delta check [17:40:59 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:bdd3c7af5ee1: avcodec/jpeg2000dec: Fix off by 1 error in JPEG2000_PGOD_CPRL handling [17:41:00 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:3049d6d8219b: avcodec/msrle: Check that the input is large enough to contain a end of picture code [17:41:01 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:6f84b1c458c6: avutil/integer: Fix integer overflow in av_mul_i() [17:41:02 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:d8ecb335fe48: avformat/flvenc: Check audio packet size [17:41:03 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:cf21323dd460: avcodec/mpegaudio_parser: Consume more than 0 bytes in case of the unsupported mp3adu case [17:41:04 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:0acb6f692ef2: avcodec/cavsdec: Propagate error codes inside decode_mb_i() [17:41:05 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:072891195110: avcodec/shorten: Fix integer overflow with offset [17:41:06 CET] <cone-652> ffmpeg 03Michael Niedermayer 07release/2.8:3ae8a4cf060e: avcodec/pngdec: Check compression method [19:26:06 CET] <durandal_1707> nobody can match with my skills! [19:27:12 CET] <kierank> durandal_1707: why [19:27:26 CET] <kierank> You are not cael [19:27:27 CET] <kierank> Carl [19:29:30 CET] <durandal_1707> None of my concern. [19:31:36 CET] <j-b> Can you do the opposite? [19:33:26 CET] <durandal_1707> what opposite? [19:40:50 CET] <j-b> create atmos metadata :) [19:41:23 CET] <j-b> also, do we render atmos correctly? [19:41:54 CET] <JEEB> pretty sure there's N different types of audio under the moniker "dolby atmos" [19:41:54 CET] <kierank> Yes, RE atmos [19:42:07 CET] <JEEB> just like dolby vision is actually N different things as well [19:42:08 CET] <JEEB> :D [19:43:31 CET] <durandal_1707> i stil not found software that understand atmos [19:45:25 CET] <durandal_1707> but iirc kostya said he looked at some binary... [19:46:29 CET] <JEEB> Description: Media Foundation Dolby Digital Atmos Decoders. [19:46:36 CET] <JEEB> DolbyDecMFT.dll [19:46:39 CET] <JEEB> maybe? [19:47:00 CET] <atomnuker> write a bsf to losslessly rotate jpeg images [19:48:37 CET] <JEEB> durandal_1707: seems to come with good ol' windows now? [19:48:48 CET] <JEEB> original file name: DolbyAtmosDecMFT.dll [19:49:10 CET] <durandal_1707> windows 7? [19:49:15 CET] <JEEB> 10 at least [19:49:16 CET] <JEEB> 1809 [19:53:26 CET] <durandal_1707> i do not think it decodes to additional channel? [19:53:35 CET] <JEEB> no idea what the flying fuck it does :D [19:55:45 CET] <j-b> the one from Xbox1? [19:57:45 CET] <durandal_1707> iirc there was some think like prohibiting atmos implementation to anything except allowed by dolby? [19:58:18 CET] <durandal_1707> i do not want to be sued by those guys [19:59:01 CET] <JEEB> they're not DTS [19:59:09 CET] <JEEB> and as long as you're not legally bound by NDA [19:59:20 CET] <JEEB> nobody gives a flying fsck [19:59:37 CET] <JEEB> (As long as your implementation is kosher and doesn't copy dolby code) [19:59:57 CET] <durandal_1707> it copies binary code [20:00:26 CET] <durandal_1707> variables are a, b, c, d, ... x, w, y, z [20:59:37 CET] <atomnuker> are the timestamps for the av1 chimera samples from netflix fucked? [21:00:05 CET] <atomnuker> they seem to restart once every 2 or so seconds [21:02:11 CET] <jamrial> atomnuker: yes, get the mp4 ones instead [23:36:26 CET] <cone-577> ffmpeg 03Paul B Mahol 07master:57815cfad5c5: avfilter/vf_overlay: fix crash with negative y [00:00:00 CET] --- Wed Nov 21 2018

1 0

ffmpeg.log.20181120
by burek 21 Nov '18

21 Nov '18

[06:29:24 CET] <pins> I need help understanding why ffmpeg will not convert a folder full of jpgs to folder\*.png [06:29:36 CET] <pins> this is what I'm using: for %%a in ("*.jpg") do ffmpeg -i "%%a" "converts\%%0d.png" [06:31:21 CET] <pins> nvm. :D [06:31:45 CET] <pins> for %%a in ("*.jpg") do ffmpeg -i "%%a" "converts\%%~na.png" [11:27:35 CET] <MarcoT> Hello, can you help me? I need to stream a mp4 file via HTTP, using just ffmpeg command line if possible (no ffserver). Can I? [11:28:11 CET] <MarcoT> I am using this command line but I have "tcp: Connection refused error" [11:28:18 CET] <MarcoT> ffmpeg -re -i testvideo.mp4 -c:v libx264 -preset veryfast -maxrate 3000k -bufsize 6000k -pix_fmt yuv420p -g 50 -an -f mp4 http://localhost:8090/test.mp4 [11:33:18 CET] <MarcoT> or can I even with ffserver? [15:21:40 CET] <Alina-malina> ffmpeg -f gdigrab -i desktop output.mp4 i am trying to grab the sscreen, but to be honest the quality is shit, is there a way how i can increase the quality? [16:19:59 CET] <Alina-malina> C:\ffmpeg\bin>ffmpeg -y -rtbufsize 100M -f gdigrab -framerate 60 -probesize 10M -draw_mouse 1 -i desktop -c:v libx264 -r 60 -preset ultrafast -tune zerolatency -crf 6 -pix_fmt yuv422p "video.mp4" ok seems i found it, but how to add some sounding? i mean i want sound to be captured from my speakers? [16:25:38 CET] <furq> don't ever use tune zerolatency [16:27:01 CET] <Mavrik> "Why do we even have that lever?" :) [16:41:24 CET] <Alina-malina> why? [16:41:48 CET] <Alina-malina> how to add sound there? [16:42:37 CET] <Alina-malina> struggling to add sound from my sound system [16:46:38 CET] <Alina-malina> ok i remove that tune zerolatency lol [16:46:40 CET] <Alina-malina> whats next? [16:46:43 CET] <Alina-malina> how to add sound [20:05:22 CET] <Alish> I have a .mkv file screencast, and in it at a specific time for some seconds I wanna add some text (explainig that there was an error). can someone guide me? [20:11:00 CET] Last message repeated 1 time(s). [20:20:02 CET] <Alish> is it possible to add a text a specified time for 2 seconds in .mkv files? [20:27:03 CET] <furq> Alish: https://ffmpeg.org/ffmpeg-filters.html#drawtext-1 and https://ffmpeg.org/ffmpeg-filters.html#Timeline-editing [20:27:57 CET] <furq> or you can just put it in a subtitle file and burn it in with -vf subtitles [20:34:52 CET] <Alish> furq: Thanks [20:45:22 CET] <ariyasu> yeah make a soft .srt sub file [20:47:03 CET] <Alish> ariyasu: Thanks. [21:00:34 CET] <Alish> There is no example for time duration. [21:00:47 CET] <Alish> furq: ^^ [21:16:08 CET] <Alish> it's working. can't I change the colors? [21:16:30 CET] <Alish> of subtitles and fonts of them? [21:46:14 CET] <pink_mist> you need ass rather than srt for that [21:46:24 CET] <pink_mist> and you need ffmpeg built with libass afaik [21:50:37 CET] <Alish> pink_mist: Thanks. [00:00:00 CET] --- Wed Nov 21 2018

1 0

ffmpeg-devel.log.20181119
by burek 20 Nov '18

20 Nov '18

[14:59:53 CET] <cone-807> ffmpeg 03Paul B Mahol 07master:cfd421226c67: avfilter: add chromashift filter [14:59:54 CET] <cone-807> ffmpeg 03Paul B Mahol 07master:6da439219d23: avfilter: add rgbashift filter [15:18:18 CET] <cone-807> ffmpeg 03Paul B Mahol 07master:277fd9c2ac72: avutil/avsscanf: use ptrdiff_t instead of off_t [16:14:23 CET] <durandal_1707> does truehd have core? [16:16:02 CET] <JEEB> no [16:16:09 CET] <JEEB> sometimes it has an AC-3 stream muxed next to it [16:16:20 CET] <JEEB> but it doesn't have a core/extension kind of thing like with DTS formats [16:17:37 CET] <nevcairiel> yeah the AC3 stream is entirely unrelated, its just compat shit for blu-rays [16:19:33 CET] <nevcairiel> some silly people try to demux the streams together, or even remux them into another container combined like that, luckily no format ever officially supported that [16:19:45 CET] <nevcairiel> and in mpeg-ts it has special extension markers that allows you to split it cleanly [16:21:53 CET] <durandal_1707> fflogger5 seems no longer coperative [18:21:19 CET] <cone-807> ffmpeg 03hwrenx 07master:bddcbb932f4c: configure: update api version of libxavs2 [18:25:22 CET] <durandal_1707> give some good idea for a/v filter [18:46:16 CET] <durandal_1707> A->A/V->V/A->V whatever [19:09:04 CET] <j-b> Dolby AC-4 decoding [19:09:29 CET] <JEEB> I've only seen AC-4 in those DASH-IF samples [19:09:34 CET] <JEEB> I wonder if it even exists anywhere else [19:14:05 CET] <durandal_1707> in lala land [19:15:16 CET] <atomnuker> TD-Linux: hasn't ac-4 started being used in atsc+1 or whatever? [19:26:13 CET] <TD-Linux> "used" [19:26:15 CET] <TD-Linux> "atsc 3" [19:27:19 CET] <JEEB> :D [19:27:26 CET] <JEEB> I've only heard of test airings [20:35:21 CET] <cone-807> ffmpeg 03Ard Oerlemans 07master:95107af31462: avcodec/libvpxenc: add VP8 temporal scalability config options [20:39:05 CET] <cone-807> ffmpeg 03James Zern 07master:e1f895838d50: libvpxenc,cosmetics: fix { placement [23:20:41 CET] <cone-807> ffmpeg 03Marton Balint 07master:972ded33c08a: avfilter/vf_freezedetect: add filter to detect frozen input [23:20:42 CET] <cone-807> ffmpeg 03Marton Balint 07master:e3a96309826d: avcodec/mpeg_er: fix clearing chroma blocks for 422 and 444 [23:41:35 CET] <cone-880> ffmpeg 03Marton Balint 07release/4.1:d4c5f515f090: avcodec/mpeg_er: fix clearing chroma blocks for 422 and 444 [00:00:00 CET] --- Tue Nov 20 2018

1 0

ffmpeg.log.20181119
by burek 20 Nov '18

20 Nov '18

[03:00:36 CET] <Buliarous> how would I extract subtitles from a VTS VOB concat like this: ffmpeg -i "concat:VTS_01_0.VOB|VTS_01_1.VOB|VTS_01_2.VOB|VTS_01_3.VOB|VTS_01_4.VOB" I have subtitles at stream 2 and 3 and want to create srt's from them [03:11:54 CET] <pink_mist> uh, I don't believe ffmpeg can create srts from vobsubs [03:12:05 CET] <pink_mist> vobsubs are basically pictures of text [03:12:33 CET] <pink_mist> so you need software that can do optical character recognition [03:12:43 CET] <pink_mist> ffmpeg is not that afaik [03:17:09 CET] <Hello71> it could give you a bunch of pictures [03:19:23 CET] <pink_mist> that's not valid srt though [03:20:23 CET] <Hello71> halfway there though [03:20:29 CET] <Hello71> well more like 20% of the way there [03:20:50 CET] <pink_mist> more like 5% I'd say [03:20:54 CET] <pink_mist> OCR is hard [03:33:41 CET] <Hello71> making ocr is hard but using ocr is not that hard [03:33:51 CET] <Hello71> hell, using ocr is usually easier than using ffmpeg [03:36:27 CET] <Buliarous> thanks pink_mist went with mkvmerge for this task, can always convert later [11:55:06 CET] <scramblez> Hi All, I downloaded 'https://www.youtube.com/watch?v=ig4g6oD9Als' but it will not render in a Mut@nt 51HD. The same box has no problem playing vp9 in webm, as long as they do not contain HDR metadata. I wanted to test using m2ts container, but copying over the streams fails. [11:57:05 CET] <scramblez> What options should I use to move it into m2ts? Will m2ts work with HDR? [12:00:05 CET] <scramblez> Here's the error I get: 'https://pastebin.com/gGj8Y60V' [12:02:16 CET] <scramblez> Ah! I removed the copy directive and it is now converting it. :-) [12:11:01 CET] <scramblez> ... but the box won't render it. :( [12:21:07 CET] <relaxed> scramblez: maybe because it's yuv420p10le [12:21:17 CET] <snatcher> >ffmpeg -tls_verify 1 -i 'https://ffmpeg.org/' >error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed [12:21:34 CET] <snatcher> why https validation failed? [12:22:26 CET] <relaxed> firefox lists https://ffmpeg.org/ as insecure [12:22:44 CET] <relaxed> oh wait! I'm wrong [12:23:29 CET] <scramblez> relaxed: yes, quite. It may be the box is not able to render 10bit - at all? Strangely, my laptop with a 10 year old radeon card is rendering the webm and m2ts seemingly without a problem. [12:25:12 CET] <relaxed> scramblez: that's because your laptop uses software decoders and your tv uses hardware, which only supports certain features [12:25:49 CET] <barc0d4> hi when I start webcam with mpv I get an error like this : [ffmpeg/demuxer] video4linux2,v4l2: Dequeued v4l2 buffer contains 613376 bytes, but 614400 were expected. Flags: 0x00012001. [12:25:59 CET] <scramblez> relaxed: you reckon it is the TV at fault here, rather than the Mut@nt set top box? [12:27:17 CET] <scramblez> Hmm ... I guess I could try connecting the laptop to the TV and see what it shows. [12:28:10 CET] <relaxed> yeah, I run hdmi from my nvidia card to my TV and with mpv I never have issues [12:29:00 CET] <scramblez> Question: how can I copy the 'Mastering Display Metadata' from the webm container to the m2ts? [12:29:46 CET] <scramblez> All the luminance data are not transfered over to the m2ts - is there some special option needed? [13:37:15 CET] <skident> hello [13:39:17 CET] <skident> I have a code which uses ffmpeg audio filters, it works fine on Windows, but the same code works incorrect on macos (sound is noisy). Does anybody know how to fix this? [13:40:42 CET] <durandal_1707> skident: show code [13:42:39 CET] <snatcher> seems >error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed ffmpeg-related, "openssl s_client -connect ffmpeg.org:443" >Verify return code: 0 (ok) [13:42:42 CET] <snatcher> any ideas? [13:46:10 CET] <skident> it is integrated into a project and I can't reproduce without a project. I need to write a simple playback code for testing my code without current project [13:47:02 CET] <skident> but unfortunately I haven't found examples of playback using ffmpeg (only portaudio) [13:47:40 CET] <durandal_1707> skident: see mpv audio lavfi code [13:54:46 CET] <skident> <durandal_1707> could you share a link? [13:59:08 CET] <durandal_1707> skident: mpv.io [15:12:46 CET] <scramblez> Hi All, I've transcoded 'https://www.youtube.com/watch?v=ig4g6oD9Als' from vp9 within webm to hevc within m2ts 'Video: hevc (Main 10) ([36][0][0][0] / 0x0024), yuv420p10le(tv), 3840x2026 [SAR 1:1 DAR 1920:1013], 24 fps, 24 tbr, 90k tbn, 24 tbc'. However, the HDR luminance etc side data were not transfered through, so the video looks washed out. How can I also transfer the HDR metadata to m2ts? [15:13:43 CET] <scramblez> This is the data I'm talking about: 'Mastering Display Metadata, has_primaries:1 has_luminance:1 r(0.6800,0.3200) g(0.2650,0.6900) b(0.1500 0.0600) wp(0.3127, 0.3290) min_luminance=0.005000, max_luminance=1000.000000' [15:15:32 CET] <tango_> hello all. I recorded a video with obs, and post-facto I realized that the video is capturing an area (of the screen) larger than I initially intended, so I was looking for a way to crop the video with ffmpeg. Ideally, I would like to keep the encoding settings for the video output stream to be exactly the same of the original input stream. my command line at the moment is: ffmpeg -i input.flv -vf [15:15:37 CET] <tango_> "crop=1024:768:0:0" -c:a copy out.flv. I couldn't find a straightforward way to tell ffmpeg to use the same encoding _settings_ as the input, though [15:15:45 CET] <tango_> the input settings are Stream #0:0: Video: h264 (High), yuv420p(progressive), 1920x1080, 2500 kb/s, 30 fps, 30 tbr, 1k tbn, 60 tbc [15:20:40 CET] <furq> tango_: there isn't a straightforward way to do that [15:20:54 CET] <furq> you can crop h264 without reencoding but it won't actually make the file any smaller [15:21:06 CET] <furq> https://www.ffmpeg.org/ffmpeg-bitstream-filters.html#h264_005fmetadata [15:26:56 CET] <tango_> furq: for the time being I'd be satisfied even with just that [15:27:05 CET] <tango_> I'll just remember to use the correct settings the next time 8-) [15:34:00 CET] <tango_> furq: thanks, -bfs:v h264_metadata=crop_right=$((1920-1024)):crop_bottom=$((1080-768)) did the thing [15:45:32 CET] <scramblez> I've used libx265 to transcode it, should have I used hevc_cuvid? Or is there some option I could pass to it to copy over the HDR metadata without having to specify each one individually? [15:52:25 CET] <scramblez> Oops! Sorry, meant to say: hevc_vaapi [16:10:48 CET] <Hello71> why is cropping part of h264? shouldn't it be a container feature? [16:10:54 CET] <Hello71> do encoders actually use it? [16:12:05 CET] <Hello71> tango_: while you're at it, you probably want crf, not cbr [16:12:38 CET] <tango_> Hello71: what's crf? [16:12:45 CET] <tango_> Hello71: (fwiw, the original video was recorded by obs) [16:13:13 CET] <Hello71> I can't remember if obs is fine or sucks [16:13:16 CET] <Hello71> iirc it's "meh"? [16:13:35 CET] <tango_> Hello71: is there anything better for linux on X without a compositor running? [16:13:43 CET] <tango_> ffmpeg directly? 8-D [16:13:49 CET] <tango_> I mean, the video comes out fine [16:13:53 CET] <Hello71> x11grab is fine if you're not picky about latency [16:14:16 CET] <tango_> I potentially want to use the same for streaming in the future [16:27:14 CET] <scramblez> Hmm ... I just came across this ticket which is broadly similar to my problem (I'm not resizing the video): 'https://trac.ffmpeg.org/ticket/7037' [16:27:28 CET] <scramblez> I assume it has not been resolved yet. [16:28:32 CET] <scramblez> ffprobe shows color_range, color_space, color_transfer, and color_primaries are not transferred over to mp4. :( [16:30:02 CET] <scramblez> Is there a way to specify these individually for the libx265 encoder? [16:30:42 CET] <scramblez> Meant to say, not transferred over to m2ts [16:33:55 CET] <ritsuka> scramblez: probably you can pass it to x265 with -x265-params [16:36:02 CET] <furq> scramblez: -colorspace -color_primaries -color_trc -color_range [16:38:13 CET] <scramblez> I also found this: '-metadata string=string' under the x265 parameters. I wonder if I could pass the whole 'Mastering Display Metadata' string to it? [16:42:23 CET] <scramblez> furq: thanks! I entered these values individually and it is running now. No errors yet! :) [16:42:49 CET] <scramblez> No luminance values though. [16:46:29 CET] <furq> i have no idea if that stuff works for tonemapping [16:48:53 CET] <GuiToris> hey, I can't find it anywhere, can I adjust the quality of jpegs with ffmpeg? [16:49:52 CET] <furq> -q [16:50:17 CET] <Hello71> although the ffmpeg jpeg encoder is usually worse than alternatives [16:50:54 CET] <GuiToris> Hello71, do alternatives mean image magick here? [16:50:59 CET] <Hello71> pretty sure it's designed for mjpeg. same format, different requirements [16:51:13 CET] <Hello71> imagemagick is fine. if you want high quality I hear the google one is pretty good [16:51:29 CET] <furq> why would it being designed for mjpeg affect the quality [16:52:18 CET] <GuiToris> Hello71, I'm not sure if Image Magick can alter the pixel format [16:52:55 CET] <Hello71> furq: moving pictures are different from non-moving pictures? [16:53:02 CET] <furq> mjpeg is intra-only [16:53:20 CET] <Hello71> right [16:53:40 CET] <Hello71> but people usually don't view moving pictures one frame at a time [16:53:44 CET] <Hello71> they view them as a... movie [16:53:56 CET] <Hello71> is the point that I'm making [16:54:07 CET] <furq> if the point is "they put less effort in because who cares" then sure [16:54:55 CET] <furq> anyway -q 2 is the best you'll get out of ffmpeg's encoder [16:55:39 CET] <furq> mozjpeg is the best in class afaik and it does support changing the subsampling [16:55:45 CET] <furq> -sample 2x2 for 4:2:0 [16:56:33 CET] <GuiToris> furq, 2? is it differs from the one that gimp uses? I've tried -q 91 and -q 50 and well both images are ugly [16:56:49 CET] <furq> it goes from 2 to 31 [16:56:55 CET] <furq> anything above 31 will be trash [16:57:04 CET] <GuiToris> so it's a different scale :/ [16:57:11 CET] <furq> the range of values for -q in ffmpeg is an eternal mystery [16:57:17 CET] <furq> it doesn't help that it's not even documented for most codecs [16:58:36 CET] <GuiToris> oh and this is mjpeg, I haven't noticed it [17:03:10 CET] <scramblez> furq: Thanks again. Your suggestions worked (at least for sorting out the colorspace). :) [17:03:28 CET] <GuiToris> furq, can I specify conventional jpeg with -f? [17:03:40 CET] <GuiToris> furq, -f jpg doesnt work [17:03:59 CET] <GuiToris> I'm using still images [17:13:09 CET] <dryft> hello all, I'm trying to stream with ffmpeg and failing, I'd appreciate anyone taking a look. https://0x0.st/sUvl.txt [17:14:22 CET] <dryft> presumably I'm missing some arguments or they're in the wrong order idk [17:17:23 CET] <furq> dryft: -f flv [17:17:55 CET] <furq> GuiToris: -f singlejpeg [17:18:52 CET] <GuiToris> great, it works :) [17:41:55 CET] <dryft> furq, thank you. [17:42:15 CET] <dryft> if only I had asked here before searching for hours [17:42:21 CET] <dryft> :D [17:45:33 CET] <Hello71> mozjpeg is the best if you want it to encode pretty fast [17:45:55 CET] <Hello71> if you are happy to spend several minutes on each file then guetzli produces much better results [18:20:17 CET] <georgeli_> Hello, I have a problem about implementing hardware encoding on raspberry pi by using openmax. I always get wrong time stamp in my AVpacket. I notice since the packet got initialized, the time stamp was assigned to be 634 (seems to be the numerator of time base) and it does not change overtime. [18:21:10 CET] <georgeli_> AVPacket pkt; printf("0-time stamp = %ld, enc = %d/%d st = %d/%d\n", pkt.pts, encoder_ctx->time_base.num,encoder_ctx->time_base.den, fmt_encoder_ctx->streams[video_stream]->time_base.num, fmt_encoder_ctx->streams[video_stream]->time_base.den); printf("avframe time stamp = %ld\n", sw_frame->pts); av_init_packet(&pkt); printf("1-time stamp = %ld, enc = %d/%d st = %d/%d\n", pkt.pts, [18:23:43 CET] <georgeli_> Sorry, I don't know how to paste code here. I always get time stamp equals 634 for my printf, even right after I initialize AVpacket. That's weird. [18:58:07 CET] <leite> hi [00:00:00 CET] --- Tue Nov 20 2018

1 0

ffmpeg-devel.log.20181118
by burek 19 Nov '18

19 Nov '18

[00:19:03 CET] <philipl> BtbN: FYI, You didn't tag the 8.2.16 release. [01:16:27 CET] <BtbN> philipl, I didn't make a 8.2.16 release, as there are outstanding patches [01:18:17 CET] <philipl> OK [01:20:29 CET] <philipl> BtbN: also: https://devtalk.nvidia.com/default/topic/1038940/video-codec-sdk/announceme… [01:20:49 CET] <philipl> I'm not entirely sure what the punchline is here - I think it means that drivers supported by SDK 8.2 were still reporting API version 8.1 [01:22:58 CET] <BtbN> That's where I got the change from. The new nvidia SDK defines it as 8.2. [01:23:07 CET] <BtbN> Weird that no driver actually reports that version. [01:25:04 CET] <philipl> latest drivers do. [01:25:10 CET] <philipl> I can use nvenc with latest headers and 415 driver [01:25:28 CET] <BtbN> Well, they failed to update the version requirement then [01:26:20 CET] <BtbN> I'll probably just revert the change, as there isn't really any gain [01:27:26 CET] <philipl> my bad. [01:27:30 CET] <philipl> 415 also says 8.1 [01:27:36 CET] <philipl> So yeah, it just doesn't match any released driver. [01:27:57 CET] <philipl> Seems like they goofed. [01:28:40 CET] <BtbN> SDK team being like "Oh we forgot to bump this", and nobody told the driver team. [01:29:37 CET] <philipl> Seems that way [01:57:53 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.3:ecbf42e13a68: avcodec/mpegaudio_parser: Consume more than 0 bytes in case of the unsupported mp3adu case [01:57:53 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.3:14c8795361a8: avcodec/cavsdec: Propagate error codes inside decode_mb_i() [01:57:53 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.3:266102567981: avcodec/shorten: Fix integer overflow with offset [01:57:53 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.3:3c4a874388a9: fftools/ffmpeg: Repair reinit_filter feature [01:57:53 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.3:14778d3fda0a: avcodec/pngdec: Check compression method [01:57:54 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.3:9cba90e5ef03: Changelog: update [02:10:58 CET] <cone-200> ffmpeg 03Martin Vignali 07n3.3.9:HEAD: avfilter: use av_clip_uintp2 instead of av_clip for 10b and 12b [03:02:29 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:b130cdb92332: avutil/pixfmt: Document chroma plane size for odd resolutions [03:02:30 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:430405eb0bd0: swresample/swresample: Fix input channel count in resample_first computation [03:02:31 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:8ed1874dab71: avcodec/diracdec: Prevent integer overflow in intermediate in global_mv() [03:02:32 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:33ee198cdbb2: avcodec/dirac_dwt_template: Fix several integer overflows in horizontal_compose_daub97i() [03:02:33 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:6ebb8f5ab566: avcodec/diracdec: Change frame_number to 64bit as its a 32bit from the bitstream and we also have a -1 special case [03:02:34 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:ec7cf0c74f75: avcodec/diracdec: Check slice numbers for overflows in relation to picture dimensions [03:02:35 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:7a7a6bf8e250: avcodec/diracdec: Check bytes count in else branch in decode_lowdelay() too [03:02:36 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:09800cbc0105: avcodec/qtrle: Check remaining bytestream in qtrle_decode_XYbpp() [03:02:37 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:1b283238226b: avformat/flvenc: Check audio packet size [03:02:38 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:cc9263c2e165: avcodec/aacpsdsp_template: Fix integer overflow in ps_stereo_interpolate_c() [03:02:39 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:1a9810dde019: avcodec/mpegaudio_parser: Initialize poutbuf* [03:02:40 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:f20a35a13e4b: avcodec/shorten: Check verbatim length [03:02:41 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:eba99acdf267: avcodec/shorten: Fix integer overflow in residual/LPC combination [03:02:42 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:c71fe632b74c: avcodec/shorten: Fix signed 32bit overflow in shift in shorten_decode_frame() [03:02:43 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:b535e8936b72: avformat/rmdec: Fix EOF check in the stream loop in ivr_read_header() [03:02:44 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:949995b22521: avformat/mlvdec: read_string() received unsigned size, make the argument unsigned [03:02:45 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:5d7bab77de26: avformat/nsvdec: Do not parse multiple NSVf [03:02:46 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:c6b7060137cf: avcodec/snowdec: Fix integer overflow with motion vector residual [03:02:47 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:2df51ffcb7f2: avcodec/vb: Check for end of bytestream before reading blocktype [03:02:48 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:368e7a099422: avcodec/hq_hqa: Check remaining input bits in hqa_decode_mb() [03:02:49 CET] <cone-200> ffmpeg 03Dale Curtis 07release/3.2:a72d42b80ff4: avformat/mov: Error on too large stsd entry counts. [03:02:50 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:a90f6d3e908b: avcodec/ra144: Fix undefined integer overflow in add_wav() [03:02:51 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:59c463f4ccb0: avcodec/h264_refs: Document last if() in ff_h264_execute_ref_pic_marking() [03:02:52 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:9f0e0a4cf29c: avcodec/dvdsubdec: Avoid branch in decode_run_8bit() [03:02:53 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:3b20cd4f6576: avcodec/shorten: Fix bitstream end check in read_header() [03:02:54 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:5fe92631fc20: avcodec/zmbv: Update decomp_len in raw frames [03:02:55 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:2baee6dd1b9e: avcodec/zmbv: Check that the decompressed data size is correct [03:02:56 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:64bccfd11dde: avcodec/mpeg4videodec: Fix undefined shift in get_amv() [03:02:57 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:7bcc4d7ea7b2: avcodec/dvdsubdec: Sanity check len in decode_rle() [03:02:58 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:72babd45c81e: avcodec/unary: Improve get_unary() docs [03:02:59 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:9181ecd2e602: avformat/utils: Fix integer overflow in discontinuity check [03:03:00 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:77d35ab73435: avformat/utils: Never store negative values in last_IP_duration [03:03:01 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:f266eca585f0: avcodec/ra144: Fix integer overflow in add_wav() [03:03:02 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:72268fbdc37e: avcodec/h264_cavlc: Check mb_skip_run [03:03:03 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:1effc4baa19d: avcodec/mpeg4videodec: Fix typo in sprite delta check [03:03:04 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:353ef58a8279: avcodec/jpeg2000dec: Fix off by 1 error in JPEG2000_PGOD_CPRL handling [03:03:05 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:e6f8f6a7eefd: avcodec/msrle: Check that the input is large enough to contain a end of picture code [03:03:06 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:c2af767c3e28: avutil/integer: Fix integer overflow in av_mul_i() [03:03:07 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:875a5d055b4c: avcodec/mpegaudio_parser: Consume more than 0 bytes in case of the unsupported mp3adu case [03:03:08 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:d10266a96f28: avcodec/cavsdec: Propagate error codes inside decode_mb_i() [03:03:09 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:0612e6a0d7de: avcodec/shorten: Fix integer overflow with offset [03:03:10 CET] <cone-200> ffmpeg 03Michael Niedermayer 07release/3.2:de1d2ea6b20a: avcodec/pngdec: Check compression method [03:11:43 CET] <philipl> BtbN: after the revert, we still need a new version number. 8.2.15.6? [03:52:36 CET] <cone-200> ffmpeg 03Carl Eugen Hoyos 07master:529debc98776: lavu/avsscanf: Do not mix declaration and code. [14:39:20 CET] <BtbN> philipl, I don't think there's need for a new release yet. I kinda want to wait a while, so we're not spamming those releases. [14:44:04 CET] <BtbN> philipl, the branch you linked does not compile for me btw: https://bpaste.net/show/ccc581f32ef6 [15:21:57 CET] Action: Compn looks around for lou and sees nothing but alaskan tumbleweeds [16:40:34 CET] <philipl> BtbN: Ok. I just would like a bumped version number so I can set the dependency correctly in mpv to use the semaphore api [16:41:18 CET] <JEEB> I think that one got done in the pc file? [16:41:41 CET] <JEEB> http://git.videolan.org/?p=ffmpeg/nv-codec-headers.git;a=summary [16:41:50 CET] <JEEB> oh that got reverted [16:43:28 CET] <philipl> yeah. because nvidia goofed their version numbers [16:43:43 CET] <philipl> The drivers don't claim to support 8.2 yet. whoops. [16:44:47 CET] <philipl> BtbN: I've updated the branch again to fix the error. Please try again. [18:39:43 CET] <cone-690> ffmpeg 03Mark Thompson 07master:7a2065647421: cbs_h265: Add a lot more SEI parsing support [18:48:22 CET] <cone-690> ffmpeg 03Mark Thompson 07master:7ba63695b78c: mjpegdec: Fill raw huffman tables with default values too [19:13:55 CET] <cone-690> ffmpeg 03Mark Thompson 07master:f951036d3243: doc/encoders: Move libxavs2 to correct place in alphabetical order [19:13:56 CET] <cone-690> ffmpeg 03Mark Thompson 07master:b2ea757f6843: doc/encoders: Add libaom-av1 [19:30:24 CET] <cone-690> ffmpeg 03Mark Thompson 07master:b67435702090: vaapi_encode: Improve log message for unsupported profiles [19:56:09 CET] <cone-690> ffmpeg 03hwren 07master:b75afd25ba9d: lavc/libdavs2: remove unused frame counter [19:56:10 CET] <cone-690> ffmpeg 03hwren 07master:31b79e0210c4: lavc/libdavs2: correct frame type setting [20:16:57 CET] <cone-690> ffmpeg 03James Almer 07master:96717a42715f: avcodec/cbs_av1: don't abort when splitting Temporal Units containing OBUs with no obu_size field [21:24:53 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:a5b28b271af3: avfilter/af_acrossover: use av_strtod() to parse single number [21:24:54 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:0c7fb6e4a035: avfilter/af_adelay: use av_sscanf() [21:24:55 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:80265dba3de1: avfilter/af_afftdn: use av_sscanf() [21:24:56 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:9b2c325060c7: avfilter/af_sofalizer: use av_sscanf() [21:24:57 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:2de165a92bac: avfilter/vf_mix: use av_sscanf() [21:24:58 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:f79750bc320a: avfilter/af_ladspa: use av_sscanf() [21:24:59 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:c834f52c0018: avfilter/af_aecho: use av_sscanf() [21:25:00 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:72482363d1b7: avfilter/f_drawgraph: use av_sscanf() [22:13:44 CET] <BtbN> philipl, it does build now. Any specific command to test it? [22:15:07 CET] <BtbN> "./mpv.exe --gpu-api=vulkan --vo=gpu --hwdec=nvdec some.mkv" results in "[vo/gpu/cuda-nvdec] cu->cuImportExternalSemaphore(&evk->ss, &s_desc) failed -> CUDA_ERROR_NOT_SUPPORTED: operation not supported" "[vo/gpu/cuda-nvdec] cuda_ext_vk_init failed" [22:16:01 CET] <BtbN> And at the same time crashes with a MinGW Runtime Assertion "video/out/vulkan/ra_vk.c Line 2071, Expression: sem_out" [22:18:48 CET] <cone-690> ffmpeg 03Andreas Rheinhardt 07master:40b74abfca39: h264_redundant_pps: Fix memleak in case of errors [22:21:17 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:f3d8a4ea4623: avfilter/vf_limiter: always limit min/max to format depth [22:21:18 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:ead0d2eb7686: avfilter/vf_limiter: remove unused header [22:24:44 CET] <philipl> BtbN: cuda error not supported... [22:24:52 CET] <philipl> Are you running latest windows drivers? [22:25:09 CET] <BtbN> 416.94 [22:25:16 CET] <philipl> Huh. That's just weird. [22:25:39 CET] <philipl> The minimum windows cuda 10 driver version is 411.xx [22:26:08 CET] <philipl> Maybe I messed up the actual code around the HANDLE [22:26:31 CET] <philipl> Maybe you're willing to try and fix it :-P [22:34:42 CET] <philipl> BtbN: that devtalk discussion has gone well past what I meaningfully comment on around nvenc. [22:35:18 CET] <BtbN> Probably won't have time until next weekend. [22:36:02 CET] <philipl> K [22:38:58 CET] <cone-690> ffmpeg 03Mark Thompson 07master:a830056b32fe: vaapi_encode_h264: Reduce SAR to valid range [22:38:59 CET] <cone-690> ffmpeg 03Mark Thompson 07master:80ce23da6533: vaapi_encode_h265: Reduce SAR to valid range [23:02:50 CET] <cone-690> ffmpeg 03Paul B Mahol 07master:a064530da8b6: avcodec/scpr: add version 3 support [23:08:27 CET] <cone-690> ffmpeg 03Mark Thompson 07master:e3991a5a459a: libaomenc: Drop unused noise-sensitivity option [23:13:39 CET] <cone-690> ffmpeg 03Andreas Rheinhardt 07master:8f875a90c440: avutil/mem: Correct documentation of av_fast_*alloc(z) [00:00:00 CET] --- Mon Nov 19 2018

1 0

ffmpeg.log.20181118
by burek 19 Nov '18

19 Nov '18

[00:25:32 CET] <Rick627> turn off notifications [01:59:03 CET] <analogical> why doesn't FFmpeg accept wildcards in filenames? [01:59:55 CET] <pink_mist> that's not up to ffmpeg, that's up to your shell [02:00:26 CET] <pink_mist> shells are the things that expands wildcards into a bunch of filenames. never utilities themselves [02:01:39 CET] <pink_mist> however, I'm pretty sure ffmpeg does support some kinds of wildcards, such as numbers going up sequentially [02:06:10 CET] <pink_mist> oh, actually, even actual wildcard globbing if you've compiled ffmpeg with that option and specify it on the commandline -- http://ffmpeg.org/ffmpeg-all.html#Examples-7 [05:07:09 CET] <GlenK> hiya. I was kinda hoping to get ffmpeg to not use all my cpus. neither -threads 1 and -filter_threads 1 seems to do the trick. any help? [05:40:03 CET] <GlenK> eh. oh well. cpulimit is my friend at least. =) [06:39:47 CET] <KombuchaKip> I am writing a unit test for my custom AVIOContext. In order to test that it is working, I'd like to pull all data that its surrounding AVFormatContext can without doing any decoding. I just want to read all the data and verify the checksum to ensure it's what was expected of the source input data. [11:33:53 CET] <amosbird> Hello, how can I debug avformat_open_input returning Invalid data found when processing input ? [13:45:57 CET] <ariyasu> [dshow @ 000001e28f38afc0] real-time buffer [AVerMedia BDA Analog Capture Secondary] [video input] too full or near too full (81% of size: 3041280 [rtbufsize parameter])! frame dropped! [13:46:07 CET] <ariyasu> is there someway to increase the buffer size? [13:51:21 CET] <furq> -rtbufsize [13:52:25 CET] <ariyasu> ty [19:09:28 CET] <KombuchaKip> I am writing a unit test for my custom AVIOContext. In order to test that it is working, I'd like to pull all data that its surrounding AVFormatContext can without doing any decoding. I just want to read all the data and verify the checksum to ensure it's what was expected of the source input data. [19:09:53 CET] <JEEB> probably opening some raw demuxer is the best way to go about that [19:10:35 CET] <JEEB> most demuxers parse the input and effectively only give you the packets within but not the actual full data :P [19:48:20 CET] <KombuchaKip> JEEB: What about avio_read(3)? [19:48:49 CET] <JEEB> if an API client can actually utilize those, I guess... just not sure if those are public [19:49:45 CET] <JEEB> https://www.ffmpeg.org/doxygen/trunk/avio_8h.html#abb4e58439be0bff0dc2e2974… [19:49:59 CET] <JEEB> well, if that's usable by public API clients then sure [19:50:19 CET] <JEEB> and yes, we've seemingly got an example using that [19:50:24 CET] <JEEB> so I think you're good with that [19:50:32 CET] <JEEB> I'm way too used to just passing the thing through to AVFormat XD [19:51:56 CET] <KombuchaKip> JEEB: Yes, of course. Thank you. [19:52:18 CET] <KombuchaKip> JEEB: It's just for a unit test to verify all my pointer arithmetic and expected behaviour is correct in my custom AVIOContext [19:55:08 CET] <JEEB> KombuchaKip: yea, makes sense to keep it limited to the AVIO context then [19:57:03 CET] <KombuchaKip> JEEB: =) [20:43:45 CET] <wwp> hi [20:45:06 CET] <wwp> any idea why my ffmpeg 4.1 compiled from the sources, static, (following https://trac.ffmpeg.org/wiki/CompilationGuide/Centos) on my centos7 is 2.5 slower than the centos7's one (2.8.15)? [20:45:48 CET] <wwp> I experience the same perfs differences on a centos6 (4.1 vs centos' 0.6.5) [20:46:21 CET] <JEEB> are you sure you are actually enabling SIMD? [20:46:26 CET] <JEEB> in both dependencies and FFmpeg itself [20:46:37 CET] <JEEB> also which modules are you finding slowness in? [20:46:38 CET] <wwp> hm. I'm not sure at all [20:46:55 CET] <wwp> I did a basic flac->mp3 convertion in a loop to measure timings [20:46:56 CET] <JEEB> FFmpeg and x264 at least would require disable-asm to be defined to let you build for x86 [20:48:09 CET] <wwp> the instructions at the link I pasted above doesn't mention any --disable-<> optimiization-related switch [20:48:48 CET] <wwp> s/doesn/don/ [20:51:03 CET] <wwp> config.h shows HAVE_SIMD_* -> 1 [20:52:48 CET] <furq> there's no builtin mp3 encoder so maybe the lame config is different [20:53:27 CET] <wwp> lame-3.100 in that case [20:54:26 CET] <wwp> lame is configured with --enable-nasm [20:54:56 CET] <furq> fun [20:55:03 CET] <furq> you should probably use an internal encoder anyway [20:55:12 CET] <furq> aac or ffv1 are probably good choices [20:56:24 CET] <wwp> my point was to compare the perfs of the ffmpeg 4.x compiled from the sources and the distro's default one [20:56:54 CET] <wwp> do you mean that the same ffmpeg command may use different encoders following how ffmpeg is compiled? [21:45:26 CET] <markweston> can you convert a video to use only 16 colors/ [21:49:16 CET] <markweston> I mean I know you can, I'm interested whether it can be easily done with ffmpeg? [21:56:42 CET] <durandal_1707> yes, pelettegen & paletteuse [22:01:52 CET] <markweston> thanks [22:34:38 CET] <taliho> Hello, I've been doing some reading on the HEVC bit stream syntax. I'm wondering if I could ask a clarification question on this forum? [22:37:36 CET] <taliho> My question is whether the NAL units generated by x265 are directly compatible with the definition of PES stream? [22:40:52 CET] <taliho> Also, if I would like mux a video stream with general binary data, do the packets of the binary data need to be in the PES syntax? (i.e https://en.wikipedia.org/wiki/Packetized_elementary_stream) [22:41:55 CET] <taliho> if the muxer is mpegts [22:45:16 CET] <JEEB> the H.222 specification has notes about AVC/HEVC (NAL unit based video formats) [22:45:39 CET] <JEEB> https://www.itu.int/rec/T-REC-H.222.0-201703-S/en [22:45:45 CET] <JEEB> see this document, the PDF is freely available [22:46:00 CET] <JEEB> and it's a new enough version to have AVC and HEVC there [22:48:19 CET] <hola1> I've got some footage from a goXtreme action cam which doesn't play fluently. How do I change the codec? [22:49:30 CET] <taliho> thanks JEEB, I'll read this document [22:49:59 CET] <JEEB> but I think what you put into MPEG-TS PES streams were just Annex B NALUs [22:50:12 CET] <JEEB> not that I'm sure but that's how it felt like :P [22:56:02 CET] <taliho> JEEB, that's what I thought too. The wiki article on PES got me confused though, because the ANNEX B NAL unit syntax is a bit different to PES syntax on wiki [22:56:55 CET] <JEEB> just read the H.222 document, that's the MPEG-TS/PS spec :P [22:57:11 CET] <JEEB> and as far as I remember it has parts on NAL unit based video (AVC/HEVC) [22:57:20 CET] <JEEB> at least I've seen the IDs there [23:14:18 CET] <kjy> Can anyone help me do rgb color operations using the "program_opencl" filter? I've got the filter itself working, but no matter what combination of format filters I use it always seems like I'm working in yuv. [23:14:27 CET] <kjy> Here is the basic command without any format filters: https://pastebin.com/61KYn4zX [23:18:00 CET] <jkqxz> kjy: It should do the right thing if you put a 'format=rgba' before the hwupload and after the hwdownload. [23:23:58 CET] <kjy> jkqxz: I've tried that, and although it does seem to change the pixel format. It doesn't seem to be working as rgba in the kernel. [23:24:29 CET] <kjy> This kernel, https://pastebin.com/ngKnwBkG, yields white. However, you would expect blue or red (depending on where alpha gets put) [23:27:44 CET] <jkqxz> Why do you have any inputs at all? [23:29:58 CET] <kjy> jkqxz: I'm just testing the waters. Normally I would be doing something with them. [23:34:08 CET] <jkqxz> Oh, duh, stupid CL trap. You need to write "(float4)(0.0f, 1.0f, 0.0f, 1.0f)". Just "(0.0f, 1.0f, 0.0f, 1.0f)" is a comma expression which evaluates to 1.0f which then gets assigned to all four components of the float4. [23:35:19 CET] <jkqxz> Works for me with that fixed. [23:38:13 CET] <kjy> jkqxz: Oh wow, thanks for that. I guess I should've spent more time reading CL tutorials. [23:38:55 CET] <kjy> jkqxz: It *seemed* straightforward so I got ahead of myself. [00:00:00 CET] --- Mon Nov 19 2018

1 0

ffmpeg-devel.log.20181117
by burek 18 Nov '18

18 Nov '18

[00:54:12 CET] <raidensnake> there's a bug trying to configure ffmpeg on the raspberry pi 3. I get the error smbclient not found. [00:55:54 CET] <BtbN> Install it, or don't enable it? [00:56:07 CET] <raidensnake> it is installed [00:56:31 CET] <raidensnake> it's saying it can't find libsmbcvlient.h when it exists/ [00:56:35 CET] <iive> you don't really need it [00:57:04 CET] <iive> it is just so you can access files from remote cift/windows network without mounting it. [00:57:19 CET] <iive> cifs [00:58:22 CET] <raidensnake> only thing is it's needed for omxplayer [01:09:32 CET] <iive> ?? why would you need samba for omxplayer?? [01:20:03 CET] <atomnuker> how is omxplayer 1.) not bitrotten to hell and 2.) not replaced by the crap but somewhat standard v4l2 hwaccel api already [02:25:07 CET] <J_Darnley> too bad he left so I can't insinuate he's too dumb to read config.log [11:14:33 CET] <cone-832> ffmpeg 03Paul B Mahol 07master:fbea2ea76ef9: avfilter/vf_neighbor: fix >8 bit deflate/inflate [14:10:00 CET] <durandal_1707> libavcodec/libaomenc.c:730:9: warning: unused variable 'pict_type' [-Wunused-variable] [17:16:54 CET] <durandal_1707> michaelni: see vf_blend code for grainextract, av_clip_uintp2 gives artifacts for bitdepth 10 & 12 [17:18:20 CET] <durandal_1707> either compiler miscompiles code or implementation of that function is faulty [17:19:16 CET] <cone-561> ffmpeg 03Philip Langdale 07master:420ab946ace2: avutil/cuda_check: Make sure this passes make fate-source [17:19:20 CET] <JEEB> would something like ubsan help with that? [17:19:40 CET] <durandal_1707> i have clang 6.0.0 so that is probably reason... [17:19:53 CET] <durandal_1707> lets see if i switch to gcc [17:22:42 CET] <durandal_1707> same output, so it is broken code [17:23:15 CET] <JEEB> no, ubsan is a module in some compilers which would find if something's UB (sometimes) [17:23:23 CET] <JEEB> and it should warn about it [17:28:01 CET] <durandal_1707> JEEB: right, it reports something [17:28:11 CET] <JEEB> cool [17:28:24 CET] <JEEB> hopefully it's useful :D [17:46:17 CET] <BBB> atomnuker: have you made any progress on the EC SIMD? [18:07:11 CET] <cone-561> ffmpeg 03Martin Vignali 07master:d3621b232142: avfilter: use av_clip_uintp2 instead of av_clip for 10b and 12b [18:53:51 CET] <durandal_1707> it is too cold to write SIMD [20:32:43 CET] <atomnuker> BBB: no, I was waiting for you to bug me again to get motivated to do it [20:46:02 CET] <BBB> atomnuker: hm.. j-b is a better bugger than me, shall I ask him to join? [20:48:42 CET] <durandal_1707> does motion compesation filter would just use motion vectors and copy block from previous or forward frame (whatever is uses forward or backward motion vectors)? [20:51:50 CET] <kierank> BBB: that sentence means something rude/funny in british english [20:52:18 CET] <BBB> <- foreigner [20:52:25 CET] <BBB> see, that's why you folks voted for brexit [20:52:30 CET] <BBB> you just don't understand me :-p [20:53:04 CET] <kierank> https://en.oxforddictionaries.com/definition/bugger [20:54:02 CET] <BBB> hehe :) [20:56:37 CET] <durandal_1707> what innocent could happen with three guys together... [20:57:42 CET] <JEEB> bink2 decoder? [21:00:31 CET] <durandal_1707> bink2 decoder is far from innocent [21:42:39 CET] <durandal_1707> looks like nobody here understand motion compensation concept... [21:43:53 CET] <atomnuker> durandal_1707: yes, motion compensation is essentially just that - copy blocks of pixels from previous frames [21:45:07 CET] <atomnuker> kierank: @the second definition of bugger - when did the oxford dictionary turn into the urbandictionary? [21:45:29 CET] <kierank> it's a widely used insult [21:45:35 CET] <kierank> not just a new one [21:45:43 CET] <nevcairiel> just have enough people use something for long enough, and it gets written into dictionaries [21:45:50 CET] <nevcairiel> language is fluid, afterall [21:45:52 CET] <atomnuker> I know, didn't know about that definition of it though [21:46:01 CET] <atomnuker> apparently its origin is australian [21:46:40 CET] <nevcairiel> not being british, i only knew the first definition (ie. 1.1) [22:13:15 CET] <rcombs> what's the difference between NVDEC and CUVID anyway [22:16:44 CET] <nevcairiel> just how its implemented [22:17:12 CET] <nevcairiel> nvdec is a proper hwaccel, so it gives you all proper metadata that the normal encoders would also give you, and cuvid is a decoder which probably lacks most of that handling =p [22:17:38 CET] <rcombs> ahh [22:26:39 CET] <philipl> BtbN: https://github.com/philipl/mpv/tree/cuda-semaphore-buffers [22:26:44 CET] <philipl> Could you test that on windows for me? [22:28:25 CET] <philipl> rcombs: cuvid also has worse compatibility because it relies on nvidia's decoder library which isn't as comprehensive as the ffmpeg ones; unsurprisingly. [22:30:38 CET] <rcombs> mmmmmm [22:30:46 CET] <rcombs> so the naming is just a bit misleading [22:31:14 CET] <JEEB> first one got named cuvid, basically :P [22:32:24 CET] <rcombs> I like the name "nvdec" a bit better if only for the symmetry with nvenc [22:34:03 CET] <philipl> Originally nvidia called it cuvid, so we called the first impl cuvid too. [22:34:10 CET] <philipl> Then they renamed it nvdec, so we used that for the hwaccel. [22:34:21 CET] <philipl> It conveniently reflects the chronology. [22:34:47 CET] <philipl> Old busted name for old busted decoder. :-) [00:00:00 CET] --- Sun Nov 18 2018

1 0

ffmpeg.log.20181117
by burek 18 Nov '18

18 Nov '18

[01:56:28 CET] <microcolonel> https://en.wikipedia.org/wiki/LDAC_(codec) This codec makes me angry [02:10:04 CET] <furq> high-resolution lossy audio is very important [03:26:49 CET] <microcolonel> furq: but choosing the "24-bit" depth and "96kHz" bandwidths is ridiculous [03:27:00 CET] <microcolonel> either they just say that, and it doesn't actually mean anything [03:27:13 CET] <microcolonel> or they've designed tremendous inefficiency into their codec [03:27:15 CET] <microcolonel> for no good reason [03:27:18 CET] <furq> you see, it's high resolution, and therefore [mumbles for 90 seconds] [03:27:41 CET] <microcolonel> they actually brag about the data rate [03:28:07 CET] <microcolonel> as though it's a good thing for it to be using twice as much duty cycle on the radio [03:28:33 CET] <microcolonel> scratch that, more like 2.5x competing standards [03:29:38 CET] <furq> don't sony make a gold plated walkman-branded pmp that plays dsd512 [03:29:46 CET] <furq> that presumably costs one million dollars [03:30:00 CET] <furq> this is very on brand for them [03:30:49 CET] <furq> https://www.amazon.co.uk/Sony-NW-WM1Z-High-Resolution-Walkman-Multi-Touch/d… [03:30:53 CET] <furq> only £2500. sorry sony [03:31:14 CET] <microcolonel> oof [03:31:31 CET] <microcolonel> yeah DSD is... shocking [03:31:47 CET] <microcolonel> like, that I can sorta allow [03:31:56 CET] <microcolonel> since it generally only shows up in audiophool products [03:32:07 CET] <microcolonel> they're subsidizing us with their idiocy [03:32:18 CET] <microcolonel> whereas this is being integrated into a lot of devices [03:32:36 CET] <furq> presumably everything still supports aptx or whatever [03:32:44 CET] <furq> and knowing sony they'll be asking some crazy licensing fee [03:32:56 CET] <microcolonel> nah [03:33:00 CET] <microcolonel> at least they made this one royalty free [03:33:05 CET] <microcolonel> because they make headphones which use it [03:33:14 CET] <microcolonel> the encoder is here https://android.googlesource.com/platform/external/libldac [03:33:14 CET] <furq> ah, of course [03:33:28 CET] <microcolonel> gives me an idea though I guess [03:33:34 CET] <microcolonel> Opus over A2DP [03:33:35 CET] <microcolonel> I mean why not [03:34:30 CET] <furq> i'll just stick with a cheaper phone and wired headphones [03:34:32 CET] <microcolonel> market it as variable-bandwidth 32-bit floating point [03:34:38 CET] <furq> lol [03:34:39 CET] <microcolonel> :- ) [03:34:49 CET] <furq> that's what i tell everyone about my 128k fraunhofer mp3s [03:35:13 CET] <microcolonel> True 32-bit audio :^ ) [03:36:08 CET] <microcolonel> I guess the one nice thing about ldac is complexity [03:36:14 CET] <microcolonel> I'm sure it costs practically nothing to encode [03:36:33 CET] <microcolonel> and the decoder is like... too small to manufacture as a separate chip [03:36:43 CET] <furq> at least it's not atrac [03:36:53 CET] <microcolonel> Unicode needs a cringe emoji [03:36:59 CET] <microcolonel> for this exact moment [03:37:13 CET] <furq> i microwaved a sonicstage cd once [03:37:15 CET] <furq> it felt pretty good [03:37:30 CET] <furq> i was half expecting it to tell me it wasn't compatible with the microwave [03:38:12 CET] <microcolonel> :- ) [03:38:25 CET] <microcolonel> I liked that joke so much that I even giggled a little. [03:39:16 CET] <microcolonel> furq: apparently LDAC actually has a 32-bit floating point mode [03:39:18 CET] <microcolonel> kill me onw. [03:39:19 CET] <microcolonel> *now [03:56:35 CET] <microcolonel> Anyway, going to see how hard it is to make a bluetooth device that implements A2DP [03:56:40 CET] <microcolonel> and test out embedding Opus [04:05:50 CET] <microcolonel> https://www.sony.com/electronics/play-music-library-near-hi-res-audio-sound… furq this is somehow worse [04:06:33 CET] <microcolonel> pages and pages of bullshit [04:08:24 CET] <furq> you know it's advanced technology when the charts have no labels on the axes [04:09:47 CET] <furq> i like the link to "Hi-Res Buzzwords" with no additional context [04:12:09 CET] <microcolonel> yeah, it's amazing too [04:12:18 CET] <microcolonel> I thought that was a broken alt text for an image [04:12:21 CET] <microcolonel> some sort of word cloud [04:12:37 CET] <microcolonel> the alt text being more honest than they otherwise are [04:13:36 CET] <furq> Lossless compression allows the original data to be reconstructed almost perfectly from the compressed data achieved by a class of algorithms. [04:13:39 CET] <furq> what [04:15:17 CET] <microcolonel> You know how people of medium to high intelligence know that when things are written in a complicated fashion, they're probably either lies or something similar? [04:15:28 CET] <microcolonel> I think that there's probably a special class of person [04:15:50 CET] <microcolonel> who is just dumb enough to not notice they're being conned [04:16:03 CET] <microcolonel> and just smart enough to be able to afford the equipment [07:30:47 CET] <jin> is there a fool proof way of going from transparent png to alpha webm? [07:30:55 CET] <jin> I been trying ffmpeg -framerate 30 -f image2 -i %03d.png -c:v libvpx -pix_fmt yuva420p -auto-alt-ref 0 out.webm [07:31:23 CET] <jin> even though I made the green transparent, it's still looks green in my webm [07:31:34 CET] <furq> what are you playing it back in [07:31:51 CET] <jin> mpv [07:32:06 CET] <jin> oh [07:32:07 CET] <furq> you need --vd=libvpx for alpha in webm [07:32:10 CET] <jin> my browser shows it transparent [07:32:29 CET] <furq> yeah the ffmpeg vp8 decoder doesn't support alpha [07:32:43 CET] <furq> --vd=libvpx forces the vpx decoder which does [07:37:36 CET] <jin> furq: im doing -c:v libvpx isnt that redundant? [07:37:47 CET] <furq> that's for encoding [07:38:04 CET] <furq> --vd=libvpx is an mpv option [07:38:12 CET] <jin> now i get it [07:38:35 CET] <jin> weird, still showed green but [07:38:38 CET] <jin> whatever, works in the browser! [07:38:44 CET] <jin> thats what i was going for anyways [07:39:07 CET] <jin> now I juts need an easier way instead of making every frame transparent from imagemagick [07:39:47 CET] <furq> ffmpeg has a chroma key filter [07:39:59 CET] <jin> yeah I tried ffmpeg -i in.mp4 -vf "colorkey=green:0.1:0.2" -c:v libvpx -b:v 1M -auto-alt-ref 0 out.webm [07:40:17 CET] <jin> didn't quite work [07:40:26 CET] <jin> or maybe it was the browser [07:40:49 CET] <furq> you might just need to tweak the values [07:41:03 CET] <furq> https://ffmpeg.org/ffmpeg-filters.html#colorkey [07:41:05 CET] <jin> is 0.1 and 0.2 a fuzz? [07:41:11 CET] <furq> yeah [07:41:19 CET] <jin> with imagemagick im using 15% [07:41:25 CET] <jin> so thats pretty much the same right [07:41:45 CET] <furq> maybe, i've not used im in a while [07:42:12 CET] <jin> got it!!!! [07:42:15 CET] <jin> ffmpeg -i green_screen_video.mp4 -vf "colorkey=#12fd00:0.1:0.2" -c:v libvpx -b:v 1M -auto-alt-ref 0 alpha_webm.webm [07:42:19 CET] <jin> works beautiful [07:42:30 CET] <furq> nice [14:14:14 CET] <livingbeef> Trying to use vaapi, I get this message: Failed to initialise VAAPI connection: -1 (unknown libva error). Failed to set value '/dev/dri/renderD128' for option 'vaapi_device': Input/output error. Is that some sort of permission error? [14:16:16 CET] <livingbeef> vainfo lists VAEntrypoingEncSlice for H264Main/High and others... [23:25:33 CET] <AntumDeluge> I'm trying to figure out if it is possible to capture audio from a single application on Windows with dshow (or any other method). Haven't been able to find an answer on the internet yet. The problem is I don't want system notification sounds or sound from any other running app included in the output. [23:25:37 CET] <AntumDeluge> I'm on Windows 10. [23:27:22 CET] <AntumDeluge> If it's not possible with FFmpeg, I can try to figure out how to mute everything else. [23:28:40 CET] <AntumDeluge> I tried muting "system sounds" in the Volume Mixer, but some app notifications still make noise. Perhaps just disabling all other background apps while recording is the answer. [00:00:00 CET] --- Sun Nov 18 2018

1 0

ffmpeg-devel.log.20181116
by burek 17 Nov '18

17 Nov '18

[01:26:54 CET] <cone-384> ffmpeg 03Chirag Lathia 07master:d6b1248fc6aa: avcodec/libvpxenc: Add a maximum constraint of 16 encoder threads. [13:17:37 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:35040048793b: fftools/ffmpeg: Repair reinit_filter feature [13:17:37 CET] <cone-187> ffmpeg 03Michael Niedermayer 07master:1f99674ddddc: avcodec/pngdec: Check compression method [17:39:49 CET] <durandal_1707> anybody against applying av_sscanf() ? [18:18:23 CET] <philipl> BtbN: that thread on the devtalk forum has generated more traffic. Someone's saying they see an nvenc performance regression in 4.1 [18:18:49 CET] <BtbN> Did they forget hwaccel_output_format? [18:19:05 CET] <philipl> Was that always necessary or a new requirement in 4.1? [18:19:22 CET] <philipl> eh. I don't know how he's decoding. [18:19:24 CET] <BtbN> Well, it is if you want to avoid needles copying around [18:19:27 CET] <philipl> I'll ask for full comamnd line [19:41:33 CET] <cone-593> ffmpeg 03Paul B Mahol 07master:4334e9221ee1: doc/filters: improve afftfilt documentation [19:52:17 CET] <philipl> BtbN: https://devtalk.nvidia.com/default/topic/1042459/video-codec-sdk/nvdec-in-f… [19:53:09 CET] <BtbN> Uhm, I don't see how ffmpeg could possibly cause that. [19:59:29 CET] <philipl> neither do I [20:08:46 CET] <nevcairiel> i thought t here a re instance limits anyway, even on quadro [20:12:28 CET] <philipl> Yeah, but he's claiming he's not hitting him as early with 4.0 [20:13:05 CET] <philipl> obviously, I'm not in a position to test without the quadro. [20:23:33 CET] <cone-593> ffmpeg 03Paul B Mahol 07master:7a73cca2546a: avfilter: add dedot video filter [20:32:16 CET] <cone-593> ffmpeg 03Paul B Mahol 07master:4b30726dd39f: lavu: add locale-independent sscanf implementation [20:32:17 CET] <cone-593> ffmpeg 03Paul B Mahol 07master:4902780d2b53: avfilter/vf_lut3d: use av_sscanf [20:36:35 CET] <JEEB> huh, I remember just some time ago both of the closed caption specs were nonfree [20:36:47 CET] <JEEB> now you can at least get the CTA versions for free? [20:36:53 CET] <JEEB> https://members.cta.tech/ctaPublicationDetails/?id=a10017f5-a01c-e811-90ce-… [20:39:48 CET] <durandal_1707> just use ccextractor? [20:43:45 CET] <JEEB> durandal_1707: for decoding we already have crap anyways [20:44:20 CET] <durandal_1707> incomplete [20:45:32 CET] <durandal_1707> anybody have pristine video samples with rainbows artifacts? [21:41:40 CET] <cone-593> ffmpeg 03Paul B Mahol 07master:395508d9b6c7: avfilter/vf_dedot: initizalize ret variable [21:46:30 CET] <cone-593> ffmpeg 03Carl Eugen Hoyos 07master:21dd05ee6ad3: lavfi/minterpolate: Remove declaration of two unused variables. [22:04:46 CET] <cone-593> ffmpeg 03Carl Eugen Hoyos 07master:fc94e9704e05: lavf/hlsenc: Do not mix declarations and code. [22:16:26 CET] <durandal_1707> the cnr(2) plugin produce ghost artifacts in chroma and does not get rid of rainbows under default settings [00:00:00 CET] --- Sat Nov 17 2018

1 0

ffmpeg.log.20181116
by burek 17 Nov '18

17 Nov '18

[00:11:22 CET] <pzy> haha oh duh, I'm dumb [00:31:49 CET] <colemickens> so I have some code here that uses ffmpeg to transcode frames from dmabuf to a video file. [00:32:12 CET] <colemickens> I'd instead like to take that and throw it into some sort of stream that I can consume with plain ole command line ffmpeg, so that I can crop, pick my own transcoding options, etc. [00:32:29 CET] <colemickens> But I'm unsure of if there's a way to do this, and/or what combination of transport and codec would be recommendable. [03:07:15 CET] <KombuchaKip> What is the best way to distinguish at compile time ffmpeg from libav headers on the user's system? [03:20:00 CET] <pink_mist> KombuchaKip: look at how mpv does it maybe? [03:21:11 CET] <pink_mist> KombuchaKip: https://github.com/mpv-player/mpv/blob/master/wscript#L437 [03:21:13 CET] <KombuchaKip> pink_mist: I didn't know it did, but do you have a specific location in the build environment or source level you'd suggest? [03:21:18 CET] <KombuchaKip> pink_mist: Thank you. [03:23:54 CET] <KombuchaKip> pink_mist: That's in Python which isn't really my first language. I see it's checking flags with pkg-config, but it's not really clear to me how it's distinguishing between the two. [03:25:45 CET] <pink_mist> uhm, well, it's using the 'waf' build system, so it's not simply python that needs to be learned :P but basically it's compiling using 'cc' or whatever the C piece of code on line 457 [03:25:51 CET] <pink_mist> and checking what the result is [03:26:05 CET] <pink_mist> wrapping it in a main and whatnot [03:26:55 CET] <pink_mist> that is, if it fails to compile it knows it's one or the other [03:27:05 CET] <pink_mist> and if it succeeds it knows it's the reverse :P [03:27:19 CET] <KombuchaKip> pink_mist: Looks like it's checking the libavcodec/version.h header to see if LIBAVCODEC_VERSION_MICRO >= 100, and if so, then this is libav fork and not ffmpeg. Is that correct? [03:27:28 CET] <pink_mist> I believe so [03:28:06 CET] <pink_mist> on line 467 it does the reverse check, just to be on the safe side [03:28:23 CET] <pink_mist> see how it switches which case returns 1 and which one returns -1? [03:29:19 CET] <KombuchaKip> pink_mist: It's a bit weird, but it looks like if micro version is >= 100 then it thinks its libav. [03:29:54 CET] <KombuchaKip> pink_mist: I'm surprised they still don't simply have an explicit macro in their headers to distinguish the two because what they're doing is pretty easy to break if ffmpeg bumps the micro version >= 100. [03:30:33 CET] <pink_mist> I think it's the other way around [03:31:28 CET] <pink_mist> I'd expect they've gotten some form of assurances that ffmpeg will always keep it at 100 or higher, and that libav will always keep it below [03:31:33 CET] <pink_mist> but I don't rightly know [03:32:24 CET] <KombuchaKip> pink_mist: Right [13:24:16 CET] <shroomM> hey guys, one question ... [13:24:28 CET] <shroomM> I'm trying to concat a couple of mp4 files [13:24:35 CET] <shroomM> using the concat demuxer as described here... https://trac.ffmpeg.org/wiki/Concatenate [13:25:16 CET] <shroomM> they are served by a web server and ffmpeg is reading them via http [13:25:42 CET] <shroomM> this works fine, but i would like to speed it up, since all I'm doing is stream copy [13:26:45 CET] <shroomM> the download of a single file is slow, if I download files in parallel, I get higher speeds [13:26:59 CET] <shroomM> is there a way to instruct ffmpeg to "pre-fetch" files or download them in parallel? [13:27:16 CET] <ariyasu> no [13:27:41 CET] <ariyasu> but you could download them with another process like curl in multiple sessions for parallel downloading [13:27:50 CET] <ariyasu> then do the concat job with ffmpeg [13:28:11 CET] <shroomM> yeah, I'm limited with the disk space, so I want to do all with ffmpeg [13:28:42 CET] <shroomM> mp4 files themselves are less than 100MB [13:29:00 CET] <bencoh> well it's either disk or ram [13:29:06 CET] <shroomM> ram is better :) [13:29:09 CET] <shroomM> so I was thinking of launching 4 processes, downloading them to memory and writing them to named pipes [13:29:13 CET] <bencoh> so you could always download to tmpfs [13:29:36 CET] <shroomM> and then specifying named pipes in the concat file [13:29:38 CET] <bencoh> just make sure to clean up temp files [13:30:01 CET] <shroomM> ok, good idea, not sure if tmpfs is available in this environment, but I will check [13:30:23 CET] <shroomM> what do you think about the named pipes, does that sound like something that should work [13:31:00 CET] <shroomM> the reason i'm asking is that i have an example set-up, but the resulting file that ffmpeg produces is invalid :S [13:31:09 CET] <shroomM> and I don't get why [13:33:02 CET] <shroomM> I have a python script with a list of urls, each url gets its own fifo named pipe. [13:33:50 CET] <shroomM> the script then starts by reading the first url in memory and starts pipe write operation, where it blocks until something starts reading it [13:34:37 CET] <shroomM> separately i manually run ffmpeg with concat file with all the named pipes listed [13:35:12 CET] <shroomM> and this actually finishes, but with a bunch of warnings from ffmpeg about "Non-monotonous DTS in output stream" [13:35:57 CET] <shroomM> the resulting file does not work as expected :/ [13:36:06 CET] <shroomM> I don't get these errors if I specify http urls in the concat file and the resulting file works then [16:54:28 CET] <atbd> hi everybody, is it possible to add to each AVPacket metadata the current epoch time during reencoding with ffmpeg cli? [16:55:59 CET] <JEEB> usually you do something like set -itsoffset to the moment you start ffmpeg.c [16:56:06 CET] <JEEB> API users have more alternatives [16:56:17 CET] <JEEB> although I have also thought about following upipe's design [16:56:24 CET] <JEEB> which has three timestamps [16:56:28 CET] <JEEB> - coded timestamp [16:56:31 CET] <JEEB> - interpreted timestamp [16:56:38 CET] <JEEB> - receipt timestamp [16:56:48 CET] <JEEB> first was what was in the source container [16:56:55 CET] <JEEB> second one applies stuff like wrap-arounds [16:57:13 CET] <JEEB> third one is just the timestamp when that AVPacket/Frame was filled [17:04:14 CET] <atbd> okay thanks [17:04:35 CET] <atbd> upipe's design as you describe it would be great for what i want [17:50:37 CET] <trashPanda_> Hello, I have question regarding the H264 decoder using the api. I am feeding the decoder packets with correct dts/pts but when I recieve the frame, the PTS has been changed to a large negative number. [17:52:01 CET] <trashPanda_> The received frames dts is still correct. This only happens with a certain video, does anyone know what could cause the decoder to give back a large negative PTS number? Im talk like -6142771919317201 large... [18:23:20 CET] <SpeakerToMeat> Hi all [18:23:22 CET] <SpeakerToMeat> Question, I'm trying to burn some subs in ass in, but ffmpeg seems ot be ignoring the color (and alpha) settings for the style :/ any idea why? everythiong else in the style (borders etc) works but not color changes. [21:54:21 CET] <xgx> hello, can someone help me understand how to make an audio only m3u8. I found the other way around but this one is a bit harder\ [21:55:33 CET] <pzy> m3u8 is just a playlist [21:56:34 CET] <pink_mist> xgx: you do that by specifying an audio file in the playlist [21:57:11 CET] <xgx> i know. but i would like to send a rtmp stream to a server and with ffmpeg create a playlist with audio only [21:57:49 CET] <pink_mist> then make an rtmp stream with only audio [21:58:58 CET] <xgx> sounds simple enough. thought maybe it still needed to have some kind of vid data [21:59:01 CET] <xgx> ill try that [21:59:20 CET] <pink_mist> I have no idea tbh [21:59:28 CET] <pink_mist> never worked with rtmp myself [22:02:55 CET] <xgx> So far I have only done the simple stuff with it [22:03:07 CET] <xgx> receive the stream and made it a multi bitrate [00:00:00 CET] --- Sat Nov 17 2018

1 0

ffmpeg-devel.log.20181115
by burek 16 Nov '18

16 Nov '18

[00:22:27 CET] <cone-545> ffmpeg 03Mark Thompson 07master:2ce3a48f30fe: configure: Add missing xlib dependency for VAAPI X11 code [00:25:51 CET] <cone-545> ffmpeg 03Mark Thompson 07release/4.1:4f1e07090a9f: configure: Add missing xlib dependency for VAAPI X11 code [00:51:55 CET] <cone-545> ffmpeg 03James Almer 07master:79831f4531d9: avcodec/cbs_av1: fix storage size for segmentation_params feature_value fields [00:51:56 CET] <cone-545> ffmpeg 03James Almer 07master:f0f2832a5ce9: avcodec/cbs_av1: fix parsing signed integer values [00:54:09 CET] <cone-545> ffmpeg 03James Almer 07release/4.1:49bc641e89f7: avcodec/cbs_av1: fix storage size for segmentation_params feature_value fields [00:54:10 CET] <cone-545> ffmpeg 03James Almer 07release/4.1:a9e9303f2659: avcodec/cbs_av1: fix parsing signed integer values [04:44:08 CET] <cone-068> ffmpeg 03Philip Langdale 07master:19d3d0c05709: avutil/hwcontext_cuda: Define and use common CHECK_CU() [04:44:08 CET] <cone-068> ffmpeg 03Philip Langdale 07master:fa74e4aef210: avfilter/yadif_common: Add field type tracking to help bwdif [04:44:08 CET] <cone-068> ffmpeg 03Philip Langdale 07master:1096614c4200: avfilter/vf_bwdif: Use common yadif frame management logic [11:51:34 CET] <cone-384> ffmpeg 03Paul B Mahol 07master:02809e7b061e: avfilter/vf_lut2: implement support for different input depths [15:05:58 CET] <cone-384> ffmpeg 03Paul B Mahol 07master:c98a32e4adec: avfilter/vf_blend: add 10bit support [15:05:59 CET] <cone-384> ffmpeg 03Paul B Mahol 07master:1342ec526e32: avfilter/vf_blend: add 12bit support [19:14:59 CET] <cone-384> ffmpeg 03Paul B Mahol 07master:662120f2b27d: avfilter/vf_convolution: use sqrtf as its faster [22:28:23 CET] <philipl> BtbN: https://devtalk.nvidia.com/default/topic/1042459/video-codec-sdk/nvdec-in-f… [22:30:02 CET] <BtbN> philipl, the issue is ffmpeg.c not supporting a decoder changing the framerate [22:30:53 CET] <philipl> I thought forcing the frame rate on the command line fixed it. [22:30:58 CET] <philipl> People in the thread tried that [22:31:07 CET] <philipl> It apparently used to work and now has problems [22:31:09 CET] <JEEB> time base, maybe [22:31:10 CET] <BtbN> It does in very simple cases [22:31:10 CET] <cone-384> ffmpeg 03Marton Balint 07master:aecd63b92681: avfilter/af_afade: fix duration maximum [22:31:27 CET] <BtbN> And is not a supported mode of operation, so it aparently broke at some point [22:34:31 CET] <BtbN> cuvid does all it can, it halves the timebase, and fixes the fps [22:34:42 CET] <BtbN> The rest of ffmpeg just isn't expecting a decoder to do that. [22:35:06 CET] <JEEB> yea, ffmpeg.c is very static in various ways [22:44:21 CET] <philipl> BtbN: Ok. So maybe I should just tell them it's a bonus if it works and not to rely on it. [22:45:11 CET] <cone-384> ffmpeg 03Mark Harris 07release/4.1:fed94c2f22fc: avfilter/vf_fade: fix start/duration max value [22:45:11 CET] <cone-384> ffmpeg 03Marton Balint 07release/4.1:bb01cd3cc01c: avfilter/af_afade: fix duration maximum [22:45:15 CET] <BtbN> It works fine. Just not with ffmpeg.c [22:45:48 CET] <BtbN> It's also not something that's easily fixed, but a major Design issue. [22:47:01 CET] <BtbN> Like, it initializes the whole filter chain with the framerate/timebase the demuxer reports, way before consulting with the decoder [22:47:01 CET] <philipl> I mean it's not to be relied on with ffmpeg.c [22:47:04 CET] <philipl> I know it works fine in general. [00:00:00 CET] --- Fri Nov 16 2018

1 0

ffmpeg.log.20181115
by burek 16 Nov '18

16 Nov '18

[00:00:04 CET] <JEEB> and a basic FFmpeg, I'm not sure it will depend on anything else than system libs? [00:00:41 CET] <AlexMax> Maybe. Right now I just don't have any clue how to tell ffmpeg libraries to find the hardware decoder on my raspberry pi [00:02:03 CET] <JEEB> can't say I found anything special in init code of the AVCodecContext in a player I poke around [00:02:49 CET] <JEEB> and we can take a look at the init function in the decoder [00:03:05 CET] <JEEB> AlexMax: also there's another function to go through decoders in a list [00:03:18 CET] <JEEB> might want to make sure your library that you're linking against has the MMAL stuff [00:03:36 CET] <AlexMax> I ran ldd on it, it look promising [00:03:54 CET] <AlexMax> i saw some linking against some mmal libraries [00:03:56 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/mmaldec.c;h=647a2… [00:04:00 CET] <JEEB> this is the init function [00:04:06 CET] <AlexMax> Ya know, I should probably do that [00:04:25 CET] <AlexMax> I just didn't know if _I_ had to link against something extra to get mmal support or something [00:04:48 CET] <JEEB> I'll check what the -codecs listing does in the mean time [00:07:22 CET] <JEEB> ok, so it loops through AVCodecIDs in the binary [00:08:43 CET] <JEEB> ok, lol. just uses a loop of av_codec_next() :P [00:09:06 CET] <JEEB> checks if codec id matches what it wanted to loop though [00:09:36 CET] <JEEB> and prints out those things [00:10:27 CET] <JEEB> and in that av_codec_next loop you definitely do get an AVCodec pointer out [00:10:35 CET] <JEEB> so if -codecs that print that bastard out [00:10:50 CET] <JEEB> then you should be able to figure that thing out :P [01:35:19 CET] <AlexMax> well this is distressing [01:35:33 CET] <AlexMax> AVCodec* current_codec = NULL; current_codec = av_codec_next(current_codec); [01:35:42 CET] <AlexMax> current_codec ends up null again [01:35:47 CET] <AlexMax> no codecs? [01:49:58 CET] <pink_mist> https://stackoverflow.com/a/50970973 [01:50:05 CET] <JEEB> AlexMax: if you are on older versions, have you tried registering them with av_register_all? [01:50:43 CET] <JEEB> newer versions don't have the need, but older versions still had some dynamicness in there (which was only usable through private APIs, which is why it was removed) [02:02:09 CET] <AlexMax> Aha, I think that worked [02:02:55 CET] <AlexMax> Kind of strange that there is no trace of av_register_all in the original repository [02:03:49 CET] <AlexMax> but this is ffmpeg 3.2 [02:10:00 CET] <AlexMax> that would explain why it didn't work for me on this linux box, but it did in my local MSYS2 setup [03:11:19 CET] <ealdegue_> Hi ! anyone could help me with CMAF encoding and FFMPEG ? thanks ! [13:33:12 CET] <emacstheviking> I just screen recorded both monitors by accident. Is there a way I can split the video to reclaim just the right hand side as a separate file? I am reading the options now but there are so many! [13:33:19 CET] <emacstheviking> worst case would be to convert all frames to jpeg, run them through imagemagick and stitch them back together. [13:35:02 CET] <Mavrik> emacstheviking: "crop" video filter is made for that [13:35:27 CET] <Mavrik> ffmpeg -i <yourfile> -codec:v libx264 -vf crop=<your crop parameters> -codec:a copy output.mp4 [13:35:34 CET] <Mavrik> look up the crop filter documentation for parameters :) [14:37:11 CET] <CoreX> !filter [14:38:32 CET] <CoreX> ohh he's gone if he comes back Mavrik you should give the person this https://ffmpeg.org/ffmpeg-filters.html#crop [14:39:50 CET] <Mavrik> I was counting on the fact that he can google it :P [14:47:47 CET] <durandal_1707> winny: 11 [14:47:58 CET] <durandal_1707> again, lol [14:48:07 CET] <winny> durandal_1707: nice game [15:24:35 CET] <ErwanD> Hey everyone [15:24:44 CET] <ErwanD> I would like to use ffmpeg with opencl. Working on windows 10 with mingw64. I downloaded ffmpeg 4.1 sources, convert cuda opencl.lib to opencl.a. Then ./configure --arch=x86_64 --enable-opencl --prefix=/mydir Everything looks good. Then mingw32-make.exe, I had some errors related to ar fixed by modifying library.mak Then install. But I donŽt see any opencl.h header in mydir/include/libavutil [15:24:49 CET] <ErwanD> Do you know if itŽs normal ? How to ensure my build is "opencl ready" ? [16:37:30 CET] <trashPanda_> Hello, does anyone have any experience using ffmpeg to restream a rtsp stream? [16:38:07 CET] <trashPanda_> I am attempting to use the cli ffmpeg to re-stream a rtsp from a camera of mine, but I get a lot of "Error parsing AU headers" messages and no output stream [16:38:29 CET] <trashPanda_> The rtsp stream does render correctly with ffplay however, is there something that could be messing with ffmpeg's ability to re-stream it? [18:01:18 CET] <kepstin> ErwanD: there should be a few opencl filters included in your ffmpeg build if the opencl stuff was configured properly. (check by running ffmpeg -filters) [18:19:49 CET] <Zexaron> Hey guys [18:20:08 CET] <Zexaron> it's pretty clear WH is encoding videos different right https://www.politico.com/story/2018/11/14/white-house-trump-has-broad-discr… [18:20:52 CET] <Zexaron> if they're using CSPAN version, what is it that's doing that motion blur in particular, some kind of bad encoder setting with framerate, etc? [18:21:20 CET] <Zexaron> the video is in the middle of the article [18:22:04 CET] <pzy> I thought it was pretty clear the video was edited [18:26:41 CET] <furq> didn't prison paul edit that and then show a clip of him editing the video in sony vegas to prove that it wasn't edited [18:26:48 CET] <furq> he's not a smart man [18:29:32 CET] <Zexaron> with all the codec artifacts it's hard to prove it [18:30:32 CET] <Zexaron> any video could be doubted, any jpeg image, the common idea that "the blurring around the UFO proves it's fake", there's a lot of fakery on both sides, the UFO fakers and the fake assholes claiming everything's fake [18:30:33 CET] <pzy> not really [18:30:40 CET] <pzy> just an easy excuse to muddy the waters [18:31:10 CET] <pzy> no one's mind has been changed and no discourse has been achieved, same ol' same ol' [18:31:27 CET] <furq> there's a good rule of thumb you can use for situations like this [18:31:37 CET] <furq> if two people each say the other one is lying, and one of them is donald trump, he's the one that is lying [18:31:49 CET] <pzy> BUT BOTH SIDESSSSSSSS [18:32:19 CET] <Zexaron> unfortunately theres absolutely no guidelines or laws on the governement or federal level on the technicalities sorrounding video evidence, what kind of quality and what kind of criteria it has to pass to be qualitifed for evidence in court (or I just don't know about it) [18:32:39 CET] <pzy> that's up to 12 people to decide [18:32:44 CET] <Zexaron> not both sides in that way, i was referring to various channels on youtube [18:34:21 CET] <Zexaron> you shouldn't be able to use a crappy JPEG image to prove something in court, like a screenshot from a video which was recorded with a cellphone of the monitor which was playing back the video of a secruity camera [18:34:44 CET] <Zexaron> only in russia tho [18:34:59 CET] <pzy> that's why judges exist to determine permissibility [18:35:10 CET] <furq> well presumably if this does go to court then they'll just request the actual footage from cspan [18:35:44 CET] <Zexaron> none of them have ANY idea about the JPEG artificats, different techniques, photoshop, encoder differences, codecs, bitrate differences, color stuff [18:36:29 CET] <pzy> that's why expert witnesses exist when necessary [18:36:48 CET] <Zexaron> many of the things youtube kids jump to conclusions are infact artifiacts and glitches, the most common is the blur-border thing, it's a false positive [18:37:35 CET] <Zexaron> yeah but it's not something usual, a lot of cort proceedings nothing happens automaticlaly, the defence has to ask for it [18:37:53 CET] <Zexaron> demand it, etc, the court system is crap in so many ways [18:38:12 CET] <ariyasu> the motion blue looks like iterpolation [18:38:14 CET] <pzy> how many trials have you attended? our justice system is pretty amazing really [18:38:23 CET] <Zexaron> the primary thing being, it takes money, which means, no money, no justice, that pretty much breaks it right there [18:38:28 CET] <kepstin> regarding the sort of thing where digital media interacts with law, this is interesting reading: http://ansuz.sooke.bc.ca/entry/23 [18:38:30 CET] <ariyasu> nothing seems doctored [18:39:01 CET] <trashPanda_> Its not doctored, just diff angle diff camera, decoded/encoded/compressed [18:39:51 CET] <pzy> just throw more technical terms at people until they forget what the actual issue is [18:42:43 CET] <kepstin> basically, the bits themselves can't tell you & don't record whether they've been edited or not, but the law cares, and so it's a tricky interaction :) [18:42:56 CET] <Zexaron> yeah I was not sure if it's the same camera, but the point is pzy, there are SO MANY factors that it would take a team of experts, even if there's one obvious sign, you sill have a full analysis, just like NTSB, even if the people saw a missile laucnhed at the airplane, the cause of crash is obvious, but a full NTSB invesitgation with all the details would still be conducted [18:43:32 CET] <Zexaron> and in that video analysis you would have to gather all the sources, raw copies, cameras themeselfs, all the technical details of the cameras, encoders, etc etc etc [18:44:14 CET] <Zexaron> only then you start working looking for signs, and you need 10 year experience knowing all the false positives, that's what it takes to be impartial and fair at finding doctoring [18:44:29 CET] <kepstin> in collecting all those details, you'll figure out the path that the video took to become the final product, which sometimes matters more than the actual contents of the video in question [18:45:11 CET] <Zexaron> because the doctoring you see on youtube is easy and everyone can figure out, if it was really doctored they would to a lot better job than making it so easy that any random reader could notice [18:47:38 CET] <Zexaron> And also, not underestimate the ignornace, especially young people, thousands of people still fall for easy youtube jokes, like the japan 2011 tsunami gas canister exploding being propagates as a "white ghost", and other stuff, it's very easy to fool people with genuine footage that's in low quality [18:49:12 CET] <Zexaron> then you have more professional* fraud like Section51, which is a 17 year old stupid kid, makes totally fake youtube videos and gets millions of views sometimes [18:49:35 CET] <pzy> is the video really the crux of this whole thing anyway? it has nothing to do with stated reasons and just serves to muddy the waters [18:51:34 CET] <furq> this isn't really a new thing [18:51:50 CET] <furq> remember all the psychics who convinced themselves that "auras" on photographs were ghosts and not just development errors [18:52:13 CET] <furq> or all those moon landing guys who convinced themselves an eyelash that fell into a photo during processing was proof that a rock said "C" on it and was therefore proof that it was from a props department [18:52:22 CET] <furq> those crazy props department and their clearly labelled props [18:52:23 CET] <furq> s [18:52:56 CET] <Zexaron> and then comes NASA with their mars rovers, maybe they got REAL RAW images but for public it's "RAW" JPEG only, if they wanted to prove they're on MARS etc etc, these JPEGs are so bad they can hardly be taken seriously. [18:53:18 CET] <Zexaron> They would go to great lengths to doctor that footage, because of what's at stake and the consequences, but because they use JPEG and shit quality there's alot of possible false positives so this is an expert job: [18:54:20 CET] <Zexaron> https://www.youtube.com/watch?v=Zj6uH-SqOZ8 [18:58:45 CET] <Zexaron> yeah, all these S marks on the Nasa Curiousy JPEGs, could it be another JPEG codec false-positive [18:59:31 CET] <pzy> I feel like we did this 20 years ago already [19:57:38 CET] <KombuchaKip> Zexaron: You might find this work I did on NASA Mars photo recovery technology. https://goo.gl/xB4CUi You can find the source here: https://github.com/cartesiantheatre/vlr [19:57:46 CET] <KombuchaKip> Hey everyone. I'd really appreciate a code review of my custom AVIOContext for reading media from memory: https://pastebin.com/NGALFw8Y [20:00:13 CET] <Zexaron> interesting [20:14:30 CET] <KombuchaKip> Zexaron: Thanks. Unfortunately I don't have any DVDs left anymore for sale. [20:21:40 CET] <pzy> really cool KombuchaKip [20:22:16 CET] <pzy> when's the blu-ray version coming out? ;) [20:22:49 CET] <KombuchaKip> pzy: Thank you guys. Hah, if you'd like to put one together I'm happy to put it on the parent project's website: https://www.avaneya.com [20:23:11 CET] <KombuchaKip> pzy: It's a GPL game long away from completion I've been working on for years. [20:27:51 CET] <pzy> is the processed but uncompressed video available somewhere, or would you have to run the utility yourself? [20:36:27 CET] <trashPanda_> Does anyone have experience with RTSP streams? I'm having an issue with restreaming a camera feed via ffmpeg. I can display the feed just fine with ffplay. [20:37:14 CET] <trashPanda_> When using ffmpeg, I get numerous "Error parsing AU Headers" messages until a timeout is reached [22:45:56 CET] <BlackBishop> hi again awesome peeps. [22:46:27 CET] <BlackBishop> any chance I can easily add a dts track to an mkv from an existing truehd/ac3 track ? [22:47:07 CET] <JEEB> you can but not sure if you want. the only real reason to add a DTS track is to have bit streaming, but you already have two formats that are bit streamable [22:47:51 CET] <BlackBishop> JEEB: the problem is that for some reason kodi on my tv has an issue with those but works just fine with dts ! [22:48:09 CET] <BlackBishop> no idea why ! [22:48:32 CET] <BlackBishop> and I'm trying to confirm the audio stream is the issue [22:48:53 CET] <BlackBishop> I got 2 mkv files, identic video codec stream .. the only difference are the audio streams [22:49:21 CET] <BlackBishop> 1 works ok (stutters from time to time a bit but it's ok) .. the other one starts to stutter and looses audio after a little while [22:49:33 CET] <BlackBishop> it works ok when I pause/play or change audio tracks [22:49:41 CET] <BlackBishop> anyhow, trying to confirm the issue [22:50:25 CET] <JEEB> anyways, there's a lossy DTS encoder called "dca" [22:51:21 CET] <JEEB> and it's marked experimental so you need -strict experimental after the input to use it [22:56:21 CET] <BlackBishop> mkay.. [23:14:42 CET] <ILove47> Hi! I have no experience in video creation/editing at all and I'm not sure, it's the right channel to ask this, but I'll try :). If I should ask it in other channel - tell me where I can do that, please. [23:14:42 CET] <ILove47> I want to programatically create a video with a scoreboard with row transitions and other pretty effects (like this: https://giphy.com/gifs/1Y6RMg4ptzpOAdlZhw/fullscreen) But I don't know what tools I'm able to use at all. I don't want to draw every single frame of that row transitions on my own but I don't know whether there are some tools or frameworks that can allow me doing this kind of things. [23:16:01 CET] <ILove47> And of course, I forgot the question. :) What can I read and look for, so that will help me in reaching my goal? [23:24:41 CET] <KombuchaKip> pzy: This is just using stills. The Viking mission only took stills. [00:00:00 CET] --- Fri Nov 16 2018

1 0

ffmpeg-devel.log.20181114
by burek 15 Nov '18

15 Nov '18

[00:20:15 CET] <akravchenko188> jkqxz: Hi, did you have a chance to review/test amf patches from November 2? [09:30:02 CET] <cone-716> ffmpeg 03Paul B Mahol 07master:759589b55ee5: avcodec/huffyuvenc: add support for gbrp16 [12:03:51 CET] <JEEB> durandal_1707: ok, I think I've got it. the initial sub2video fix that's in master now works, except it blocks the filter chain for quite a while. the one on the mailing list fixes that behavior, but instead it seems to lead to slowness XD [12:07:55 CET] <JEEB> I think I'll have to look into it a bit more later, but at this point I'm almost tempted to just note FU at sub2video's current incarnation [12:11:58 CET] <durandal_1707> sub2video is for bitmap subtitles? [12:34:02 CET] <JEEB> durandal_1707: yea [12:34:28 CET] <JEEB> I tried to document how it works with the patch on the ML [12:35:03 CET] <JEEB> that said it seems ot base a lot on stuff that used to be true/limited [13:48:29 CET] <abi> hi [14:06:34 CET] <cone-041> ffmpeg 03James Almer 07master:752659327d4a: avcodec/libdav1d: fix build after a recent API break [14:53:22 CET] <thardin> what would it take to get listed on the consulting section of the website? [14:56:05 CET] <jamrial> a patch, i guess [14:58:48 CET] <thardin> this is also reminding me that I have some mxfdec patches to develop [17:24:20 CET] <durandal_1707> who will update web page news for 4.1? [17:27:41 CET] <jamrial> it wasn't updated? [17:27:55 CET] <jamrial> huh. i'll do it then [19:26:47 CET] <JEEB> cool, I didn't even notice that the libdav1d wrapper got applied [19:39:48 CET] <nevcairiel> and a week later it broke [19:39:52 CET] <nevcairiel> a good sign that it was too early =p [19:47:04 CET] <cone-041> ffmpeg 03Jan Ekström 07master:b31a4be15f56: lavf/dashenc: set FLAC manifest codec string to "flac" [19:47:38 CET] <JEEB> I didn't even think anyone would use FLAC in MPEG-DASH, but there was at least one user to report it was broken in #ffmpeg [19:47:41 CET] <JEEB> lol [23:59:52 CET] <jamrial> jkqxz: guess you took it from cbs_vp9 [00:00:00 CET] --- Thu Nov 15 2018

1 0

ffmpeg.log.20181114
by burek 15 Nov '18

15 Nov '18

[00:00:57 CET] <JEEB> can't speak for everyone but if it's his own code and he has only used any leaked stuff for hints at how things work, then the result is still under his copyright. not as great as full RE of course :P [00:01:46 CET] <Pdrome> I don't come here to put project contribs noses out of joint or to cause trouble, I just don't feel confident starting to use FFmpeg with regards to a commercial game if RADs leaked code is in use then could 'FFmpeg using gamedevs' be liable? [00:02:41 CET] <JEEB> I don't know what hypothetical scenario we're talking about, nor about juristiction etc [00:02:46 CET] <Pdrome> I promise I will be back with link to the mailing list entry soon as i refind it [00:02:48 CET] <JEEB> nor am I a lawyer [00:03:40 CET] <JEEB> also for the record, you can disable complete functionality in FFmpeg if you don't like it or don't want to distribute binaries with that functionality in it [00:03:59 CET] <Pdrome> ^interesting thanks. [00:04:28 CET] <JEEB> quite a few people do it anyways since FFmpeg has so many minor/barely tested formats implemented [00:11:02 CET] <Pdrome> I should also state that rads source code is still up online after all these years - it is in one of the avp1 source code mirros [00:11:05 CET] <Pdrome> mirrors [00:13:02 CET] <JEEB> I would be surprised if that was the actual RAD codec code [00:13:32 CET] <JEEB> game developers generally don't need to get that far, they just need the headers etc to be able to use bink dlls [00:18:15 CET] <JEEB> also I didn't even notice Kostya had published a WIP patch for bink2 when he gave up [00:18:29 CET] <JEEB> you just caused me to go through his blog posts regarding bink [00:18:30 CET] <JEEB> lol [00:18:53 CET] <Pdrome> it is bink.c [00:19:08 CET] <Pdrome> the headers are also near the C file [00:19:56 CET] <JEEB> ok, don't worry then it seems :P that seems to be their wrapper around the bink interfaces :P [00:20:02 CET] <JEEB> stuff link BINK *binkHandle [00:20:03 CET] <Pdrome> I wonder if avp1 is where the leak that ffmpeg acquired originated from [00:20:56 CET] <JEEB> nope [00:21:05 CET] <JEEB> it literally uses bink's APIs to open a file and play it [00:21:17 CET] <Pdrome> also the DLLs are there as well and also bink_Rad.h [00:21:21 CET] <JEEB> looking at that bink.{c,h} and bink_Rad.h [00:21:42 CET] <JEEB> so MPC might have gotten the bink_Rad.h from something like this back in ye olden days [00:21:47 CET] <JEEB> but that only documents the interfaces [00:21:59 CET] <Pdrome> yeah avp1 was nearly 20yrs old now [00:22:19 CET] <JEEB> so what you would be able to do with this is just know how to take bink dll and play a video with it :P [00:22:25 CET] <Pdrome> and the later avp1 source code release has removed those radgame tools source code files [00:22:49 CET] <Pdrome> Thanks JEEB for the insight [00:22:57 CET] <JEEB> I hope the differencei s clear between these things :P [00:23:50 CET] <Pdrome> it isn't crystal clear (as I am new to gamedev) but it is simple enough for me to appreciate basic understanding given my gamedev research up to this point ;o) [00:26:49 CET] <Pdrome> I know other commercial developers have used FFmpeg in games but without knowing if in the clear or not, most could be unaware of the rad source code leak association to FFmpeg project [00:27:11 CET] <Pdrome> the more I just googled the more I think it was realting to RADs Bink Video [00:30:33 CET] <JEEB> basically, the stuff that avp dump has is 100% useless for FFmpeg [00:30:45 CET] <JEEB> since FFmpeg doesn't want to just use the bink DLLs [00:31:18 CET] <JEEB> also you may poke Kostya yourself if you want to ask some questions to him to be honest [00:31:26 CET] <JEEB> https://codecs.multimedia.cx/2009/07/brief-notes-about-bink/ is his blog [00:34:04 CET] <Pdrome> How would I poke him? [00:34:17 CET] <JEEB> e-mail or otherwise [00:34:44 CET] <JEEB> you can see his e-mail in his commits :P http://git.videolan.org/?p=ffmpeg.git;a=search;h=HEAD;s=Kostya+Shishkov;st=… [00:35:29 CET] <JEEB> originator of http://git.videolan.org/?p=ffmpeg.git;a=commit;h=342c7dfdbb46b1ff778ef142dc… [00:36:13 CET] <Pdrome> Oh nice, what is his link to FFmepg - eg is he the main developer or is he just best to ask as he worked on RE bink so much?# [00:36:26 CET] <JEEB> he is the author of the decoder :P [00:36:30 CET] <JEEB> he reverse engineered it [00:38:04 CET] <Pdrome> I will try to email him, thanks for all your help JEEB [00:45:28 CET] <JEEB> the only thing I find with any relevant keywords from GOOG is where a person mentions that he has access to the Real Video 4 spec, but that he is under an NDA and cannot share it from 2010's IRC logs [00:45:47 CET] <JEEB> which a) isn't Bink b) pretty clear that nothing was passed on [00:46:10 CET] <JEEB> so so far I can just see you making random allegations [00:46:26 CET] <JEEB> possibly out of not understanding what is discussed [00:48:47 CET] <Pdrome> thats is the thing, I don't want to alledge anything, I just want to understand what the situation is regarding what i found about RAD source on an old FFmpeg mailing list entry.. and until I can actually refind that specific mailing list entry, I dont really have any position to ask from [00:49:06 CET] <Pdrome> earlier above I said the allude as that was the best word i could think of [00:49:31 CET] <Pdrome> I dont want to alledge anything as even I am not sure what exactly the situation is [00:49:47 CET] <JEEB> given the history of kostya regarding random game formats getting done (obviously by means of RE/open research) I would also pretty much default to it being RE'd :P [00:49:54 CET] <Pdrome> you have helped JEEB fill in some things for me which is both interesting and helpful [00:50:21 CET] <Pdrome> I have now sent the email so at the very least Kostya will get a laugh out of my obvious ignorance ;o) [00:51:43 CET] <Pdrome> i am 100% sure the mailing list entry was regarding RAD software and not Real [02:22:07 CET] <JEEB> auri_: sent a patch for the FLAC MPEG-DASH identifier to the mailing list [02:25:55 CET] <ealdeguer> Hi FFMPEG Team, [02:25:56 CET] <ealdeguer> I attended Demux in SF, and Will Law(akamai) demoed CMAF low latency. [02:25:56 CET] <ealdeguer> I understood it was possible to package a source to HLS - CMAF with FFMPEG [02:25:57 CET] <ealdeguer> but I'm not sure how and can't find much on google, can you guys help ? [02:26:00 CET] <ealdeguer> I already managed to take a source, encode it with x264, package it as regular HLS with ffmpeg and push it to Akamai, but I wonder how it could work with CMAF ? [02:53:52 CET] <KombuchaKip> Hey everyone. I'd really appreciate a code review of my custom AVIOContext for reading media from memory: https://pastebin.com/NGALFw8Y [02:54:24 CET] <remlap> Hi I am trying to build ffmpeg but I get ERROR: chromaprint not found but I have it installed [03:02:23 CET] <relaxed> remlap: do you have libchromaprint-dev installed? [03:02:27 CET] <remlap> yes [03:02:42 CET] <relaxed> pastebin.com the end of ffmpeg's config.log [03:04:12 CET] <remlap> https://pastebin.com/ieZ28BvT this enough? [03:05:45 CET] <relaxed> /usr/bin/ld: warning: libcodec2.so.0.7, needed by //usr/local/lib/libavcodec.so.58, not found [03:06:57 CET] <remlap> libcodec2-dev install [03:07:34 CET] <remlap> libcodec2-0.8.1 install [03:07:46 CET] <remlap> 'ed [03:09:05 CET] <relaxed> that provides /usr/lib/x86_64-linux-gnu/libcodec2.so.0.8.1 [03:09:39 CET] <remlap> yes so to new? [03:10:35 CET] <relaxed> yes [03:11:05 CET] <remlap> how would I correct for that (noob sorry) [03:11:27 CET] <relaxed> you need libcodec2.so.0.7 installed [03:13:21 CET] <remlap> my os comes with libcodec2.so.0.8.1 so I am bit out of luck? [03:25:57 CET] <remlap> that's weird [03:25:59 CET] <remlap> its building now [03:26:35 CET] <remlap> configure went ok and now make is running, fingers crossed [04:06:16 CET] <PhantomOfNyx> there we go I'm not used to IRC, quick question in libx264, I've kinda succeded getting the quality for my stream i want but the high quality of getting rid of all grain and blurr results in another thing becoming more obvious .... color banding [04:07:46 CET] <PhantomOfNyx> I had truly not noticed until I cleaned out the blur but is there anyone in here with any smart ideas for reducing that or making it blend further, it's only in obvious stuff like areas with a lot of the same color ( loading screens or specifically torches lighting effect in minecraft becomes bad, mainly because it happens in game but it's just amplified 10x on after the encode) <3 [04:53:00 CET] <Hello71> did you turn off dithering [05:22:45 CET] <saucecode_> I'm using OBS to record gameplays, and I've set it up to record an audio streams for the game, and a stream for my microphone. So it outputs a file with a video stream and two audio streams. [05:23:28 CET] <saucecode_> Is there an easy way to merge the two audio streams from the same file into one stream, with copied video? [09:18:00 CET] <bencc> how can I convert mp3 file to mp4 with black video frames? [09:18:42 CET] <bencc> ffmpeg -i audio.mp3 -c:a aac -b:a 128k -vn out.m4a [09:19:05 CET] <bencc> this encode to m4a. now I'm trying to encode to mp4 with black video stream [09:27:41 CET] <furq> bencc: -i audio.mp3 -f lavfi -i color=black:s=1280x720 -shortest -c:a copy out.m4a [09:27:53 CET] <furq> or -c:a aac if you want aac [09:28:05 CET] <furq> no reason to do that unless you need to though [09:29:02 CET] <furq> also that should be out.mp4 although i don't remember how much difference it makes, if any [09:31:43 CET] <bencc> furq: thanks [09:31:46 CET] <bencc> no reason to do what? [09:31:53 CET] <furq> no reason to reencode the audio [09:32:03 CET] <furq> most stuff supports mp3 in mp4 [09:32:27 CET] <bencc> ok. I'll try without reencoding [09:32:53 CET] <bencc> I saw people do "-c:v libx264 -tune stillimage -pix_fmt yuv420p -shortest" [09:33:36 CET] <furq> you don't need either but they won't hurt [09:33:51 CET] <furq> i can't imagine stillimage will have any effect on a totally black frame [09:33:58 CET] <furq> and the color source defaults to yuv420p anyway [09:37:17 CET] <bencc> cool [09:38:28 CET] <ritsuka> apple doesn't support mp3 in mp4, so avoid it if you need to play it back on a mac or iphone [09:48:19 CET] <bencc> ritsuka: I do need safari. I'll avoid it [10:31:33 CET] <llanotheroom> hi, I developed a library based on libav*. It adds run time control when do video audio/video process, such as add new input/output streams, request I frame, etc.. I would like to seek your suggestions, also desired features not supported by current version of FFmpeg which I may give a try then. [10:31:45 CET] <llanotheroom> https://github.com/Xingtao/FFdynamic [13:49:14 CET] <abi> hi [14:24:08 CET] <th3_v0ice> Why would a packet coming from the UDP given by av_read_frame have AV_NOPTS_VALUE for PTS and DTS while having size and data != 0 ? [16:36:02 CET] <fahadash> What is wrong with this? ffmpeg -i .\PostmanInterceptor.webm -filter_complex "[0:v]trim=duration=13[a];[0:v]trim=start=40:duration=60,setpts=PTS-STARTPTS[b];concat=[a][b]" -y Postmaninterceptor.MOV [16:36:24 CET] <fahadash> I am getting this error [AVFilterGraph @ 03e1cbc0] No output pad can be associated to link label 'b'. [16:38:37 CET] <fahadash> Found it nvm [16:47:25 CET] <th3_v0ice> If I have an input video of 12288x5760 and want to have 1080p output without black bars, how can I do that? [16:48:11 CET] <kepstin> th3_v0ice: crop off the sides [16:49:27 CET] <th3_v0ice> How? One more question. The original is in yuva444p10le, specifying pix_fmt yuv420p removes alot of color from the output, is there a way to remedy this? [16:49:29 CET] <furq> or resize to 1920*900 [16:49:45 CET] <furq> that's still technically 1080p [16:49:52 CET] <th3_v0ice> I am more interested in 1080 height, [16:50:31 CET] <furq> going from 10-bit 4:4:4 to 8-bit 4:2:0 is specifically going to remove colour information [16:51:30 CET] <kepstin> th3_v0ice: 10-bit video like that might be HDR, you might have to tonemap it to make it look good [16:51:57 CET] <furq> i'm guessing from the input res that it's from a camera so yeah [16:52:48 CET] <th3_v0ice> Yeah, raw information from camerea. So there is no automatic option that tries to map the colors? [16:53:30 CET] <furq> there's a tonemap filter but i've never had cause to use it [16:53:33 CET] <furq> https://ffmpeg.org/ffmpeg-filters.html#tonemap-1 [16:55:33 CET] <furq> anyway to crop and scale it you want -vf "crop=ih*(16/9):ih:(iw-ow)/2:0,scale=1920:1080" [16:59:31 CET] <th3_v0ice> That did the trick for the video size, thanks! Let me try the tonemapping part. [17:01:51 CET] <th3_v0ice> Guess I am out of luck for that one :) code 3074: no path between colorspaces [17:11:34 CET] <JEEB> you can only convert video to float with zscale [17:11:36 CET] <JEEB> (and back) [19:49:49 CET] <JEEB> auri_: the patch is now in master [19:53:38 CET] <ChocolateArmpits> Is there any practical purpose to "initial_pause" rtsp parameter? [21:52:39 CET] <GuiToris> hey! can you help me with the presets of the x26[45]? I don't understand it. According to the ffmpeg guide they only affect the file size ( quote: The visual quality will be the same) Well I compared ultrafast and veryslow, and ffmpeg guide cannot be true. There are huge differencies between ultrafast and veryslow. [21:53:02 CET] <GuiToris> So I don't know, does it really improve the quality? [21:53:36 CET] <GuiToris> or they are just filters and they look nicer [21:54:01 CET] <JEEB> for the same file size if the only difference is the preset, the slower ones will look better of course [21:54:26 CET] <JEEB> since with each slower preset you enable features and use more CPU time to try and compress the video better [21:55:10 CET] <GuiToris> JEEB: should I use a slower preset or a higher crf number to achieve better quality? [21:55:39 CET] <GuiToris> JEEB: hey, no the file sizes were different [21:55:42 CET] <JEEB> both. you use the slowest preset that is still fast enough for you, and then you tweak the CRF value to your needs [21:56:11 CET] <JEEB> because the results of the CRF value depend on the preset [21:56:25 CET] <GuiToris> ultrafast was big and ugly while veryslow was smaller and nicer (with the same crf value) [21:56:39 CET] <JEEB> yes, the CRF values' results differ between presets [21:56:47 CET] <JEEB> since the preset enables and disables 'eyes' [21:57:42 CET] <GuiToris> I was wondering if let's say crf 28, preset ultrafast is equal to crf 30, preset veryslow [21:58:07 CET] <JEEB> there is no straight relevancy [21:58:25 CET] <JEEB> just pick like 2500 frames or so of a clip that show you what sort of content you're encoding [21:58:28 CET] <JEEB> then decide on your preset [21:58:32 CET] <JEEB> then tweak CRF [21:58:34 CET] <JEEB> that is it [22:01:01 CET] <GuiToris> that's exactly what I did, then it took about 23h to encode the final video, and I wasn't content with the result. I used x265, crf 30 and slower preset. I didn't know which value should I modify and also keep a small filesize [22:01:32 CET] <GuiToris> in the end I chose crf 28, and hoping it'll be good [22:03:04 CET] <GuiToris> during the tests I saw color banding with ultrafast but I didn't with veryslow. I was thinking that it didn't actually improve the quality it just use some kind of blur [22:03:07 CET] <JEEB> GuiToris: that is why you look at the result of those 2500 or so frames? and you make sure you try to see that it looks like a point in the video that is actually of relevant type/movement etc for the rest of that sort of content you are going to encode [22:03:15 CET] <JEEB> also x265 is completely different from x264 internally btw [22:03:35 CET] <JEEB> so any specifics regarding other than x265 has something called crf and something called presets [22:03:43 CET] <JEEB> goes out of the window :P [22:03:46 CET] <JEEB> also x265 can have bugs [22:03:59 CET] <JEEB> in any case, just use relevant samples for the actual CRF tweaking [22:04:00 CET] <GuiToris> I read that x264 crf 23 ~ x265 crf 28 [22:04:21 CET] <JEEB> I have no idea. the ranges were quite different the last I checked x265 [22:04:38 CET] <JEEB> while x265 can use x264's code because of the licensing agreement [22:04:45 CET] <JEEB> it's still very, very different [22:04:56 CET] <JEEB> of course the general idea should still match [22:05:12 CET] <JEEB> pick preset, use actually relevant sample of 2500-3000 frames to test CRF values with [22:07:17 CET] <GuiToris> JEEB: I'd have liked to compare frames. I used something like $ffmpeg -ss 2 -i input -frames 1 output.png does this keep the actual quality? [22:07:39 CET] <GuiToris> I'm not sure whether png modifies the image [22:07:48 CET] <JEEB> for comparison you can use something like vapoursynth editor with ffms2 [22:08:00 CET] <JEEB> has preview, is frame exact etc [22:08:19 CET] <GuiToris> I'll check it, thank you [22:08:45 CET] <GuiToris> does ffmpeg have a similar tool? [22:09:10 CET] <JEEB> no since FFmpeg by itself is not frame exact. you can use video players like mpv for *some* containers, but they make no promises [22:09:30 CET] <JEEB> ffms2 for example indexes the files with FFmpeg's APIs and makes things frame exact by keeping an index [22:10:16 CET] <JEEB> also you can use vapoursynth to do the clip cutting and then you can pipe the output of that vapoursynth script into ffmpeg.c with vspipe [22:10:33 CET] <JEEB> so you can cut different short clips of the longer clip you were going to test your parameters with [22:10:53 CET] <JEEB> because it makes no sense to wait for hours just to figure out if a CRF value is good enough at a certain preset for your content [22:11:16 CET] <GuiToris> it seems vapoursynth is superior to ffmpeg [22:11:17 CET] <JEEB> of course -ss and -t with ffmpeg.c does the same thing with ffmpeg.c if you only need a single clip [22:11:26 CET] <JEEB> vapoursynth is different [22:11:36 CET] <JEEB> and source modules for vapoursynth of course use FFmpeg [22:11:40 CET] <JEEB> the APIs that is [22:11:48 CET] <JEEB> because there's nothing like FFmpeg handling random multimedia [22:12:00 CET] <JEEB> but then vapoursynth is a frame server :P [22:13:00 CET] <GuiToris> thank you for your help [22:13:03 CET] <GuiToris> one last thing [22:13:12 CET] <GuiToris> is flac also a container? [22:13:27 CET] <JEEB> pretty sure what "flac" generally is is FLAC audio in ogg? [22:13:47 CET] Action: JEEB checks libavformat [22:14:12 CET] <GuiToris> My audio files are named something.flac rather than something.ogg [22:14:19 CET] <c_14> isn't flac raw? [22:14:28 CET] <JEEB> oh, just looked through lavf/flacdec.c [22:14:37 CET] <JEEB> it might actually be raw FLAC with a header [22:14:46 CET] <JEEB> also > 2001 copyright header [22:14:51 CET] <JEEB> ye olden times' code [22:15:53 CET] <GuiToris> I only named them ogg when I used vorbis [22:16:34 CET] <GuiToris> I've never used opus but I think I would name it also something.opus, which might not be correct [22:17:01 CET] <c_14> .ogg and .opus are both fine, .opus is an extension for "opus in ogg" [22:17:24 CET] <JEEB> yes, they stopped trying to do raw audio I think [22:17:38 CET] <JEEB> which is a good thing because containers generally let you do stuff like seeking [22:18:05 CET] <GuiToris> so, is .flac also fine? I don't have to use ogg, do I? [22:18:56 CET] <c_14> .flac is what's normally used, but you can pack it into a container if you want [22:20:27 CET] <GuiToris> thank you so much [22:42:48 CET] <trashPanda_> Hello, I am attempting to read in a rtsp stream with ffmpeg and I get numerous "Error parsing AU headers" errors. I am able to view the stream in ffplay, but I want to restream it via ffmpeg. [22:42:57 CET] <trashPanda_> the command line I'm using is [22:43:27 CET] <trashPanda_> "ffmpeg -fflags nobuffer -rtsp_transport udp -i (sourceaddress) -vcodec copy -f mpegts (outaddress) [22:43:35 CET] <Hello71> youtube prefers opus in matroska, dunno why [22:43:40 CET] <Hello71> maybe it's a hls requirement or something [22:43:48 CET] <Hello71> er, not hls, the other thing [22:43:55 CET] <Hello71> dash [23:27:18 CET] <BlackBishop> for a 'Stream #0:0: Video: hevc (Main 10), yuv420p10le(tv, bt2020nc/bt2020/smpte2084), 3840x2160 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc (default)' 'Duration: 02:18:17.50, start: 0.000000, bitrate: 71665 kb/s' I assume I'd need a 71mb/s connection, right ? [23:27:40 CET] <BlackBishop> which should be well within a 802.11ac connection, right ? [23:29:57 CET] <JEEB> hope that your connection actually goes that fast [23:30:33 CET] <JEEB> I think I was able to remux some ultra hd blu-rays into mp4 and then tried streaming them off through wifi onto my TV [23:30:46 CET] <JEEB> but with blade runner f.ex. it just didn't work very well [23:31:00 CET] <JEEB> so I do recommend a cable when possible [23:31:43 CET] <BlackBishop> the AP can do it .. https://www.ubnt.com/unifi/unifi-ap-ac-pro/ [23:31:59 CET] <BlackBishop> and I just noticed, I can play it just fine over http on my macbook pro ( on vlc ) [23:32:14 CET] <BlackBishop> but not on the tv ( sony kd-55xf9005 ) [23:32:23 CET] <BlackBishop> so it's not the wifi [23:32:28 CET] <JEEB> welcome to plastic boxes [23:32:47 CET] <BlackBishop> it also plays great when on usb in the tv :| [23:33:03 CET] <JEEB> wow, in my case the USB input seems worse off than network :P [23:33:14 CET] <BlackBishop> the usb plays perfect. [23:33:47 CET] <BlackBishop> the wire does seem better than the wifi though [23:34:04 CET] <BlackBishop> I tried ssh/ftp/http/nfs, nfs seems to be the best of all [23:34:16 CET] <BlackBishop> but it still stutters and at some point it totally looses audio. [23:36:38 CET] <BlackBishop> let's see if vlc will do a better job than kodi over http [23:45:54 CET] <BlackBishop> vlc doesn't want to play it at all... [23:49:31 CET] <AlexMax> Hi folks. Got a question. From what I understand, FFMPEG can use hardware acceleration to decode h264. However, I'd like to do this from a library. I'm trying to repurpose some existing h264 code, but the problem is that I'm not exactly sure how I'm suppsoed to specify "Hey, please fetch the h264 decoder that is mmal-accelerated". [23:50:12 CET] <AlexMax> ffmpeg lists the decoder as "h264_mmal" [23:50:30 CET] <JEEB> all hwaccels can have some specifics because they might need outside init for system-specific contexts etc, but if it's not a hwaccel then it's "just a decoder" [23:51:04 CET] <AlexMax> Well, I think I have to use -hwaccel to get the decoder to show up [23:51:17 CET] <AlexMax> well...actually... [23:51:31 CET] <AlexMax> if I do ffmpeg -codecs, i get: [23:51:53 CET] <AlexMax> DEV.LS h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (decoders: h264 h264_mmal h264_vdpau ) (encoders: libx264 libx264rgb h264_omx h264_vaapi ) [23:52:09 CET] <JEEB> yea, so try to get the AVCodec by name with the _mmal [23:52:21 CET] <AlexMax> JEEB: Tried that, got an error [23:52:24 CET] <JEEB> ok [23:52:25 CET] <AlexMax> avcodec_find_decoder_by_name("h264_mmal"); right? [23:52:31 CET] <JEEB> yea [23:52:36 CET] <AlexMax> some codec not found error [23:53:15 CET] <AlexMax> "Couldn't find decoder" was the specific error [23:54:04 CET] <AlexMax> To be frank, I know very little about the ffmpeg API and I've been repurposing some existing working code that was built for nearly my exact use-case [23:54:55 CET] <AlexMax> but in the old codebase, ffmpeg was not used to hardware-decode on the raspberry pi, presumably because the ffmpeg provided didn't support it [23:55:21 CET] <AlexMax> Now it does, and I want to reduce code complexity by having as much stuff as possible funnel through ffmpeg [23:55:43 CET] <AlexMax> with maybe a non-ffmpeg fallback on windows [23:56:52 CET] <JEEB> windows has d3d11 and dxva2 [23:57:05 CET] <JEEB> *d3d11va [23:58:48 CET] <AlexMax> Well, more to not have an ffmpeg dependency on windows - building stuff on windows is a pain anyway, and at the end of the day I need a single .dll file with no other depenedncies [23:59:16 CET] <JEEB> dunno, mingw-w64 works comfy enough for me [23:59:32 CET] <JEEB> the only thing I hate about windows dev is that there's no valgrind [23:59:37 CET] <AlexMax> I dunno, that's kinda beside the point [00:00:00 CET] --- Thu Nov 15 2018

1 0

ffmpeg-devel.log.20181113
by burek 14 Nov '18

14 Nov '18

[08:17:28 CET] <cone-337> ffmpeg 03Tobias Rapp 07master:7992c2867efc: avcodec/libopenh264enc: fix class_name [11:40:01 CET] <cone-428> ffmpeg 03Martin Vignali 07master:13ee107afd3d: avcodec/prores_metadata_bsf : fix colorspace check [12:42:29 CET] <thardin> that sscanf() solution sure is something [12:55:03 CET] <thardin> there has to be a better solution that reimplementing it or fork()ing [12:56:16 CET] <durandal_1707> no [12:59:31 CET] <thardin> fflibc when? [13:02:29 CET] <thardin> my inner formal verifier cries a little looking at code like this [13:30:05 CET] <BtbN> locale issues with printf/scanf are a huge pain, and there really does not seem to be a proper solution [13:30:26 CET] <BtbN> You'd wish there'd be something like disable_locale() reenable_locale() you can just slap around it [13:35:00 CET] <thardin> that wouldn't be reentrant [13:35:17 CET] <BtbN> depends on if it's set globally or thread local [13:35:26 CET] <thardin> right [13:36:09 CET] <thardin> this reminds me of errno, and the efforts to try and make it less annoying [13:36:37 CET] <thardin> a libified libc (liblibc if you will) would be useful [13:36:37 CET] <BtbN> errno just seems like a design flaw that somehow dragged on to me [13:38:06 CET] <durandal_1707> tell him that: https://github.com/wm4/libinsanity [13:41:01 CET] <thardin> gpl might be a hindrance (not for me though) [16:33:54 CET] <orbea> Hi, I am having a segfault in the ffmpeg master (as of a few days ago) and older releases when running a single alephone scenerio (game). Here is a backtrace https://pastebin.com/YZ1qSnCC and the alephone issue. https://github.com/Aleph-One-Marathon/alephone/issues/121 My question, is this an ffempg bug or an alephone bug? maybe it could be both? [16:35:01 CET] <orbea> my limited understanding its crashing on an audio frame [16:44:37 CET] <jamrial> orbea: https://github.com/Aleph-One-Marathon/alephone/blob/master/Source_Files/FFm… [16:44:40 CET] <jamrial> sizeof(AVFrame) is not a part of the public ABI [16:45:11 CET] <jamrial> they should allocate a frame with av_frame_alloc(), and not in stack [16:47:09 CET] <jamrial> it's crashing because that frame on stack is uninitialized [16:50:40 CET] <orbea> ah, thanks! [16:50:54 CET] <orbea> I'll make sure that is conveyed to the alephone devs [16:51:25 CET] <orbea> would you mind if I quoted irc for that issue? [16:58:23 CET] <atomnuker> its written in libavutil/frame.h, line 219 in current git master [16:59:31 CET] <orbea> oh, thanks. *looks for the comments* [17:17:20 CET] <JEEB> durandal_1707: finally got some time to test things, and seems like it isn't framesync2 since if I take upstream + revert my first attempt at fixing the sub2video issue (e760c12aeef608aa8b416664687b9aca3a2c6f68) then the performance seems good [17:17:55 CET] <JEEB> it's most likely one of hte sub2video things I've posted since on the ML [17:19:10 CET] <JEEB> I was going to say that it's funny because the affected input had no subtitles to begin with, but oh yes it did :P [17:24:33 CET] <JEEB> it's not completely derped though, since another thing has now run N inputs and two with subtitle overlay [17:24:54 CET] <JEEB> but yea, will be doing more does it vs does it not testing [18:43:15 CET] <durandal_1707> anybody have dot-crawl samples to share? [18:43:59 CET] <atomnuker> yeah, JEEB had one [18:45:39 CET] <atomnuker> https://fushizen.eu/u/jeeb/derp/00002.MPS [18:46:31 CET] <JEEB> ah yes, that's why it popped up during the PSP video discussions [18:46:41 CET] <atomnuker> you'll need to deinterlace, but the dot crawl is monsterous [18:46:56 CET] <JEEB> mistagged full range, dot crawl, ATRAC3+ in MPEG-PS [18:50:44 CET] <atomnuker> eh? the file has audio? [18:51:41 CET] <JEEB> yes [18:51:50 CET] <JEEB> if you apply mistydemeo's patch it probably sounds [18:52:10 CET] <JEEB> although last I checked the parser or something barfs if you try to seek [18:53:16 CET] <atomnuker> where did you get that, can't be from a dvd [18:53:45 CET] <JEEB> UMD Video [18:53:55 CET] <atomnuker> didn't know that was a thing [18:54:06 CET] <JEEB> oh yes, I also have Spiderman on it [18:54:14 CET] <JEEB> since Sony gave them out if you registered your PSP [18:54:14 CET] <JEEB> :D [18:59:04 CET] <gnafu> Classic. [19:24:02 CET] <BBB> michaelni: do you remember what the ffmpeg flag is called to not scale frames? e.g. if the input is scalable, and I want the output frames to be variably scaled also, you mentioned an option a few years back to not scale to one standard size in the output stream, what was that called again? [20:23:51 CET] <michaelni> BBB,i think you refer to "-reinit_filter 0" the intend of that flag IIRC was to send input changes into the filter graph so the filter graph could handle them in any way choosen without loosing state. If the filter graph didnt remove the change then it would be passed on the encoder this was more a sideeffect and possibly usefull but not why the flag was there. It seems the flag doenst work in master anymore [20:26:51 CET] <BBB> ok, thanks, little bit unfortunate but Ill find some other way then [20:26:55 CET] <BBB> thanks for answering [20:50:43 CET] <durandal_1707> anybody against ffvhuff patch? i gonna push it now! [21:43:12 CET] <kierank> durandal_1707: if you are bored you can finish this patch https://trac.ffmpeg.org/ticket/7163#comment:7 [21:54:11 CET] <durandal_1707> kierank: that patch is incomplete [21:54:25 CET] <kierank> "Here is unfinished patch" [21:54:27 CET] <kierank> no shit [21:55:04 CET] <durandal_1707> kierank: unfinished patch is incomplete [21:55:19 CET] <durandal_1707> partial [21:55:28 CET] <durandal_1707> missing some code i think? [21:55:37 CET] <kierank> 12-bit alpha [21:55:52 CET] <durandal_1707> not that [21:57:27 CET] <durandal_1707> do we have 12 bit idct [21:58:47 CET] <kierank> we do [21:58:52 CET] <kierank> that patch produces clean output [22:03:52 CET] <durandal_1707> kierank: write simd code [22:04:10 CET] <kierank> maybe at christmas [22:15:34 CET] <cone-428> ffmpeg 03Andreas Rheinhardt 07master:959521b58de9: h264_levels, h264_metadata_bsf: Fix levels typo [22:15:35 CET] <cone-428> ffmpeg 03Andreas Rheinhardt 07master:3c9c9b1568ef: h264_metadata: Don't use inferred value of buffering frames [23:01:38 CET] <cone-428> ffmpeg 03Michael Niedermayer 07master:6b4c9854daba: avcodec/golomb: Speed up long ur_golomb codes [23:01:39 CET] <cone-428> ffmpeg 03Michael Niedermayer 07master:2f888771cd1c: avcodec/shorten: Fix integer overflow with offset [00:00:00 CET] --- Wed Nov 14 2018

1 0

ffmpeg.log.20181113
by burek 14 Nov '18

14 Nov '18

[00:06:34 CET] <auri_> Hello! I'm currently working with ffmpeg (cli) doing MPEG-DASH encoding/segmentation of lossless (FLAC) audio and have encountered interesting behaviour - it sets "codecs" xml attribute (part of MPD.Period.AdaptationSet.Representation) to "fLaC" and not "flac", where latter is the only way browsers accept it [00:07:18 CET] <auri_> command is `ffmpeg -i <infile.flac> -c:a copy -strict -2 -f dash manifest.mpd`, nothing fancy [00:08:14 CET] <JEEB> hint 1: -strict -2 is actually "-strict experimental" [00:08:31 CET] <JEEB> otherwise i think it's a mktag issue [00:08:51 CET] <BtbN> I think that's the official fourcc for flac [00:09:11 CET] <JEEB> yea, the ogg/flacdec lavf things use that identifier [00:09:14 CET] <JEEB> fLaC [00:09:31 CET] <auri_> if thats the case, should I report it as a bug to mozilla since firefox doesn't accept it? [00:09:39 CET] <JEEB> no, that's separate [00:09:45 CET] <JEEB> need to find the specification for it [00:10:22 CET] <JEEB> https://git.xiph.org/?p=flac.git;a=blob;f=doc/isoflac.txt [00:10:33 CET] <JEEB> > + The codingname of the sample entry is 'fLaC'. [00:10:45 CET] <JEEB> ok, so the mp4 side sample entry identifier is that [00:12:31 CET] <auri_> I was hesitant to ask here because its just the browsers that won't accept it [00:12:40 CET] <auri_> players like vlc and mpv play it back just fine [00:12:48 CET] <JEEB> yes, because mpv f.ex. just uses lavf [00:12:55 CET] <JEEB> VLC has its own mp4 demuxer [00:13:15 CET] <JEEB> but it seems like it was never properly finalized, and the specification doesn't touch the DASH manifest side [00:13:39 CET] <auri_> oh, its always good to stumble on undocumented behaviour [00:13:50 CET] <auri_> right now I just modify the manifest myself [00:14:01 CET] <auri_> s/fLaC/flac/ so all players accept it [00:14:17 CET] <JEEB> yes, so they clearly take teh identifier fLaC inside the container [00:17:10 CET] <JEEB> auri_: I see a manual string identifier in dashenc.c [00:18:09 CET] <JEEB> if you want to test it try adding this line http://up-cat.net/p/6b76769f [00:18:47 CET] <JEEB> if that works then that might just be worth a patch since we've got at least one (1) user [00:19:21 CET] <auri_> lemme build it real quick [00:19:30 CET] <auri_> I'll be back with the results in ~10 minutes [00:22:06 CET] <auri_> JEEB: that worked [00:22:25 CET] <auri_> it now outputs a proper codecs attribute [00:23:11 CET] <JEEB> I'd like that identifier be specified somewhere, although I guess it might be something browser-side/generic content type identifier [00:23:33 CET] <JEEB> but great to know that that's enough for it [00:23:33 CET] <auri_> whatwg wiki says it should be codecs=flac [00:23:44 CET] <auri_> *shrug* [00:23:53 CET] <auri_> oh yes while I'm still here [00:25:14 CET] <auri_> was there ever patch for an option like -initial_offset (-f segment) for dashenc [00:25:38 CET] <auri_> I know Plex fork has it and I skimmed over their impl, seems simple enough [00:26:02 CET] <JEEB> oh yes, people seem to really like that on the muxer side while in general they should just be offsetting their input timestamps a la -itsoffset [00:26:23 CET] <JEEB> (itsoffset being an ffmpeg.c option) [00:26:45 CET] <auri_> that changes segment filenames? [00:26:58 CET] <JEEB> well that sounds like the initial timestamp offset, no? [00:27:03 CET] <JEEB> so that timestamps don't start at zero [00:27:15 CET] <auri_> makes sense yeah [00:27:33 CET] <auri_> so after all the option is there, just in a different way [00:27:49 CET] <auri_> thanks for letting me know [00:27:51 CET] <JEEB> there's a generic ffmpeg.c option for it, but I think you need to utilize -copyts with it or so [00:28:03 CET] <JEEB> or some other -vsync mode than the default [00:28:23 CET] <JEEB> vsync is just a fancy wording for ffmpeg.c dropping or duplicating frames :P [00:28:34 CET] <JEEB> (also the logic can make a grown man cry) [00:28:40 CET] <auri_> I'll see what I can do with it tomorrow, EU times suck [00:28:53 CET] <auri_> basically I need it to handle seeking in a streaming application [00:29:08 CET] <auri_> kill ffmpeg -> restart wit -ss and skipped segments [00:29:37 CET] <auri_> unless there's a better way to handle it I'm not aware of [00:31:43 CET] <JEEB> at some point using the APIs becomes a viable alternative [06:45:59 CET] <KombuchaKip> Hey everyone. I'd really appreciate a code review of my custom AVIOContext for reading media from memory: https://pastebin.com/NGALFw8Y [11:13:36 CET] <termos> I have an incoming stream where time `pts` of the audio and video is out of sync by a constant amount, is there a way to fix this somehow? [11:14:27 CET] <termos> I thought of offsetting one of them by the difference of `pts` in the first incoming packages, but they always seem to have `pts` 0 [11:29:52 CET] <ponyrider> seems like there is a well known issue surrounding trimming a video with -c copy? what is the work around for this? [11:30:24 CET] <ponyrider> the command i have is: ffmpeg -i video.mp4 -ss 00:00:10 -to 00:00:15 -c copy output.mp4 [11:31:14 CET] <durandal_1707> you cant trim exact with copy, it just can not work with lossy encodes with inter frames [11:31:44 CET] <ponyrider> SO says: ffmpeg -ss 191 -i vid.mp4 -vcodec libx264 -acodec libmp3lame -t 6 someoutput3.mp4 [11:32:01 CET] <ponyrider> to re-encode. is this what you would recommend? [11:32:27 CET] <ponyrider> durandal_1707: ^ [11:33:04 CET] <durandal_1707> re-encode will degrade quality [11:33:16 CET] <durandal_1707> with above line [11:33:41 CET] <ponyrider> what is the solution to get a more precise trim as lossless as possible? [11:34:02 CET] <Mavrik> Black magic? [11:35:16 CET] <ponyrider> i just want to trim youtube videos ... :\ [11:36:01 CET] <Mavrik> Noone is stopping you, but there are fundamental limitations around trimming videos stored in modern formats. [11:36:11 CET] <Mavrik> And caring about lossless trimming for YT content is a bit wierd. [11:36:21 CET] <ponyrider> well i dont mind about the losses so much [11:36:32 CET] <ponyrider> just a precise trim. sometimes it works. sometimes it doesnt [11:39:43 CET] <furq> ponyrider: -ss 123 -i foo.mp4 -t 456 -c:v libx264 -crf 20 -c:a copy out.mp4 [11:39:58 CET] <furq> also you probably want to download vp9 from youtube if you're doing further editing to it [11:41:03 CET] <ponyrider> alright if that's the best answer i will take it! [11:41:29 CET] <furq> idk about best but it works and it's not a huge amount of hassle [11:41:54 CET] <furq> in an ideal world you'd only reencode the first and last gop and then stitch it together [11:42:07 CET] <furq> but that is a lot of hassle and if you're clipping six seconds then there's only going to be one or two gops anyway [11:44:36 CET] <ponyrider> i just want accurate trim w no hassle thank you v much [12:00:59 CET] <ponyrider> the -to option is faulty [12:01:08 CET] <ponyrider> its giving me a video of 44 seconds longs [12:02:21 CET] <furq> -to has to be in the same place as -ss [12:02:35 CET] <furq> -ss as an input option and -to as an output option doesn't work [12:02:39 CET] <furq> or it does, but it just does the same thing as -t [12:02:42 CET] <ponyrider> i see so -ss cannot go before -i? [12:02:51 CET] <furq> put them both before -i [12:03:00 CET] <furq> or both after, i don't think it matters if you're reencoding the video [12:03:18 CET] <furq> but yeah -ss/-t/-to behave differently depending on whether they're input or output options [12:04:03 CET] <relaxed> isn't -ss more accurate when placedafter the input -i ? [12:10:57 CET] <ponyrider> ok furq its working well the quality is not any worse either thanks man [12:14:34 CET] <furq> relaxed: i don't think it matters if accurate_seek is enabled [12:14:37 CET] <furq> which it is by default [14:58:54 CET] <somy> hi [14:59:31 CET] <somy> I'm trying to modify ffplay to use hwdec on rockchip and I'm a bit lost [15:00:55 CET] <somy> It' doesn't output anything, I must do something wrong but as soon as it uses the queues to store decoded stuff I get no image.. [15:01:12 CET] <somy> any pointers on where to look for ? [15:55:40 CET] <plalorenz> I want to create a video from images. Let's say I have 1000 images that are named like: "index_dynamic_dynamic_dynamic_dynamic.jpg" so index would go from 0 to 999 and dynamic parts of the file name would be various information. I have tried to create the video with this command: https://pastebin.com/FDwmPX72 but it's not sequential. I know that I am able to say %d in the file name, but how do I mix %d with *? [17:03:00 CET] Last message repeated 1 time(s). [18:22:17 CET] <KombuchaKip> Hey everyone. I'd really appreciate a code review of my custom AVIOContext for reading media from memory: https://pastebin.com/NGALFw8Y [19:01:35 CET] <Hello71> pretty sure that already exists [19:01:46 CET] <Hello71> also eww c++ [19:19:38 CET] <KombuchaKip> Hello71: Sorry, what already exists? [21:29:48 CET] <somy> Does anybody know howto use av_frame_ref / unref / free in hwdec context? [21:50:30 CET] <BtbN> Same way as usual? [21:55:13 CET] <somy> I'm a bit lost either I get a double free, if I use free or the hwedc "explode" trying to free leaked data (because they weren't freed... [22:44:17 CET] <leif> So, is `av_register_all` no longer required to call for ffmpeg 4? [22:46:21 CET] <JEEB> yes [22:46:59 CET] <JEEB> the list is static nowadays since the ability to register formats during run-time was not possible to use from the outside (since it required private symbols) [22:48:17 CET] <leif> JEEB: makes sense. So basically programs can work otherwise the same, just without calling it?) [22:49:19 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/APIchanges;h=db1879e6e23… [22:49:44 CET] <JEEB> search around that for av_register_all and see the notes about newly added functions [22:49:50 CET] <JEEB> but generally, yes [22:53:31 CET] <leif> JEEB: okay cool, thanks. [23:30:42 CET] <Pdrome> Hi is the the right place to ask a question about FFmpeg? [23:31:11 CET] <JEEB> yes [23:32:23 CET] <Pdrome> Thanks. I wanted to ask: Does FFmpeg use any code owned by RAD Game Tools? [23:32:45 CET] <JEEB> not that I know. all of the closed source stuff as far as I know has been reverse engineered [23:33:48 CET] <JEEB> so unless a) RAD game tools contributed some code to FFmpeg (under LGPL) or b) someone got access to RAD Game Tools code and put it in , there should be no RAD Game Tools code in FFmpeg [23:34:26 CET] <JEEB> after reading your question I had to add case a) as well, since I own the copyright to the code I contribute to FFmpeg [23:34:27 CET] <Pdrome> The reason I ask is that I was doing research into gamedev about 2 years ago and found a mailing list entry whereby the FFmpeg devs had got hold of RAD Game Tools source code maybe through a leak etc and I wanted to know what the llegal situation is there ragrding FFmpeg [23:35:39 CET] <pink_mist> the legal situation is that the ffmpeg devs would not be allowed to use said code unless RAD Game Tools specifically gave them a license to do so [23:35:52 CET] <JEEB> yes [23:36:34 CET] <JEEB> but if someone just studied the code and wrote his own, that is the author's thing. [23:36:46 CET] <JEEB> although that's still less kosher than RE [23:37:07 CET] <Pdrome> controversial question I know, but I am hesitant to use FFmpeg for commercial projects given my findings as I dont know if the FFmpeg project could be liable and in turn myself for using FFmpeg. [23:37:47 CET] <JEEB> nothing much controversial about it. [23:38:15 CET] <pink_mist> pretty sure all the code in ffmpeg has been thoroughly vetted so that the authors have given the appropriate license for the code [23:38:20 CET] <JEEB> if someone else's code was used verbatim that's always bad [23:38:29 CET] <Mavrik> Sue Google, see what they say, resolve once and for all :P [23:38:47 CET] <JEEB> if you can provide a reference and ask for details on the mailing list or the issue tracker that could get the ball rolling [23:40:37 CET] <Pdrome> JEEB are you a dev? I can post the mailinglist link regarding the source code possible usage [23:41:07 CET] <Pdrome> It was FFmpegs own mailing list btw. Let me just refind it... [23:41:28 CET] <JEEB> Pdrome: also for the record the history between one of the people implementing RAD game tools' formats and RAD Game Tools seemed to be generally positive. they seemed to winkingly link info on new bink formats etc to the guy [23:42:52 CET] <JEEB> and if someone would come after you for using FFmpeg it'd probably be the IP owners for actually standardized formats [23:43:01 CET] <JEEB> as opposed to something like RAD [23:43:05 CET] <JEEB> IANAL of course [23:44:35 CET] <Pdrome> cheers JEEB, I am slightly reassured. Would love to get input from contribs regarding the mailing list entry I found. Still looking for it... [23:44:45 CET] <JEEB> sure, agree with that [23:49:19 CET] <Pdrome> Doh! Can't find it by re-googling but here's an interesting quote: [23:49:21 CET] <Pdrome> Rad Game Tools claimed that the MPC interface to the DLLs was based on leaked/proprietary information which is why they compelled MPC to remove support. But not before I reverse engineered and documented the DLL interfaces: see http://wiki.multimedia.cx/index.php?title=RAD_Game_Tools_Bink_API and http://wiki.multimedia.cx/index.php?title=RAD_Game_Tools_Smacker_API . RGT once contacted me and requested that I remove those descriptions fr [23:49:41 CET] <Pdrome> from here: https://multimedia.cx/eggs/bink-video-in-ffmpeg/ [23:50:06 CET] <Pdrome> I must have the mailing list entry bookmarked on another machine [23:50:25 CET] <JEEB> ok, FFmpeg doesn't utilize the official RAD APIs [23:50:49 CET] <JEEB> that talks about the interfaces against RAD's own libraries [23:51:14 CET] <JEEB> seems like at some point Gabest's MPC (the media player I guess) had support for the APIs [23:51:43 CET] <Pdrome> I am gutted I can't find the mailinglist entry - it was very clear [23:52:13 CET] <Pdrome> When I do find it I will pop back and post again, thanks so far though for your adive and thoughts [23:52:48 CET] <JEEB> so basically what that means is that mpc used interfaces (probably headers) which were leaked and not under an open source license, and thus incompatible with being in an open source thing (like MPC) [23:53:07 CET] <JEEB> and those interfaces are against RAD's binkw*.dll I guess [23:53:27 CET] <JEEB> now, FFmpeg's stuff is of course completely separate from the bink DLLs [23:54:20 CET] <Pdrome> Here's a hypothetical... If FFmpeg did use RADs algorithms etc, would that be illegal if it was acquired through means of a RAD game tools source code leak from somewhere? [23:55:07 CET] <JEEB> if the code was verbatim RAD code then that's not kosher [23:55:17 CET] <Pdrome> as my above is closest to what I remember the mailing list alluded to [23:57:13 CET] <JEEB> that said, since it's kostya I'd find it rather unlikely that he would have done something like that. esp. given him having fans (?) within RAD seemingly sending him challenges ("psst, did you know that we have *another* new format around? ;)") [23:57:41 CET] <JEEB> and given that he's rather capable in actual RE as it's become obvious during the years [23:57:57 CET] <JEEB> doesn't make it an impossibility of course [23:59:26 CET] <JEEB> hah, he has just started poking at real's new china format according to his own blog :P [23:59:31 CET] <Pdrome> that is my other thought... where would the FFmpeg project stand if it uses rads source code from a leak but it is just rewritten etc + if other aspects not leaked were reverse engineered? [00:00:00 CET] --- Wed Nov 14 2018

1 0

ffmpeg-devel.log.20181112
by burek 13 Nov '18

13 Nov '18

[01:14:07 CET] <cone-065> ffmpeg 03Andreas Rheinhardt 07master:ee47ac97d793: cbs_h2645: Improve performance of writing slices [01:26:37 CET] <philipl> BtbN: do you think it's ok as a single change or should break out the updates for each codec/filter? [01:56:27 CET] <iive> if you do that, would it help bisect or would it make it slower? [02:23:44 CET] <philipl> I think it makes little difference. The most likely problem is I goof one of the codecs/filters and it segfaults. You'd know which change to look that either way. [09:42:27 CET] <cone-674> ffmpeg 03Paul B Mahol 07master:a09411a0ee02: avfilter/af_afftfilt: calculate all channels FFT values upfront [09:42:28 CET] <cone-674> ffmpeg 03Paul B Mahol 07master:0f2cfa3d809b: avfilter/af_headphone: do not leak input frames on error [09:46:00 CET] <cone-674> ffmpeg 03Paul B Mahol 07master:26772b789b95: avcodec/tiff: add initial bayer and sub image support [09:54:13 CET] <durandal_1707> no reviewers - dead project [10:28:43 CET] <dualz> so I have a filter that passes through a video and an audio input 2->2. I am using activate(). My question is there a good guide on syncing audio and video? Looking at my activate callback does it look like the frames are going to get out of sync? https://gist.github.com/ryanmarin/86808a7ea2eccd8117f7855dfdf412e9 [10:29:40 CET] <durandal_1707> it will get out of sync if you do not look at pts [10:31:48 CET] <dualz> what am i supposed to do after I look at them [10:33:48 CET] <durandal_1707> depending on inlink and outlink timebase change pts, but synchronization is done via pts, so i fail to see what you want to do with that filte [10:33:51 CET] <durandal_1707> *filter [10:39:09 CET] <dualz> the filter takes in a video and an audio input (some constant stream such as nullsrc), when it starts it simply passes through the input to the output. Then it has a process_command for opening a file, while the file is open it passes on it's av frames to the output instead of passing through the input. Once that file EOFs it falls back to passing through the inputs [10:39:19 CET] <BtbN> philipl, seems fine to me as one change, but no strong opinion there [10:41:19 CET] <durandal_1707> dualz: then you will need to always change frame pts, if you want to pts appear monotonous(no gaps) and increasing in output [10:43:44 CET] <dualz> is there any filter I can look at that does similarly to what I need? [10:45:22 CET] <cone-674> ffmpeg 03Steven Liu 07master:43cc2e39c009: avformat/hlsenc: check dirname and fix vtt problem [10:45:29 CET] <durandal_1707> dualz: another issue are what about if videos are VFR [10:45:57 CET] <durandal_1707> dualz: look at tpad and loop filters how they change pts [10:50:50 CET] <cone-674> ffmpeg 03Paul B Mahol 07master:29f2893318a2: avfilter/vf_alphamerge: switch to activate [11:35:15 CET] <m2adave> Hello! For our project, we need ATSC AC3 descriptors in the MPEGTS output. This has been discussed before in https://trac.ffmpeg.org/ticket/5837 (associated discussion at http://ffmpeg.org/pipermail/ffmpeg-user/2016-September/033536.html) and https://trac.ffmpeg.org/ticket/2162. [11:35:15 CET] <m2adave> I've started looking at mpegtsenc.c, and adding the registration descriptor is straightforward enough, but adding the AC-3_audio_stream_descriptor (0x81) is complicated because it needs bitstream parameters (bsid, bsmod, acmod etc), and there's no extradata available for AC-3. [11:35:15 CET] <m2adave> Can anyone suggest a way I can access the required parameters? I note that movenc waits for a packet of AC3 and then parses the header, which seems reasonable, but I'm not sure how this works in a streaming scenario where the header should be available before the data. In MPEGTS, the PMT is potentially sent before the first AC3 packet has arrived anyway, so this doesn't seem like an option. [11:35:15 CET] <m2adave> It would be easy to stick the AC3 bsi in extradata or something, but I guess this isn't the right solution. [11:35:15 CET] <m2adave> Thanks in advance for your help. [11:37:21 CET] <nevcairiel> mpegts doesnt have any global headers anyway, its all in-stream, so waiting for data seems trivial to me [11:42:21 CET] <m2adave> @nevcairiel, thanks. That is an option but the problem is that a PMT will be sent at the beginning which does not correctly describe the stream (unless the first data packet sent is an AC3 packet). Then later, when the first AC3 packet arrives, either a forced SI update needs to occur at the first AC3 PES packet, or some packets would be missed. [12:14:48 CET] <cone-674> ffmpeg 03Jun Zhao 07master:991cd7915537: fate: Add test for vc1test demuxer [12:33:43 CET] <cone-674> ffmpeg 03Akemi 07master:233cd8905640: lavc/videotoolbox: Fix cropping with HEVC and H264 videos [12:35:39 CET] <cone-674> ffmpeg 03Paul B Mahol 07master:e96742dd3bdf: avfilter/af_afftfilt: use first input frame timestamp [12:43:50 CET] <cone-674> ffmpeg 03Mark Harris 07master:ae4323548ae8: avfilter/vf_fade: fix start/duration max value [12:49:09 CET] <dualz> durandal_1707: I'm looking at the loop filter and how it keeps the PTS in the context then increments it depending on the number of samples or duration of a frame [12:50:28 CET] <dualz> would I increment it by nb_samples every audio frame and pkt_duration every video? [12:53:08 CET] <durandal_1707> dualz: your case is simpler imho, you just output first inlink with pts from 0 and than when you receive input file you just offset input file pts with last pts you received from inlink [13:13:01 CET] <dualz> so in it's original state the audio and video should be in sync, it's only when I open files do I need to offset the PTS? [13:13:15 CET] <dualz> or am I confused [13:20:34 CET] <durandal_1707> you are not going to compare audio and video pts for sync [13:20:53 CET] <durandal_1707> this could be just V->V and A->A filter [13:22:53 CET] <durandal_1707> offset pts = increase evvery pts received from input file by same ammount received from last inlink, similar pts from inlink after input file finished would be increased by last pts of output frame [14:00:46 CET] <durandal_1707> so we want DOS support but not XP support? [15:09:37 CET] <j-b> Tested-by: ponpon [15:09:38 CET] <j-b> wtf [15:10:46 CET] <J_Darnley> Is there an x86 SIMD instruction that does both pshufd and blendps together [15:13:43 CET] <J_Darnley> A note for others: avoid opening the intel instruction set reference in LibreOffice Draw [15:15:17 CET] <atomnuker> J_Darnley: welp, you can shuffle in zeroes and OR whatever you need on there [15:20:58 CET] <J_Darnley> yeah I could [15:21:28 CET] <J_Darnley> probably best I keep it simple [15:21:52 CET] <J_Darnley> a pair of pshufd and movq [15:24:34 CET] <philipl> BtbN: https://github.com/philipl/mpv/tree/cuda-vkimage [15:25:23 CET] <philipl> BtbN: This is VkImage based interop. I think it should work on windows with 416.xx drivers, if you want to try it out. [15:35:49 CET] <BtbN> philipl, Does that allow bi-directional stuff? Probably now? [15:35:52 CET] <BtbN> *not [15:39:07 CET] <philipl> no [16:31:16 CET] <J_Darnley> what the hell? where is my return instruction? [16:51:25 CET] <Gramner> J_Darnley: shufps may be what you're looking for depending on what order you need the elements to be in [17:16:28 CET] <dualz> durandal_1707: taking it back a step to the original pass through, I am noticing a max of 10FPS to my output, this is my activate() logic: https://gist.github.com/ryanmarin/6b9e12d24b720ae8e485c497948a2926 [17:17:01 CET] <dualz> is there any reason i should be seeing a reduction in framerate [18:51:15 CET] <durandal_1707> dualz: no reduction in framerate for simple pass should happen [18:58:58 CET] <durandal_1707> no comments for av_sscanf expect usual suspects, so i can push it asap? [19:00:52 CET] <jamrial> durandal_1707: add a proper copyright and resubmit [19:00:57 CET] <jamrial> otherwise there will be drama [19:01:49 CET] <jamrial> don't mention every author if it's an impossible task, just state it's from musl [19:20:13 CET] <dualz> durandal_1707: I pasted the two commands I used to get different framerates: https://gist.github.com/ryanmarin/a8ef8cc5d7d8f9b74b2abd4d8b8b696a [19:48:12 CET] <durandal_1707> jamrial: done [19:49:47 CET] <durandal_1707> dualz: and what is CPU usage with feed filter? [19:52:32 CET] <atomnuker> durandal_1707: that'll fail fate, since fate checks for license header [19:53:09 CET] <durandal_1707> what? how? fix it how? [19:53:34 CET] <atomnuker> http://ffmpeg.org/pipermail/ffmpeg-devel/2016-January/186280.html [19:53:42 CET] <atomnuker> look at how I did the license header at the top [19:54:16 CET] <atomnuker> Just replace the xiph license with whatever musl had [19:54:33 CET] <durandal_1707> fate source passed here.... [19:54:52 CET] <durandal_1707> is not smart enough to detect dual license [20:19:48 CET] <durandal_1707> do we need %m in av_sscanf() ? [20:21:59 CET] <durandal_1707> atomnuker: jamrial said it maybe should be better to put this function in separate file, what you think? [20:23:45 CET] <atomnuker> that'd be fine, as long as its still exposed in avstring.h [22:10:25 CET] <cone-728> ffmpeg 03Paul B Mahol 07master:23f589e07340: avfilter/vf_alphamerge: do not check for ret value if all wanted frames are present [00:00:00 CET] --- Tue Nov 13 2018

1 0

ffmpeg.log.20181112
by burek 13 Nov '18

13 Nov '18

[00:55:07 CET] <foo> I'm wanting to convert an MP3 to FLAC (first 30 seconds) only. I can A) transcribe with ffmpeg on filesystem, store as flac, use file, then delete or B) I can do this all in memory somehow and never actually write the file to disk. Anyone happen to know which might be preferred? I know I can do A) really easy... but converting + deleting file sounds unnecessary if I can do it all in memory... although doing it all in memory may use more memory... ... [00:55:13 CET] <foo> ... originating mp3 file can be < 100MB [01:00:41 CET] <hololeap> foo: if you are in linux you could mount a tmpfs to the filesystem and get the best of both worlds [01:01:26 CET] <iive> why -t 30 is not working for your case? [01:01:57 CET] <foo> hololeap: I'm doing this in python, FWIW. I'll delete the resulting file afterwards [01:02:16 CET] <foo> iive: oh, ffmpeg -t 10 -i Jared.mp3 Jared.flac works fine [01:02:29 CET] <foo> I'm wanting to do this on the fly, use file once, then delete it... debating doing this in memory or on FS [01:03:10 CET] <hololeap> depending on your distro, a tmpfs (basically a RAM filesystem) may be mounted at /tmp, so if you store it there temporarily, it may never touch your disks [01:04:33 CET] <foo> hololeap: oh, interesting, I may not be familiar with this [01:04:47 CET] <foo> yeah, since I'm ultimately wanting to tie it into this... with io.open(file_name, 'rb') as audio_file: content = audio_file.read() ... [01:04:54 CET] <foo> I believe with python-flask I could do this too [01:04:56 CET] Action: foo investigates [01:07:37 CET] <somy> hi there, anybody has any experience using hwdec in the libs... I'm totally lost... it's half working I get all the frames in decode order that would be ok if the pts where not equals to the dts... [01:09:00 CET] <foo> hmm, anyone see what I'm doing wrong? https://paste.ofcode.org/tTgRrgUfwVrsNwgTNu5b6Z [01:09:07 CET] <foo> I may have to explicitly pass conversion from mp3 to flac [01:13:59 CET] <foo> ah, nevermind [01:15:27 CET] <foo> ]trim=duration=10[s0] matches no streams [01:15:49 CET] <foo> I see the filter here https://github.com/kkroening/ffmpeg-python/blob/master/ffmpeg/_filters.py [01:17:07 CET] <foo> .filter('trim', duration=10) looks right too. /me scratches head [01:22:17 CET] <foo> .filter('trim', start=0, end=10) still throws Stream specifier '' in filtergraph description [0]trim=end=10:start=0[s0] matches no streams. [01:22:20 CET] <foo> huh [01:27:19 CET] <foo> aha, got it [02:10:56 CET] <Buliarous> hey so i want to break a movie up into pieces using segments or something such that each piece is no larger than 128mb (maybe 120mb to be safe) is there an easy way to do this? [02:19:42 CET] <relaxed> Buliarous: I recommend mkvmerge for this: https://mkvtoolnix.download/doc/mkvmerge.html#mkvmerge.description.split [02:20:23 CET] <relaxed> if matroska is an ok output container [02:20:59 CET] <Buliarous> must be avi unfortunately ... but that is good to keep in mind for the future [02:23:29 CET] <relaxed> https://ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c… [02:25:49 CET] <k-man> hi [02:26:57 CET] <relaxed> Buliarous: might be easier to use mkvmerge and then remux to avi with ffmpeg [02:32:42 CET] <Buliarous> relaxed: thanks! i usually work with mkv so definitely will be keeping that in mind for later. I found this to work for my purposes: https://superuser.com/questions/712893/how-to-split-a-video-file-by-size-wi… The split-video.sh script works really well and you can do something like ./split-video.sh myMovie\ transcode.avi 120000000 "-vcodec copy -acodec copy" to split into pieces of [02:32:48 CET] <Buliarous> 120mb and keep the video and audio codecs, which is pretty great [02:34:10 CET] <relaxed> that looks pretty hacky [02:34:25 CET] <Buliarous> kudos to LukeLR whoever you are [02:41:27 CET] <Buliarous> relaxed: sure is [16:10:07 CET] <th3_v0ice> For UDP input, is FFmpeg automatically setting cicrular buffer or some other parameters? I am getting "Input/Output Error" while reading UDP with av_read_frame(). Circular buffer overflow is shown above it. [17:07:38 CET] <JEEB> th3_v0ice: that just sounds like you're running out of buffer because you're not processing stuff quickly enough [17:07:54 CET] <JEEB> if you need more buffer you can configure that [21:21:20 CET] <th3_v0ice> JEEB: Ok, thanks. Other things point out that that seems to be the case. [00:00:00 CET] --- Tue Nov 13 2018

1 0

ffmpeg-devel.log.20181111
by burek 12 Nov '18

12 Nov '18

[00:12:37 CET] <cone-517> ffmpeg 03Andreas Rheinhardt 07master:6dafcb6fdb62: h264_redundant_pps: Fix logging context [00:12:38 CET] <cone-517> ffmpeg 03Michael Niedermayer 07master:c305e134ce23: avcodec/imm4: Use ff_set_dimensions() [02:13:30 CET] <cone-517> ffmpeg 03Jun Zhao 07master:81f2a9f136ca: fftools/ffmpeg: Remove the micor like "#if 1" [02:13:31 CET] <cone-517> ffmpeg 03Jun Zhao 07master:3eccf5be9287: fftools/ffmpeg: Put the variable declaration at uppper for block. [02:13:32 CET] <cone-517> ffmpeg 03Jun Zhao 07master:e24a754916be: fftools/ffmpeg: Indent the code [12:23:06 CET] <BtbN> If I << a char, it will be auto promoted to an int, right? And not just shift out stuff after the first 8 bit? [12:28:34 CET] <atomnuker> I'd cast it to make it obvious and be on the safe side [12:45:55 CET] <thardin> I believe most/all artithmetic promote to at least int [12:46:16 CET] <thardin> being explicity might be nice tho [13:42:35 CET] <cone-542> ffmpeg 03Paul B Mahol 07master:8860d307311a: avfilter/vf_lut3d: ignore last whitespace when comparing LUT size string [13:53:23 CET] <durandal_1707> what about replacing locale dependent functions that we use with native variants? so . is always used and not ,? [13:59:33 CET] <atomnuker> yes, that would be awesome, lavu already replaces some so there can't be that many left [15:08:38 CET] <durandal_1707> sscanf is not trivial to do [15:12:22 CET] <atomnuker> yeah, seems so, wm4 wrote https://github.com/wm4/libinsanity and its strtod is complex and gplv3 [15:13:31 CET] <durandal_1707> that code is useless for us [15:16:13 CET] <atomnuker> yeah, but you could rewrite parts of it to use in sscanf [15:23:32 CET] <durandal_1707> atomnuker: why? i think about just using one of BSD implementations [15:26:00 CET] <iive> what the license of musl? [15:26:06 CET] <iive> what is... [15:26:21 CET] <durandal_1707> AGPL+ [15:28:00 CET] <durandal_1707> MIT actually [15:42:46 CET] <durandal_1707> are there other functions that handle . for floats differently? [15:46:16 CET] <durandal_1707> also should i add new header lavfu/avstdio.h or put it into lavu/avstring.h ? [15:49:42 CET] <durandal_1707> anyone dare to answer? [15:50:04 CET] <atomnuker> I think this belongs in avstring.h [15:50:07 CET] <BtbN> Can't ffmpeg just force the locale to C on startup or something? [15:50:21 CET] <atomnuker> libraries shouldn't mess with locales [15:50:38 CET] <BtbN> Was thinking more of the cli tool [15:50:58 CET] <atomnuker> also the c.utf8 locale isn't available everywhere [15:51:14 CET] <durandal_1707> its our mess, we need to fix it, and this is long issue [15:51:19 CET] <JEEB> mpv got some weird PR regarding that which doesn't build on windows [15:54:42 CET] <JEEB> durandal_1707: btw do you want to give any hints how a filter chain could be profiled? http://up-cat.net/p/e9f4a0cc [15:56:12 CET] <durandal_1707> JEEB: profiled for what kind of issues? [15:56:26 CET] <JEEB> blocking/slowness [16:03:11 CET] <durandal_1707> JEEB: you could try to insert graphmonitor filter to see if something queues to much frames, and to see what is outputing frames at least rate, you will need to change your graph, and add second video output which you could encode to and watch immediately in separate process (no pipe, as that would block others) [16:04:34 CET] <durandal_1707> do you work with files? [16:05:05 CET] <JEEB> generally live input, although you can always dump the input into a file if a reproducible thing is required [16:05:18 CET] <JEEB> although that might not repro it because the input can get new stuff right away [16:06:10 CET] <durandal_1707> if that bug happens after parameters change midstream that is currently not supported generally by lavfi [16:06:39 CET] <JEEB> I don't think that's the case [16:07:57 CET] <durandal_1707> then slowest encoder may slow others [16:11:36 CET] <JEEB> yea, I'll probably be able to give more info later next week, one of my hypotheses was that pre-framesync2 seemingly handled stuff better compared to the later versions. but I'll have to double-check that it isn't the sub2video patches I've posted on the ML fixing issues with overlays receiving premature EOS or ending up unscaled after a re-init [16:13:21 CET] <JEEB> although it seemed to be there even during the first sub2video fix version (which didn't hit the premature EOS root reason properly) [16:13:37 CET] <JEEB> and which went to master because it passed the single FATE test for it [16:13:59 CET] <JEEB> (did I mention how much I €!%!& sub2video?) [16:14:27 CET] <JEEB> it seems to be confusingly overcomplicated for a thing that's just supposed to provide overlay inputs [16:15:59 CET] <durandal_1707> sometimes filter my just hang for whatever reason, usually at eof [16:18:02 CET] <JEEB> well, EOF shouldn't be a problem. false EOS can happen with single filter chain and reconfig of say audio [16:18:38 CET] <JEEB> but false EOS generally just results in overlay's secondary input stopping, but output seems to work [16:18:46 CET] <JEEB> just that you stop getting subtitles overlayed [17:33:07 CET] <jkqxz> jamrial: Got a stream lying around with the alternative transfer characteristics message in? [17:35:54 CET] <jamrial> no, but koda probably does. he's the one that added support to the decoder [17:36:26 CET] <jamrial> you can look at it later in any case. no need to delay that patch just for it [17:36:49 CET] <jkqxz> I've already written it (took two minutes). Would be nice to test! [17:36:57 CET] <jamrial> ah :p [17:40:40 CET] <jkqxz> None of the conformance streams have it. [17:44:05 CET] <durandal_1707> atomnuker: see my quick & dirty hack skills on ML - comments welcome [17:46:34 CET] <durandal_1707> i copied parts of musl MIT code [17:47:23 CET] <jkqxz> FFFFFILE is a funny type name. [17:48:45 CET] <cone-542> ffmpeg 03Mark Thompson 07master:fd1d735c0b98: fate/libavcodec: Fix config dependency of h264-levels test [17:53:00 CET] <jamrial> jkqxz: it's a relatively recent addition for backwards compat. decoders compliant with and old spec will use the standard trc reported in the bitstream, new ones will look at the sei and use that instead [17:53:17 CET] <jamrial> probably to support hdr videos on old decoders or such [17:57:58 CET] <jamrial> jkqxz: x265 --atc-sei option generates one [17:58:17 CET] <jamrial> not ideal, i guess [18:00:21 CET] <jkqxz> I'm looking at some random HLG samples now. (<https://4kmedia.org/travelxp-4k-hdr-hlg-sample/> lacks the SEI message, but it does kill Intel GPUs with VAAPI. Oops.) [18:02:01 CET] <JEEB> the travelxp sample I think has general corruption issues [18:02:44 CET] <rcombs> jkqxz: have you seen dolby vision samples [18:03:56 CET] <rcombs> they went and made up a color space [18:03:57 CET] <jkqxz> Hmph. That actually seems to have messed the GPU up pretty badly; I probably need to reboot to get it back. [18:04:27 CET] <rcombs> wow [18:04:32 CET] <nevcairiel> doesnt DV use ICtCp? That was added to the standards some time ago [18:04:41 CET] <rcombs> nevcairiel: not quite [18:04:46 CET] <jkqxz> rcombs: Trying to avoid them. [18:04:50 CET] <rcombs> it's something like ICtCp but not quite [18:04:50 CET] <JEEB> nevcairiel: they adjust it. just adding base ICtCp support with BT.2020/PQ doesn't cut it [18:05:01 CET] <rcombs> here have some samples https://dl.dropboxusercontent.com/s/bsgtnhnix56zutb/Aerial-vid-URLs.rar [18:05:04 CET] <nevcairiel> of course DV is not PQ [18:05:15 CET] <rcombs> we're pretty sure it's PQ [18:05:22 CET] <JEEB> also we're specifically talking about profile 5 [18:05:28 CET] <JEEB> which is the backwards-incompatible one [18:05:44 CET] <JEEB> and which uses a bastardization of PQ+ICtCp etc [18:06:33 CET] <durandal_1707> what is that? [18:06:41 CET] <rcombs> happy fun times https://gist.github.com/a8c91d940d4291120bb677eafc96e41d [18:07:41 CET] <JEEB> ah, so that's the parser for the DOVIConfigurationBox [18:07:47 CET] <nevcairiel> (note that the first part of the commit is technically wrong, a decoder should be capable of unsetting those things) [18:08:23 CET] <rcombs> nevcairiel: I'm not sure if it should in the "the bitstream doesn't have those values at all" case, though [18:09:34 CET] <nevcairiel> where else, the decoder should keep the context up to date with what the bitstream says, and if it says unspecified, then it should set that [18:09:36 CET] <rcombs> I've also been looking into this mostly just out of curiosity; the only files I've seen using it are random online demos/trailers, and those apple Aerial screensavers I just linked [18:09:36 CET] Action: nevcairiel shrugs [18:10:20 CET] <rcombs> well if the decoder overwrites with "unspecified" after the container set something, that leaves the consumer with no way to find out what the container said [18:10:43 CET] <nevcairiel> or maybe there needs to be a proper api to separate that information [18:10:50 CET] <nevcairiel> what if the stream contains conflicting into [18:10:50 CET] <rcombs> maybe [18:10:54 CET] <nevcairiel> you still wont know [18:11:01 CET] <rcombs> fair point, and I think one of apple's streams actually does that [18:14:47 CET] <jkqxz> <https://4kmedia.org/lg-cymatic-jazz-hdr-hlg-uhd-4k-demo/> contains the message, but it also has broken AUDs so requires a bit of hacking to make CBS read it. [18:16:21 CET] <jamrial> i have the hdr10 version of that video, and it doesn't have the sei [18:17:14 CET] <jkqxz> HDR10 doesn't have the backward compatibility of HLG, so it would make less sense there. [18:19:09 CET] <jkqxz> <https://0x0.st/slfp.txt> [18:19:43 CET] <jkqxz> (With rbsp_stop_one_bit being zero in the AUDs. Yay.) [18:24:52 CET] <cone-542> ffmpeg 03Mark Thompson 07master:252e79663de8: cbs_h265: Add PTL parsing for sublayers [18:26:50 CET] <jamrial> extract_extradata should probably also extract prefix sei messages i guess [18:29:52 CET] <jkqxz> It only wants some message types, though. [18:30:42 CET] <jkqxz> And maybe not necessarily all of the time if they they change through the stream, which they could do independently of the parameter sets. [18:54:11 CET] <nevcairiel> SEI messages have a defined lifetime, those with the highest one might make sense to extract [19:17:12 CET] <durandal_1707> why: ffmpeg -i MUSIC_WITH_COVER -i ONLY_MUSIC -lavfi amix=inputs=2:duration=shortest -f null - stops when first video frame is encoded? [20:24:37 CET] <cone-542> ffmpeg 03Andreas Rheinhardt 07master:6df9020f45ea: cbs_mpeg2: Improve performance of writing slices [20:28:53 CET] <philipl> BtbN: https://github.com/philipl/FFmpeg/commit/1c915dfc426512288559cd8f18cba382ec… [20:29:00 CET] <philipl> Common CHECK_CU implementation. [20:35:53 CET] <cone-542> ffmpeg 03Marton Balint 07master:6c2a7a8e9a36: avfilter/vf_framerate: factorize SAD functions which compute SAD for a whole frame [20:35:54 CET] <cone-542> ffmpeg 03Marton Balint 07master:7748f395de88: avfilter/vf_select: use common scene sad functions [20:35:55 CET] <cone-542> ffmpeg 03Marton Balint 07master:936d18fb42bb: avfilter/vf_minterpolate: use common scene sad functions [20:39:15 CET] <nevcairiel> philipl: public api for that makes me feel bad [20:47:48 CET] <philipl> nevcairiel: alternative? [20:50:21 CET] <jkqxz> nevcairiel: +1 [20:50:36 CET] <jkqxz> philipl: Include it in each library separately? [20:51:30 CET] <philipl> Ok. It seems we need three copies. That's still less than eight I guess. [20:54:10 CET] <philipl> Is there a legal way to share a single .c file between the various libs? [20:55:42 CET] <jkqxz> #include [20:56:23 CET] <philipl> I mean file placement... [20:56:32 CET] <philipl> Do we do this anywhere today? [20:56:47 CET] <wbs> there's a few cases, look at the various log2_tab.c [20:56:52 CET] <JEEB> :) [21:01:27 CET] <jamrial> avpriv? [21:01:46 CET] <jamrial> also, what wbs said works as well [21:07:38 CET] <cone-542> ffmpeg 03Martin Vignali 07master:0aba92d42d51: avcodec : add prores_metadata bsf for set the color property of each prores frame [21:07:39 CET] <cone-542> ffmpeg 03Martin Vignali 07master:679ad3146972: fate/prores_metadata_bsf : add test for setting color property [21:15:03 CET] <philipl> avpriv just means I change the prefix? [21:15:09 CET] <philipl> That seems the easy option [21:21:50 CET] <wbs> the problem with that is that even though it's supposed to be private, for shared libraries it more or less ends up as something that needs to be kept stable anyway, depending on how flexible linking setups you want to consider [21:22:51 CET] <philipl> This would be stable in practice. We just don't want it to be part of the public API, right? [21:24:04 CET] <philipl> But Ok, I see how I can reproduce what is done for log2_tab.c [21:26:22 CET] <wbs> log2_tab.c is fine as long as the amount of data/code is small so adding a few bytes per library is easier than handling the symbol across dynamic linking boundaries (even though iirc such things should work fine even for msvc setups now) [21:26:54 CET] <philipl> It's a single small function. [21:37:40 CET] <cone-542> ffmpeg 03Paul B Mahol 07master:bdc66c50dd69: avfilter/af_afftfilt: extend filter functionality [21:37:41 CET] <cone-542> ffmpeg 03Paul B Mahol 07master:d03030c07109: avfilter/af_afftfilt: add more window types [21:44:26 CET] <cone-542> ffmpeg 03Martin Vignali 07master:752bf1f64c4c: fate/prores_metadata : fix md5 value [21:44:35 CET] <philipl> BtbN: https://github.com/philipl/FFmpeg/commit/be8f30cc355d553003d0cb9772f872d752… [21:44:46 CET] <philipl> and everyone else who had opinions :-) [21:45:12 CET] <BtbN> looks sensible to me on first glance [22:13:49 CET] <JEEB> > Let me add that split filter derived sources seems to be also single threaded. [22:14:00 CET] <JEEB> hmm, I wonder if that's correct [22:22:03 CET] <durandal_1707> it is not correct for asplit and audio, same for video (didn't tested video) [22:33:35 CET] <durandal_1707> but after split, slowest one will slow all others after split [23:43:51 CET] <rcombs> nevcairiel: so apparently tls_schannel hard-fails if the OCSP service is down [23:43:54 CET] <rcombs> any idea how to not [23:44:21 CET] <rcombs> fails in InitializeSecurityContext [23:51:23 CET] <nevcairiel> no clue [00:00:00 CET] --- Mon Nov 12 2018

1 0

ffmpeg.log.20181111
by burek 12 Nov '18

12 Nov '18

[14:39:26 CET] <tombb> Hi All, can ffmpeg do LKFS? I need to convert a file and normalize the audio to -24db LKFS... is that possible? [14:41:19 CET] <JEEB> "Loudness units relative to full scale is a synonym for LKFS that was introduced in EBU R128." [14:41:37 CET] <JEEB> so I would guess an audio normalization filter applying EBU R128 should be good enough? [14:41:55 CET] <JEEB> https://www.ffmpeg.org/ffmpeg-all.html#loudnorm [14:42:00 CET] <JEEB> see this filter's documentation [14:45:00 CET] <tombb> oh! lufs == lkfs then? [14:45:05 CET] <tombb> cool - thank you! [14:46:10 CET] <JEEB> it seems so at least according to a random quote from wikipedia [14:46:19 CET] <JEEB> given that broadcast went for the EBU R128 spec [14:46:28 CET] <JEEB> I would expect there to be relevant parameters :P [14:54:26 CET] <tombb> stupid EBU :P - thanks! [15:39:57 CET] <tombb> what would be the right way to combine filter_complex and loudnorm? I would like to split one stereo stream to 2 monos (pan=1c|c0=c0[left];[0:1]pan=1c|c0=c1[right]) and also control their loudness [15:41:19 CET] <tombb> (the stereo to mono part works fine, but I can't add filter_complex and loudnorm in the same command) [15:41:47 CET] <durandal_1707> pan=1c|c0=c0,loudnorm=....[left];[0:1]pan=1c|c0=c1,loudnorm=.....[right] [15:42:23 CET] <furq> -lavfi channelsplit[a0][a1];[a0]loudnorm=xyz[out0];[a1]loudnorm=xyz[out1] [15:57:06 CET] <tombb> awesome! thank you guys [22:51:19 CET] <hololeap> i'm trying to broadcast a file to the network with RTP, like so: ffmpeg -i example.mp3 rtp://10.0.0.255 [22:51:51 CET] <hololeap> but i get an error: [22:51:51 CET] <hololeap> [NULL @ 0x564dac4314d0] Unable to find a suitable output format for 'rtp://10.0.0.255' [22:51:51 CET] <hololeap> rtp://10.0.0.255: Invalid argument [22:53:28 CET] <JEEB> you need to select a muxer [22:53:34 CET] <JEEB> since there's no extension or anything for RTP [22:55:09 CET] <hololeap> is there a passthrough-style muxer? i don't need to re-encode it, just broadcast it [22:56:04 CET] <JEEB> those are separate [22:56:11 CET] <JEEB> -c copy is for copying AVPackets from input to output [22:56:18 CET] <JEEB> -f XXX is for the container format [22:59:15 CET] <hololeap> [rtp @ 0x55f4c8500a20] Data doesn't look like RTP packets, make sure the RTP muxer is used [22:59:22 CET] <hololeap> what is the flag for setting the muxer? [22:59:47 CET] <JEEB> -f XXX [23:00:29 CET] <JEEB> there's at least two muxers that have rtp in their name [23:00:32 CET] <JEEB> rtp and rtp_mpegts [23:04:20 CET] <hololeap> yeah, i'm not sure what's going wrong: https://paste.pound-python.org/show/98jf0P6eAPjDVnNf8AEe/ [23:05:45 CET] <JEEB> well for one thing you've forgotten the "please do not re-encode this" switch [23:05:51 CET] <JEEB> which I mentioned earlier [23:06:35 CET] <JEEB> the rest I don't know but you might also need -re before -i INPUT to simulate reading the input with a specific speed [23:06:45 CET] <JEEB> which basically sleeps between reads :P [23:06:53 CET] <hololeap> right... and i figured out the other part. it was apparently trying to use a <1024 port as non-root. it seems to be working now. [23:13:05 CET] <hololeap> is there a way to slow down the transcoding so that it outputs at 1:1 speed in relation to the length of the file? it seems to be trying to output as fast as possible [23:13:38 CET] <JEEB> I mentioned an option for the sleep()s ;P [23:13:55 CET] <hololeap> oh, sorry :) [23:18:14 CET] <hololeap> it's working great, thanks JEEB ... except that it is broadcasting to the address i specified over the wrong interface. it is using lo... you wouldn't happen to know how to change that, would you? [23:19:12 CET] <JEEB> I would looke at hte AVOptions for the rtp protocol [23:19:44 CET] <JEEB> https://www.ffmpeg.org/ffmpeg-all.html#rtp [23:19:51 CET] <JEEB> there's probably a way to get it through teh command line as well [23:20:57 CET] <hololeap> hmm... yeah i don't see anything that talks about the interface unfortunately [23:21:31 CET] <hololeap> i might be able to work around it with iptables [23:21:55 CET] <JEEB> yea, the UDP protocol had an option -localaddr [23:22:06 CET] <JEEB> which let you force the interface with a specific address [23:22:18 CET] <JEEB> (it was useful with multicast if your routes were incorrectly set for receiving) [23:23:40 CET] <JEEB> I think the RTP protocol underneath uses the UDP protocol but to be honest I've not looked into RTP in lavf enough :P [23:26:54 CET] <hololeap> i tried rtp://10.0.0.100:5000?localaddr=10.0.0.104, but it didn't make a difference [23:27:37 CET] <JEEB> yea because the rtp thing doesn't take that thing in and you have to separately pass things through :P [23:27:46 CET] <JEEB> within lavf [23:28:17 CET] <JEEB> and tbh I didn't check enough of the code to see if the UDP code is actually using stuff from udp.c [00:00:00 CET] --- Mon Nov 12 2018

1 0

ffmpeg-devel.log.20181110
by burek 11 Nov '18

11 Nov '18

[02:02:31 CET] <philipl> BtbN: https://github.com/philipl/nv-codec-headers/commit/7b88dd0250e3cc8055e9e639… [02:02:56 CET] <BtbN> So it was the syncornization after all? [02:03:27 CET] <philipl> No [02:03:38 CET] <philipl> But I got synchronization working and that will be necessary [02:04:09 CET] <philipl> It's not really possible to do a pool strategy with images like I did with the buffer implementation [02:04:26 CET] <philipl> so I need synchronization to make sure we don't copy the next frame until the previous one has been read out of the image. [02:04:46 CET] <BtbN> I'll have a look tomorrow, way too late right now [02:04:52 CET] <philipl> Sure. I'm surprised you're up [02:05:51 CET] <BtbN> Well, there was a certain chess match. [02:06:04 CET] <BtbN> That ended up taking almost 8 hours. [02:47:52 CET] <cone-221> ffmpeg 03James Zern 07master:6d3890ee34fd: libvpxenc,vp9: add enable-tpl option [02:59:13 CET] <cone-221> ffmpeg 03James Zern 07master:55e021f39b02: libvpxenc: extend auto-alt-ref range [10:53:12 CET] <BtbN> philipl, lgtm. I'll wait a bit until doing another round of releases, in case something else comes up, which has happened surprisingly frequently lately. [16:46:39 CET] <philipl> BtbN: thanks. Would you take the change with the kernel launch API stuff? It's not actually very much. [16:47:46 CET] <philipl> https://github.com/philipl/nv-codec-headers/commit/692ef24e3242286a0b6bfa4d… [16:47:49 CET] <philipl> for reference [16:49:25 CET] <BtbN> Well, it won't make it non non-free [16:49:47 CET] <philipl> I know. [16:50:04 CET] <philipl> But it at least scopes the problem to the ptx compilation [16:53:36 CET] <BtbN> hm, I guess it also makes the binary at least runable without the nvidia driver installed, which might be a plus to some internal deployments. [16:58:13 CET] <philipl> Yeah [17:07:56 CET] <BtbN> Go ahead then [17:26:17 CET] <philipl> Ok. I'll push these but I won't tag for release. Yeah? [17:55:33 CET] <philipl> BtbN: https://opensource.stackexchange.com/questions/4304/redistributing-lgpl-2-1… [17:55:43 CET] <philipl> not much clarity [17:56:54 CET] <BtbN> Reads pretty clear to me [17:57:09 CET] <BtbN> Unless we get it to compile and work with clang, it's non-free. [17:57:18 CET] <philipl> I investigated that. [17:58:17 CET] <BtbN> I wonder how things would look if the ptx would be commited. But I doubt that'd be accepted. [17:58:40 CET] <philipl> There are missing language features, and, I think, the thing that sinks the whole ship, is that you end up linking in a copy of cudart to provide the cuda api implementations. [17:58:51 CET] <philipl> It's not like there's a libc equivalent handing out on the GPU side. [17:59:37 CET] <philipl> The other thing I was wondering about was what if the ptx was loaded as a separate file off the filesystem. That means you can rebuild the lgpl part any time you want however you want. and the ptx runs on the card (so separate address space, etc) [18:00:00 CET] <philipl> Arguably, the ptx being inside the library is aggregation and not linking. [18:00:10 CET] <philipl> but it makese the replacement requirement much harder [18:01:10 CET] <philipl> So you could say "Here is ffmpeg". "If you download this ptx and place it in the right place, the filter works" [18:02:35 CET] <BtbN> I don't think there is much arguing here. It's just non-free [18:02:46 CET] <philipl> But there's an important point there. [18:03:01 CET] <philipl> If you do what I just said what about the ffmpeg build is non-free if the ptx isn't in there? [18:03:22 CET] <philipl> The argument then is that the ptx is non-free and how do you get that [18:03:29 CET] <BtbN> It seems like cheap trickery to me [18:03:51 CET] <BtbN> But you'll have to discuss that with an fsf lawyer [18:04:18 CET] <philipl> It's not cheap. If you say the ptx is non-free and non-redistributable, the functionality is just as unavailable as it was before. [18:05:10 CET] <philipl> I don't have have any commerical motivtions behind any of this; I'm just interested in maximising availability to end users. [18:05:33 CET] <BtbN> I'm not against that at all, but that solution seems as anti-end-user as it gets. [18:05:56 CET] <BtbN> (And would also be an API break) [18:06:26 CET] <philipl> Sure. [18:06:31 CET] <philipl> I suspect this is all futile anyway. [18:06:52 CET] <philipl> It's really the proprietary compiler question. Everything else is rearranging deck chairs on the titanic. [18:16:07 CET] <cone-517> ffmpeg 03bnnm 07master:27e114b4511b: avcodec/wmaprodec: improve XMA missing samples [18:18:19 CET] <uau> hevc frameskip still seems broken (https://github.com/mpv-player/mpv/issues/6218) there's also a patch from Mark Wu on the mailing list without replies (from 2018-10-23) [18:18:37 CET] <uau> could someone fix/apply that? [18:20:14 CET] <philipl> BtbN: And now the pix fmt question. The easy/lazy thing is to punt on pix fmts for now and just do 444p16. Otherwise it's organising a vote?!? [18:20:33 CET] <BtbN> Seems like it. [18:21:56 CET] <philipl> Even though I'm in favour of the new pix fmts, I'd feel better if I could find some discussion of the industry position on this (like there is for P010). But I can't find any. [18:23:24 CET] <philipl> We know the nvdec docs won't help - they only talk about P016 today and so I'm sure they'll just say 444p16 when they are updated. [18:29:56 CET] <jamrial> uau: you don't have push access to the repo? [18:32:06 CET] <uau> no [18:32:50 CET] <jamrial> ok, will push in a bit [18:38:12 CET] <cone-517> ffmpeg 03Mark Wu 07master:10bc4c3a7df7: avcodec/hevcdec: fix non-ref frame judgement [19:05:48 CET] <j-b> 'morning [19:14:19 CET] <durandal_1707> do you type in every channel that? [19:33:00 CET] <atomnuker> its 18:00, its not even morning [19:33:28 CET] <durandal_1707> he sleep too much [19:33:37 CET] <atomnuker> worse, imagine if it was morning, then you'd be reminded its morning [19:36:05 CET] <BradleyS> good morning :) [19:36:33 CET] <cone-517> ffmpeg 03Mark Wu 07release/4.0:263b3ad40749: avcodec/hevcdec: fix non-ref frame judgement [19:36:35 CET] <cone-517> ffmpeg 03Mark Wu 07release/4.1:11dff170ef79: avcodec/hevcdec: fix non-ref frame judgement [19:47:22 CET] <durandal_1707> michaelni: when you will resign as leader for real? [19:54:24 CET] <jamrial> durandal_1707: do something else if you're bored instead of stirring shit, please [19:55:42 CET] <kierank> durandal_1707: why you troll michael [20:18:06 CET] <durandal_1707> kierank: because he is still leader [20:18:18 CET] <kierank> i don't disagree but why raise it now? [20:21:09 CET] <durandal_1707> he is blocking CoC [20:22:22 CET] <kierank> not the only one [20:22:32 CET] <kierank> you got to follow vote process or whatever "rules" exist [20:24:01 CET] <durandal_1707> kierank: no, we must use force now! [20:24:15 CET] <kierank> should have said that at vdd [20:26:28 CET] <BBB> yeah, we had this discussion at VDD [20:26:33 CET] <BBB> I told all you guys to be there [20:26:37 CET] <BBB> you werent there [20:26:38 CET] <BBB> sorry [20:26:50 CET] <BBB> I believe thilo said he would move it forward [20:27:07 CET] <durandal_1707> jamrial: i got impression, that you are not very fond of me? [20:27:34 CET] <BBB> I think jamrial wishes you could be more positive [20:27:40 CET] <BBB> but I wont put words in his mouth [20:27:51 CET] <jamrial> durandal_1707: i am fond of you, but you really seem to like to troll people when things are quiet [20:41:27 CET] <durandal_1707> kierank: ok, i gonna do Carl statue then [20:43:44 CET] <kierank> durandal_1707: you can buy bricks from elon musk boring company [20:51:33 CET] <durandal_1707> kierank: i thought to make it from space frozen trash [22:16:42 CET] <cone-517> ffmpeg 03Paul B Mahol 07master:0bc2326b90a7: avfilter/vf_lut3d: add cosine 1D interpolation [22:16:43 CET] <cone-517> ffmpeg 03Paul B Mahol 07master:c47b48c0a629: avfilter/vf_lut3d: add spline 1D interpolation [22:22:46 CET] <durandal_1707> 9 more commits [00:00:00 CET] --- Sun Nov 11 2018

1 0

ffmpeg.log.20181110
by burek 11 Nov '18

11 Nov '18

[04:44:30 CET] <`St0ner> how do i report the bug i mentioned above regarding converting OPUS audio inside a WEBM file directly to ALAC? it does not provide an MD5 match with the original audio and results in a file that is triple the size of the equivalent WAV, so i have to convert to WAV first, and then convert WAV to ALAC [04:46:02 CET] <Pocari> I'm still not sure that's because of some bug in ffmpeg [04:46:34 CET] <Pocari> when you tried to play that malformed ALAC, did it work= [04:46:36 CET] <Pocari> ?* [05:12:12 CET] <`St0ner> will remake and check [05:14:10 CET] <Pocari> k [11:52:00 CET] <roger21> hi there [11:52:35 CET] <roger21> is it possible to convert idx/sub files into str? [11:55:17 CET] <roger21> i tried "ffmpeg -i a.idx -c:s srt a.srt" doesn't work [12:21:14 CET] <ariyasu> you need to ocr them roger21 [15:39:09 CET] <hwdyki> i'm building 4.1 from source with the following opts: --target-os=linux --enable-gpl --enable-version3 --enable-nonfree --enable-shared --disable-w32threads --enable-bzlib --enable-libfontconfig --enable-libfreetype --enable-libmp3lame --enable-libspeex --enable-libtheora --enable-libvorbis --enable-openssl --enable-libx264 --enable-zlib --disable-x86asm --disable-random [15:39:43 CET] <durandal_1707> --disable-x86asm --> VERY BAD IDEA [15:40:23 CET] <furq> i don't know what disable-w32threads or disable-random are doing there [15:40:36 CET] <hwdyki> and i'm getting Unknown input format: 'concat' when running ffmpeg -f concat -i chunks -c copy FILE [15:41:00 CET] <hwdyki> do i need any ext libs for concat? [15:41:01 CET] <furq> well yeah disable-random randomly disables components [15:41:28 CET] <hwdyki> hmm. [15:43:10 CET] <durandal_1707> hwdyki: from where you got source? [15:45:11 CET] <furq> https://www.johnvansickle.com/ffmpeg/ [15:45:16 CET] <furq> you might just want to use the builds from here [15:45:38 CET] <hwdyki> ffmpeg.org [15:47:49 CET] <durandal_1707> hwdyki: than make sure you use right ffmpeg version and not some prahistoric one in path [15:48:11 CET] <furq> well also don't use an option that randomly disables stuff [15:48:41 CET] <hwdyki> 4.1 is prahistoric? [15:49:33 CET] <relaxed> you could have an older version in your $PATH [15:49:51 CET] <hwdyki> nope. [15:51:47 CET] <durandal_1707> pastebin full uncut ffmpeg console output [15:52:19 CET] <furq> uh [15:55:51 CET] <hwdyki> took out --disable-random, still the same. [15:57:12 CET] <relaxed> what does this return? ffmpeg -formats 2>&1|grep concat [15:58:22 CET] <hwdyki> nvm fixed it. [16:11:35 CET] <dariken> nice, 4.1, no builds yet. av1 encoding on acceptable speeds now? [16:11:56 CET] <JEEB> that's up to libaom or rav1e [16:12:14 CET] <JEEB> although I don't think rav1e has a c api yet [23:01:41 CET] <zinger> I'm trying to devoces some ALAC files to WAV. The ALAC files are 24-bit but FFmpeg makes the into 16-bit WAV files. How do I tell FFmpeg not to downgrade the files from 24-bit to 16-bit? [23:01:53 CET] <zinger> devoces->decode [23:05:28 CET] <Pocari_> zinger, try ffmpeg -i in.m4a -af aformat=s24:44100 out.wav [23:05:46 CET] <relaxed> zinger: try -c:a pcm_s24le -f wav output.wav [23:06:33 CET] <Pocari_> zinger, that's assuming your sample rate is 44100 Hz, if not change it to whatever it is [23:07:56 CET] <zinger> isn't there a setting that just tells FFmpeg to use the same settings as the source file? [23:09:34 CET] <relaxed> zinger: the default codec for wav is pcm_s16le, changing it to -c:a pcm_s24le should keep it lossless [23:10:20 CET] <Pocari_> does that also copy over the sampling rate (not just the word length) losslessly? [23:11:58 CET] <relaxed> encode both to pcm and hash them to see if it's still lossless [23:12:10 CET] <Pocari_> yeah, I was also going to suggest md5 too [23:13:40 CET] <zinger> for some reason FFmpeg kept the original 88,2 kHz when it downgraded the ALAC from 24-bit to 16-bit [23:14:28 CET] <Pocari_> ok, now md5 hash both the ALAC and the WAV to see if it's genuinely lossless [23:14:28 CET] <relaxed> he tries to maintain as much as possible [23:24:53 CET] <zinger> thanks for the help guys! :-) [23:32:50 CET] <rom1v> hi [23:33:37 CET] <rom1v> in doc/APIchanges: "2016-04-11 - 6f69f7a / 9200514 - lavf 57.33.100 / 57.5.0 - avformat.h " [23:33:59 CET] <rom1v> how I am supposed to test it properly using LIBAVCODEC_VERSION_INT, since there are 2 versions? [23:47:23 CET] <rom1v> maybe #if (LIBAVFORMAT_VERSION_MICRO == 100 && LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(57, 33, 100)) || LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(57, 5, 0) [23:47:49 CET] <rom1v> missing: && LIBAVFORMAT_VERSION_MICRO == 100 [23:47:52 CET] <rom1v> == [23:47:54 CET] <rom1v> 0 [00:00:00 CET] --- Sun Nov 11 2018

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ffmpeg-devel.log.20181109
by burek 10 Nov '18

10 Nov '18

[11:41:42 CET] <cone-790> ffmpeg 03Paul B Mahol 07master:16d91b7e99cd: avfilter/af_afir: kill useless code which increases latency for no reason [13:43:36 CET] <atomnuker> oh hey my server's time which my client is on and my system time match again [13:45:44 CET] <atomnuker> fuck dst, it shouldn't have existed at all, or should have varied with latitude, sunlight at 03:00 is torture [14:02:48 CET] <iive> the whole world should move to UTC [14:10:25 CET] <atomnuker> yep [17:05:00 CET] <philipl> BtbN: the glitching is baffling [18:53:16 CET] <cone-790> ffmpeg 03Paul B Mahol 07master:29fdaaa80d7d: avfilter/af_afir: fix hang because video is first requested [19:35:25 CET] <JohnsonAskot> When using the "gdigrab" format, is it possible to set a specific codec programmatically? I've tried using av_set_dict "vcodec" when passing the options to avformat_open_input, but that doesn't seem to work. [19:41:56 CET] <kepstin> JohnsonAskot: this sounds like more of a user problem, not ffmpeg development. [19:42:14 CET] <kepstin> JohnsonAskot: anyways, gdigrab gets bitmap images from windows apis. whatever you do with them after that is up to you [19:42:38 CET] <kepstin> JohnsonAskot: you'd probably want to decode those, use swscale to convert to a format, then encode with a codec of your choice. [19:46:52 CET] <JohnsonAskot> My bad. [20:07:22 CET] <durandal_1707> i have no ideas what to do next, so... [20:10:52 CET] <atomnuker> new lavfi api [20:17:07 CET] <kierank> durandal_1707: build statue of carl [20:17:11 CET] <kierank> like in india they do [20:26:33 CET] <durandal_1707> something related with audio... [20:29:52 CET] <atomnuker> write a silk encoder? [20:30:51 CET] <durandal_1707> why? [20:32:34 CET] <atomnuker> for fun if you're into slow linear filters [20:36:19 CET] <durandal_1707> better not [20:38:47 CET] <atomnuker> yeah, better not, if you know simd just doing completely unsimdable code is enough to get on your nerves [21:56:05 CET] <cone-790> ffmpeg 03Martin Vignali 07master:75625c555cbe: avfilter/af_headphone : fix mem leak [00:00:00 CET] --- Sat Nov 10 2018

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ffmpeg.log.20181109
by burek 10 Nov '18

10 Nov '18

[00:08:35 CET] <fella> ffmpeg -i MOV.SUF1 -i SND.SUF2 -map 0:0 -map 0:1 -map 1:0 -filter_complex "[0]volume=0.5[some_name];[some_name]amix" OUTPUT.SUF1 [00:31:17 CET] <petecouture> When compiling ffmpeg from source with rtmpdump, do you need the full rtmpdump library or can you just use the included librtmp ver 2.4? [00:33:42 CET] <JEEB> I think librtmp is what FFmpeg utilizes, but check yourself. for the record, there's an internal rtmp implementation as well in libavformat for RTMP(E) [00:37:52 CET] <petecouture> @JEEB is that why the latest compliation build no longer include rtmp dump in them? [00:38:14 CET] <JEEB> I have no idea, there are no official compiles [00:38:25 CET] <JEEB> FFmpeg only provides source code of FFmpeg [00:38:47 CET] <JEEB> I think there are some 3rd party builds linked on the site, but those are 100% under the control of those providing those [00:39:04 CET] <petecouture> I spent the night learning rewriting the librtmp build so it would work with the ffmpeg command line compile. [00:39:16 CET] <furq> i assume librtmp still does something that the internal rtmp stuff doesn't [00:39:18 CET] <JEEB> also the internal support for RTMP(E) is a rather old thing. [00:39:22 CET] <furq> otherwise someone ought to have got rid of it by now [00:39:28 CET] <furq> but nobody has ever been able to tell me what that is [00:39:33 CET] <petecouture> It used to be included but I wondered why it was removed in the latest compliation script ffmpeg.org provides [00:39:46 CET] <JEEB> I don't think we provide compilation scripts? [00:39:54 CET] <JEEB> other than possibly something on the community wiki [00:40:13 CET] <JEEB> furq: mostly what I've seen is that the protocols' AVOptions don't match [00:40:21 CET] <petecouture> Let me pull up the link [00:40:36 CET] <JEEB> so people cannot just use the same command line with ffmpeg.c, for example [00:40:42 CET] <petecouture> https://trac.ffmpeg.org/wiki/CompilationGuide/Centos [00:40:44 CET] <furq> that's a pretty lame reason [00:40:53 CET] <JEEB> yes, that is 100% community based [00:40:55 CET] <petecouture> That guide used to include a librtmp build I believe [00:40:57 CET] <furq> there have been at least two major versions since then [00:41:36 CET] <JEEB> the only other thing is that there's those DMCA'd half-v10 supporting code dumps [00:41:43 CET] <JEEB> but I am not sure how many actually use that [00:42:01 CET] <petecouture> :-/ [00:42:03 CET] <furq> well that's an actual feature so that's ok [00:42:17 CET] <JEEB> it's not in an official rtmpdump release or the repo [00:42:25 CET] <furq> nice [00:42:28 CET] <JEEB> that's what I mean with "how many actually use that" [00:42:44 CET] <JEEB> since most people who build rtmpdump don't generally go searching for some shady repos [00:43:09 CET] <petecouture> I think youll find a bit of people new to ffmpeg that might try [00:43:49 CET] <petecouture> Are there shady repos of it? [00:43:52 CET] <furq> yeah configure could make it clearer that rtmp is builtin anyway [00:43:54 CET] <JEEB> usually no, those who go for that stuff tend to know more or less either what they need or that they need some specific less-official version of rtmpdump [00:44:21 CET] <JEEB> the nr1 reason for wanting to build with librtmp that I've seen is that people know the AVOptions [00:44:24 CET] <JEEB> and guides are made around them [00:44:34 CET] <JEEB> compared to the internal RTMP thing [00:44:48 CET] <furq> based on this channel, the number one reason seems to be that people didn't know about internal rtmp [00:44:52 CET] <petecouture> So building with RTMP is more powerful then using the built in library? [00:44:57 CET] <petecouture> lol [00:45:31 CET] <JEEB> furq: ok, yes. that's the nr1 thing, and then those who try the "native" RTMP support attempt to use the same option names and get stuck or something [00:45:38 CET] <petecouture> I was going to fork the rtmpdump repo and with the changes I made to the build script so it can be compiled in [00:45:45 CET] <furq> petecouture: if you don't know that you need librtmp then you probably don't [00:45:54 CET] <furq> but yeah you might need to rewrite your command [00:46:00 CET] <petecouture> I know enough to be dangerous [00:46:30 CET] <petecouture> Im used to building with it for the raspi arm [00:47:19 CET] <petecouture> Can I pull up the difference between using the librtmp and the internal libraries? [00:47:24 CET] <petecouture> Just to do a compair [00:47:58 CET] <furq> https://ffmpeg.org/ffmpeg-protocols.html#rtmp [00:48:00 CET] <furq> that's the internal one [00:48:13 CET] <furq> i don't think the librtmp stuff is documented on site [00:48:35 CET] <furq> oh nvm it's just not sorted properly [00:48:38 CET] <furq> https://ffmpeg.org/ffmpeg-protocols.html#librtmp-rtmp_002c-rtmpe_002c-rtmps… [00:49:27 CET] <petecouture> @furq thank you! [01:30:43 CET] <orbea> Hi, I am experiencing a crash with ffmpeg while running a single alephone scenario (Game), any ideas if this is a ffmpeg or alephone bug? Or maybe both? https://pastebin.com/YZ1qSnCC I made an alephone issue for now with more details - https://github.com/Aleph-One-Marathon/alephone/issues/121 [01:31:10 CET] <orbea> this is with the ffmpeg master (As of today) [01:41:43 CET] <MarioMey> Hi, there. I have a movie with japanese and english audio streams. I want to add a spanish one, from another file. But I need to recode the audio stream, because by copying it, it doesn't work. Command to copy all streams is: [01:42:23 CET] <MarioMey> ffmpeg -i movie_jap_eng.mkv -i movie_esp.avi -map 0 -map 1:1 -codec copy movie.mkv [01:42:52 CET] <MarioMey> (taken from https://stackoverflow.com/a/44136072/3868690 ) [01:43:12 CET] <MarioMey> How should I encode only the spanish audio stream? [01:46:51 CET] <dreamseeker> can you show us what error you get when you try to copy the Spanish audio stream? [01:47:03 CET] <dreamseeker> like, the full command and output, in a pastebin [01:59:13 CET] <furq> MarioMey: -c:a:0 copy -c:a:1 copy -c:a:2 aac [01:59:15 CET] <furq> or whatever codec you need [02:50:25 CET] <MarioMey> dreamseeker: by encoding, it shows more or less 16 errors like this: [02:50:26 CET] <MarioMey> [matroska @ 0x56026f190960] Starting new cluster due to timestampate=9804.8kbits/s speed=26.4x [02:50:46 CET] <MarioMey> VLC open the file, but audio is unsynch... and shows some other errors. Like: [02:50:54 CET] <MarioMey> [00005623dca7acb0] vlcpulse audio output error: digital pass-through stream connection failure: No soportado [02:51:03 CET] <MarioMey> [00005623dca7acb0] main audio output error: module not functional [02:51:23 CET] <MarioMey> [00007f9200ce04a0] main decoder error: failed to create audio output [02:51:46 CET] <MarioMey> Now, I'm encoding again with what furq said. [02:52:15 CET] <MarioMey> I didn't know that "-c:a:0" way to do it. [08:44:14 CET] <LigH> Hi. [08:45:36 CET] <LigH> Is it possible to let ffmpeg cut n seconds from the end of a clip, without first discovering its length and using it explicitly as parameter? (provided the length can be determined by ffmpeg from header data) [09:01:32 CET] <LigH> I guess --sseof is the one. [09:02:31 CET] <skident> Hi guys, does anybody know is it ok to call function avfilter_register_all (void) from different threads and more than once? [09:03:47 CET] <LigH> Hi ... it's quiet here. America asleep? [09:11:37 CET] <LigH> Bye. [12:50:13 CET] <skident> Let's ask again :) [12:50:14 CET] <skident> does anybody know is it ok to call function avfilter_register_all (void) from different threads and more than once? [12:51:27 CET] <durandal_1707> yes, and note that such function have been deprecated in latest version [12:51:30 CET] <skident> I did some tests an everything work fine, but maybe somebody knows something about it for sure? [12:54:47 CET] <atomnuker> I think they were safe to call before they were deprecated as well given every action was locked [16:17:09 CET] <Pinchiukas> Can I somehow suppress the printing of library information in ffprobe? I can't find an option in the manpage. [16:18:30 CET] <TheAMM> -hide_banner [16:19:48 CET] <TheAMM> -loglevel panic (or such) if you want to hide the rest [16:31:31 CET] <Pinchiukas> -hide_banner doesn't hide it. [16:31:36 CET] <Pinchiukas> It only hides the banner. :) [18:49:27 CET] <ingodid> anyone have a good one-liner for extracting a volume table from an audio file? [18:52:09 CET] <durandal_1707> ingodid: volume what? [18:53:15 CET] <ingodid> a CSV (or STDOUT) table of frames and volumes [18:53:42 CET] <ingodid> i've found something similar for video here "https://stackoverflow.com/questions/38056970/ffmpeg-txt-from-audio-levels" [18:54:04 CET] <ingodid> just starting to wrap my head around the CLI switches [18:55:40 CET] <ChocolateArmpits> ingodid, do you want the volume value for every sample? [18:56:00 CET] <ingodid> wait -- that does seem to work with a WAV [18:56:03 CET] <ingodid> indeed [20:29:53 CET] <ingodid> now onto FFT fun :) [21:19:17 CET] <`St0ner> i'm trying to extract OPUS audio from a YouTube WEBM file losslessly into a format playable by my iPod touch (so either ALAC or WAV). a WAV file is 42.6MB while the ALAC M4A file is 120MB. is it normal for ALAC to be triple the file size of WAV? i thought ALAC is supposed to use lossless compression [21:21:22 CET] <Pocari> `St0ner, how did you go from the Opus embedded in the YouTube video to WAV and ALAC? [21:22:21 CET] <Pocari> what you're saying is not possible... at most, the ALAC file's audio stream will be as big as the uncompressed WAV, only being the same in the case of uniform white noise [21:22:30 CET] <Pocari> so maybe there's something going wrong during the conversion [21:22:52 CET] <`St0ner> for WAV: ffmpeg.exe -i "<SourceFileName>" -acodec pcm_s16le -ar 48000 -ac 2 "<OutputPath><OutputFileName>.wav" [21:23:11 CET] <`St0ner> for ALAC: ffmpeg.exe -i "<SourceFileName>" -acodec alac "<OutputPath><OutputFileName>.m4a" [21:23:44 CET] <`St0ner> ffmpeg build 4.0.2 win64 static dll [21:25:08 CET] <Pocari> `St0ner, check if both WAV and the ALAC, as well as the original Opus, have the same audio hashsum: ffmpeg -i your_file.foo -vn -f md5 - [21:27:13 CET] <Pocari> so, since you're seemingly on Windows, ffmpeg.exe -i your_file.foo -vn -f md5 - [21:33:52 CET] <`St0ner> the MD5= line is different [21:34:03 CET] <Pocari> for all three?? [21:34:41 CET] <`St0ner> sorry didnt run the original. the original and the wav match, the ALAC doesnt [21:35:14 CET] <Pocari> right, that figures... so it's the ALAC indeed that has an issue, as the overbloated filesize indicated [21:36:38 CET] <`St0ner> ok, and is this a problem with the ffmpeg build or should i be feeding non-default parameters into the "acodec alac" section? [21:37:10 CET] <Pocari> `St0ner, try using the WAV then to go to the ALAC; ffmpeg.exe -i track.wav -acodec alac track.m4a [21:39:16 CET] <`St0ner> 31.5MB when i do it that way, which is lower than the WAV file [21:39:23 CET] <`St0ner> so that seemed to have worked [21:39:32 CET] <Pocari> good sign, now to make sure, md5 check that one too [21:40:22 CET] <`St0ner> yes, md5 matches [21:40:42 CET] <Pocari> excellent, so that's a genuine lossless copy... not too sure why the Opus one failed, tbh [21:41:23 CET] <`St0ner> hmm. how can i avoid doing a 2step process from the youtube files? [22:14:27 CET] <ChocolateArmpits> `St0ner, youtube-dl has some encoding options [22:15:01 CET] <ChocolateArmpits> you can download the audio track along with it [22:15:39 CET] <`St0ner> i ended up using AnotherGUI, converted source to WAV, then used the second pass section to convert WAV to ALAC, then used the postprocessing section to delete the WAV. seems to work fine [22:17:01 CET] <saml> how can I write a filter like mandelbrot? [22:17:01 CET] <Pocari> you could have also written a batch file to process the downloaded Opus file with ffmpeg, I guess [22:17:09 CET] <saml> I want my own visualization [22:35:56 CET] <saml> https://www.ffmpeg.org/ffmpeg-filters.html#silenceremove how can I make it also remove video so that there won't be no a-v sync [22:48:39 CET] <threebar> hey everyone, what's the first version of ffmpeg to support 64bit windows builds? [23:03:29 CET] <JEEB> threebar: pretty sure it goes far enough it doesn't really matter [23:06:02 CET] <JEEB> officially change log gained win64 related stuff @ http://git.videolan.org/?p=ffmpeg.git;a=commit;h=61b1f0f909f6adb5e1e32c8b45… [23:06:23 CET] <JEEB> but it clearly seemed to build post-2008? [23:06:52 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=dcc01c06650b193c11d3021346… [23:07:01 CET] <JEEB> because this is the first commit that mentions win64 [23:08:25 CET] <kepstin> win64 stuff wasn't really usable until vista or win7 time period, so around 2008 is a place in the timeline that makes sense. [23:12:04 CET] <threebar> so, what would be the earliest usable/compilable version? [23:13:06 CET] <JEEB> decide by yourself whatever hits your criteria http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=win64 [23:13:17 CET] <JEEB> clearly it was compilable (?) in 2008 already [23:14:04 CET] <JEEB> I know around 2010 people started to want to use it more [23:14:14 CET] <JEEB> which is when the change log entry appeared [23:16:04 CET] <kepstin> win 7 came out in 2009, so that would make sense as being when people started actually caring - upgrades from 32bit xp to 64bit win7. (vista was kinda unpopular) [23:17:13 CET] <kepstin> if you're just going be release tarballs, i wouldn't be surprised if the 0.5 (oldest version still listed on ffmpeg.org/olddownload.html ) would compile, if you can get the right compiler versions. [23:21:43 CET] <JEEB> a snapshot of dec, 2010 by archive.org of fate shows a gcc 4.4.4 x86_64 mingw32 test :P [23:21:54 CET] <JEEB> https://web.archive.org/web/20101221074132/http://fate.ffmpeg.org:80/ [23:22:31 CET] <JEEB> which matches with teh win64 change log part [23:22:45 CET] <JEEB> 727/737 tests pass [23:23:07 CET] <durandal_1707> WOAH [23:26:14 CET] <threebar> nice [23:26:15 CET] <threebar> thank you! [23:27:05 CET] <JEEB> you'd have to grep the IRC logs and the mailing list regarding whenever someone was able to test building for win64 [23:28:09 CET] <threebar> how do i match this to an ffmpeg version number> [23:28:10 CET] <threebar> ? [23:29:14 CET] <JEEB> https://github.com/FFmpeg/FFmpeg/commit/61b1f0f909f6adb5e1e32c8b45a692f67a5… [23:29:18 CET] <JEEB> that's the change log entry [23:29:26 CET] <JEEB> only linking on github because it shows a list of branches [23:29:31 CET] <JEEB> in which the commit is contained [23:30:05 CET] <JEEB> 0.7.1 by the looks of it? [23:30:14 CET] <JEEB> which was released like half a year later [23:30:26 CET] <JEEB> unless those releases are not in the git repo [23:41:17 CET] <threebar> hmm [23:41:44 CET] <threebar> 0.7.1's ./configure says "--target-os=win64" is not a valid option for target-os [23:42:05 CET] <JEEB> not surprising [23:42:50 CET] <JEEB> whatever was used back then with all the configure parameters :P [23:43:02 CET] <JEEB> I don't even know why you're attempting to build a version from 2011 in 2018 [23:43:20 CET] <threebar> getting old code to run [23:44:11 CET] <JEEB> then you should have much better ideas regarding how old something you need is [23:44:14 CET] <JEEB> like SONAMEs [23:47:07 CET] <JEEB> just because it's highly unlikely what I just explained to be what you need :P [00:00:00 CET] --- Sat Nov 10 2018

1 0

ffmpeg-devel.log.20181108
by burek 09 Nov '18

09 Nov '18

[01:33:10 CET] <atomnuker> BBB: so you accepted the ridiculous pr to not adapt but didn't even check its coding style [01:33:44 CET] <atomnuker> honestly, predicted branch or not its utterly ridiculous in case of cdf size one, as in the bool decode function [01:40:38 CET] <BBB> ? [02:49:27 CET] <JC_Yang> questions: should the rtsp client handle/workaround buggy rtsp server implementations? if there're known buggy server implementations, should we add workaround in the client codes? or we should report the bugs to the devices vendor and urge them to fix their own shits? how about if the latter path does not work? [03:39:27 CET] <Compn> JC_Yang : yes [03:39:38 CET] <Compn> as long as you document the buggy ness in the code with some good docs [03:43:35 CET] <Compn> both i mean [03:47:50 CET] <Compn> tlol [04:28:37 CET] <JC_Yang> ok, get it [06:01:39 CET] <Lord_Nightmare> Compn: I saw you put a notice about asking you about codecs on the https://wiki.multimedia.cx/index.php/TwoCC page [06:01:54 CET] <Lord_Nightmare> The "Yamaha ADPCM" codec is not listed on that page [06:02:12 CET] <Lord_Nightmare> but it can be found at https://web.archive.org/web/20160323215457/http://smaf-yamaha.com/jp/tools/… [06:04:45 CET] <Lord_Nightmare> its a plain old Jayant ADPCM codec, almost identical to jayant's 1973 paper at https://ia902601.us.archive.org/13/items/bstj52-7-1105/bstj52-7-1105.pdf and his 1974 paper at https://ia902600.us.archive.org/31/items/bstj52-7-1119/bstj52-7-1119.pdf (see page 23) [06:04:57 CET] <Lord_Nightmare> its also used on a bunch of yamaha ADPCM playback chips [06:05:15 CET] <Lord_Nightmare> like the ym2608 and ym2610 [06:06:14 CET] <Compn> I mean you edit the wiki , you can add it :D [06:06:25 CET] <Compn> do we already have yamaha decoder? [06:08:16 CET] <Lord_Nightmare> not sure, i think ffmpeg does have one. if not, how to make one is pretty easy [06:08:38 CET] <Lord_Nightmare> it uses the same coefficients as jayant's 1974 paper does, just multiplied by 256 (iirc) [06:09:27 CET] <Lord_Nightmare> i.e. 0.8 0.8 0.8 0.8 1.2 1.6 2.0 2.4 [06:09:47 CET] <Lord_Nightmare> but twice, since the sign bit doesn't affect the magnitude of the step size multiplier [06:09:57 CET] <Lord_Nightmare> for each 4 bit sample [06:10:21 CET] <Lord_Nightmare> (MSADPCM is weird and has the table laid out differently, plus adds the value '3.0' for the -8 case) [06:10:52 CET] <Lord_Nightmare> (and BSMT2000 adpcm is sort of like MSADPCM i think except they don't add the 3.0 value?) [06:11:14 CET] <Lord_Nightmare> (and they don't have the 2-previous-samples IIR filter either, which MSADPCM has) [06:12:42 CET] <Lord_Nightmare> i'm still trying to figure out how the 'ADPCM2' mode of the oki MSM6295 works [06:14:10 CET] <Lord_Nightmare> afaik dialogic oki adpcm is 100% identical to the MSM5205 except differences with the preset value for the first sample(?) [06:14:27 CET] <Lord_Nightmare> i'm not sure the difference between 0x0010 OKI ADPCM and 0x0017 Dialogic OKI ADPCM [06:14:47 CET] <Lord_Nightmare> oki adpcm is a subset if IMA ADPCM afaik [06:14:51 CET] <Lord_Nightmare> *of [06:15:02 CET] <Compn> yeah i havent a clue [06:15:06 CET] <Compn> i just collect the twocc :D [06:15:18 CET] <Compn> try to find acm files when people ask [06:15:40 CET] <Lord_Nightmare> I was trying to hack together a codec for 0x0005 IBM CVSD IBM Corporation which was later bought by Digispeech [06:15:47 CET] <Compn> and yes ffmpeg has adpcm_yamaha decoder [06:16:24 CET] <Lord_Nightmare> i don't think there was ever a 'real' acm codec for IBM CVSD (4-bit cvsd), i think the real codec was just a driver that passed the speech data along to an isa card and the PS/2 speech adaptor [06:16:44 CET] <Compn> btw yamaha is on there [06:16:44 CET] <Compn> 0x0020 Yamaha ADPCM [06:16:49 CET] <Compn> on the twocc page [06:17:19 CET] <Lord_Nightmare> yes, and theres a Yamaha_ADPCM page as well [06:17:31 CET] <Lord_Nightmare> i didn't even KNOW about the twoCC page until i checked recent edits [06:18:01 CET] <Compn> yeah i wanted to put it in the main nav box [06:18:15 CET] <Compn> but i think mike gave me the runaround when i asked him howto get it there [06:18:58 CET] <Lord_Nightmare> afaik the PS/2 Speech adapter isa card has an MC3418 4-bit cvsd codec chip on it, and an adjustable clock generator for it for 8 fixed samplerates http://arcarc.xmission.com/Tech/Datasheets/MC3418.pdf [06:19:15 CET] <Compn> Lord_Nightmare : theres also this page I maintain https://wiki.multimedia.cx/index.php/Category_talk:Video_FourCCs [06:19:42 CET] <Compn> both used to get a lot of hits from google. but i cant track the hits anymore since the wiki upgrade took out the "popular pages" and hit counts out :( [06:19:52 CET] <Lord_Nightmare> the CVSD typically used on arcade games and pinballs though is the MC3417 3-bit cvsd codec chip, and its harris clones the HC55516, HC55532 and HC55564, and the decode-only HC55536 [06:20:08 CET] <Compn> i just sold my pinball games :( [06:20:09 CET] <Compn> ehe [06:20:39 CET] <Lord_Nightmare> motorola made a later chip called MC34115 which i believe is an updated MC3417 with multiple filters and a bandpass to select which one is active, like the HC55564 has [06:20:59 CET] <Lord_Nightmare> the hc55516 filters are tuned for 16khz, and hc55532 is tuned for 32khz [06:21:19 CET] <Lord_Nightmare> hc55564 uses some cute internal circuit with an analog bandpass to select one of two sets of filters, or so i believe [06:21:28 CET] <Lord_Nightmare> depending on how fast the clock in is [06:21:44 CET] <Lord_Nightmare> sorry for info dump [06:22:40 CET] <Lord_Nightmare> jeri ellisworth made an fpga reimplementation of the 3-bit cvsd chip to fix one of her pinballs, but i believe she forgot one of the analog filters when doing so. it still seems to sound ok, though, surprisingly... [06:23:33 CET] <Lord_Nightmare> https://www.youtube.com/watch?v=2FRGwuxFDE4 [06:23:42 CET] <Compn> have the pinmame people done some of these codecs ? [06:24:05 CET] <Lord_Nightmare> MAME has an hc55536 implementation used on exidy's mouse trap arcade game [06:24:19 CET] <Lord_Nightmare> but again i'm not sure how accurate it is [06:24:34 CET] <Lord_Nightmare> its also used on some williams games i believe [06:25:19 CET] <Compn> hah i just sold my getaway [06:25:32 CET] <Compn> williams game [06:25:35 CET] <Compn> and my taxi [06:25:43 CET] <Compn> nother williams [06:26:24 CET] <Lord_Nightmare> a real cvsd chip has two analog filters: one controls the rise and fall of the slope magnitude value if it received 3 or 4 bits of the same value for the most recent 3 or 4 bits [06:26:58 CET] <Lord_Nightmare> and another filters this magnitude value as a lowpass when it is applied to the comparator op-amp vs the input [06:27:21 CET] <Compn> dont think i'd wire in an fpga though [06:27:21 CET] <Lord_Nightmare> jeri emulated the first of these as if it is perfectly symmetrical gain and loss, and didnt' emulate the second one at all [06:27:35 CET] <Compn> easier to fix the boards , get the roms on new chips and put it back together [06:27:55 CET] <Lord_Nightmare> yeah, i don't think obtaining a NOS or used but working CVSD chip is as hard as jeri made it out to be [06:27:58 CET] <Compn> unless the entire sound board was toast. then... [06:28:01 CET] <Compn> ah [06:28:12 CET] <Lord_Nightmare> I'm going to email her and see if she still has the blown out chip, since i'd love to get a micrograph done of it [06:28:26 CET] <Compn> the chip or the board ? [06:28:34 CET] <Compn> you can just straight out buy new chips [06:29:25 CET] <Compn> probably you mean board :D [06:30:16 CET] <Compn> and when you cannot get boards anymore, just toss in the emulator on an rpi [06:30:27 CET] <Lord_Nightmare> nah she had a working board with just that one chip blown [06:30:51 CET] <Lord_Nightmare> sound worked except the speech was just noise [06:30:56 CET] <Compn> ah [06:31:05 CET] <Compn> then yeah its as easy as getting a new chip [06:33:19 CET] <Compn> or re-programming the old one if its just corrupt [06:33:40 CET] <Lord_Nightmare> you can't reprogram a CVSD chip, its a fixed function mixed-analog-digital ASIC of sorts [06:33:48 CET] <Compn> oh i'm thinking of the sound samples [06:33:49 CET] <Compn> my bad [06:33:52 CET] <Compn> not the chip [06:33:57 CET] <Compn> my brain is off right now [06:34:06 CET] <Lord_Nightmare> in theory you MIGHT be able to replace it with a very fast PIC emulating a CVSD chip [06:34:15 CET] <Lord_Nightmare> and emulating the analog filters digitally [06:35:43 CET] <Compn> sorry [06:35:44 CET] <Compn> ehe [06:37:23 CET] <Compn> proc thats what its called [06:38:00 CET] <Compn> https://www.multimorphic.com/store/circuit-boards/p-roc/ [06:38:44 CET] <Compn> oh that uses the existing boards, i'm completely incorrect about it haha [06:39:13 CET] <Compn> thats pretty neat replacing a sound decoder chip with a fpga though [07:05:43 CET] <JC_Yang> do we internally have any indicator whether AVStream.avg_frame_rate is valid? or any guidelines about when it is usable as a reference? [10:49:52 CET] <BtbN> Scientific build systems sure are fun. [10:50:30 CET] <BtbN> This one expects to be cloned to .../src/github.com/syslabs/singularity. Otherwise it will fail to build. [10:51:00 CET] <BtbN> The only place I learned about that was in a github issue about it. [11:03:14 CET] <nevcairiel> some people just care about working on their code, not making it easy on other people [11:06:23 CET] <BtbN> This is aparently how go is supposed to work. [11:07:01 CET] <BtbN> Even though you don't even see it's Go, cause their self-built mconfig script downloads it in the background [11:07:14 CET] <nevcairiel> thats at least something [13:07:45 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:fcbde5e6a4a5: avfilter/af_afir: remove redundant code [13:07:46 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:8dc5eb43b070: avfilter/af_afir: return early when giving video frame [13:07:47 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:1d9fe1fdf65b: avfilter/af_afir: implement rate option [13:07:48 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:0f0f84071659: avfilter/af_aiir: implement rate option [13:39:05 CET] <cone-132> ffmpeg 03Carl Eugen Hoyos 07master:5ff080fdfa25: lavf/apngdec: Return EOF for incomplete files. [15:21:51 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:09fdfc11b796: avfilter/avf_showspectrum: add cividis colormap [18:00:35 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:70463b702697: avfilter/avf_showspectrum: add terrain color map [18:38:08 CET] <BtbN> philipl, 415 is out [18:44:25 CET] <philipl> BtbN: yes. I'll be trying it out tonight/tomorrow [20:40:28 CET] <durandal_1707> FFmpeg - officially dead [20:43:00 CET] <atomnuker> wouldn't be so dead if companies didn't stop making their own custom codeccs [20:44:56 CET] <durandal_1707> it is dead, no new commits in latest days except mine [20:47:55 CET] <kepstin> is everyone busy on dav1d instead? :) [20:48:13 CET] <atomnuker> yeah [20:48:42 CET] <atomnuker> wish there were some 4k samples to test its speed, I'm getting 220+ fps on 1080p and its so fast its boring [20:49:36 CET] <mmu_man> plop [20:49:37 CET] <mmu_man> anyone seen my metadata patch? [20:50:48 CET] <durandal_1707> this one: [20:50:51 CET] <durandal_1707> [FFmpeg-devel] [PATCH] Optimize libavformat/metadata.c? [20:51:35 CET] <durandal_1707> ah, michael replied to another metadata patch [20:51:56 CET] <mmu_man> ah yes, just saw [20:52:17 CET] <jamrial> atomnuker: elecard has some 4k sampes [20:59:36 CET] <BBB> netflix also, no? [21:00:06 CET] <atomnuker> netflix had chimera, which can curiously fully load my cpu, but a similar resolution and bitrate from youtube can only half load it [21:00:36 CET] <atomnuker> maybe the gop length is different [21:02:09 CET] <BBB> how weird [21:02:14 CET] <BBB> maybe bitrate? [21:02:30 CET] <BBB> with most things simded, bitrate significantly affects cpu load [21:02:35 CET] <BBB> so netflix may be better for your purposes [21:02:41 CET] <BBB> since bitrate/ec-ratio is higher [21:03:14 CET] <BBB> bitrate (= entropy coding-ratio) is higher [21:03:19 CET] <BBB> you know what I mea [21:03:25 CET] <atomnuker> I'm getting the same performance over 300 frames too, 1.4 seconds to decode from both [21:03:36 CET] <atomnuker> its just the load is higher on one than the other [21:03:47 CET] <kurosu__> there also may be some tool that both is more consuming and increase the delay for frame multithreading ? [21:03:51 CET] <atomnuker> yeah, on netflix the ec overhead is higher [21:05:06 CET] <kurosu__> atomnuker, btw, for msac_decode_symbol, pmulhuw exists [21:06:51 CET] <atomnuker> yeah, I know [21:07:12 CET] <atomnuker> the elecard 4k sample kills performance though, barely 44fps or so here [21:10:44 CET] <BBB> oh no, only 44 fps" [21:10:59 CET] <BBB> I used pmulhuw in some of my tests instead of psraw, it did seem to help, yes [21:11:02 CET] <BBB> and prevents stack use [21:11:23 CET] <atomnuker> its the attitude you need to optimize further :) [21:11:33 CET] <BBB> true [21:11:47 CET] <BBB> I think after having spent several months writign the decoder, I want to do encoding work again [21:12:10 CET] <BBB> it does what it needs to do, Im happy to do more work on it but Id also like to sort of sit back and see if people want to use it etc. [21:12:11 CET] <atomnuker> your opening quote mark was converted to some non-ascii one but your closing quote mark is ascii I think [21:12:29 CET] <BBB> "now it's ok"? [21:12:38 CET] <BBB> my irc client is shit [21:13:29 CET] <atomnuker> yeah, its okay now [21:14:02 CET] <atomnuker> could be eviler, it could detect dates and convert them to messed up formats or worse, detect fractions and change the dot to a comma [21:14:38 CET] <atomnuker> and remove spaces/thin spaces for thousands separations and replace them with dots and or commas [21:15:30 CET] <BBB> mine also changed dashes [21:15:38 CET] <BBB> dav1d's CLI doesn't scan for unused arguments [21:15:52 CET] <BBB> so if you copypaste my non-standard dashes on the CLI, it doesn'tknow it's a dash and it ignores the argument [21:15:54 CET] <BBB> remember? :) [21:16:27 CET] <atomnuker> yeah, that took some head scratching to figure out, maybe with a different font they'd be more visible [21:28:48 CET] <atomnuker> elecard's 1080p version is about 115fps here, so its slightly more than 2x slower [21:29:07 CET] <atomnuker> they must be using some bottlenecking bitstream features then [22:10:06 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:698e67bef2ee: avfilter/af_afir: do not leak frame and clear pointer to freed frame [22:10:07 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:36348d77b86a: avfilter/af_afir: add minp/maxp options to control latency and speed [22:19:46 CET] <cone-132> ffmpeg 03Andreas Rheinhardt 07master:8d1cf2d89481: cbs_h264: Fix handling of auxiliary pictures [22:22:20 CET] <cone-132> ffmpeg 03Martin Vignali 07master:105891cd580b: avcodec/prores_aw : change src pix_fmt and alpha description [22:22:21 CET] <cone-132> ffmpeg 03Martin Vignali 07master:4d32c2b4e0df: avcodec/prores_aw : cosmetic indent and add comment to describe frame header value [22:22:22 CET] <cone-132> ffmpeg 03Martin Vignali 07master:ff3b4f91cd56: avcodec/prores_aw : add support for prores 444 with alpha [22:41:31 CET] <cone-132> ffmpeg 03Michael Niedermayer 07master:07bc603757ca: avcodec/qpeg: Optimize full width runs in qpeg_decode_intra() [00:00:00 CET] --- Fri Nov 9 2018

1 0

ffmpeg.log.20181108
by burek 09 Nov '18

09 Nov '18

[11:31:04 CET] <Akien> Hi there! I'm trying to debug a video playback issue in the emulator PPSSPP when built against FFmpeg 3.1+ (upstream uses 3.0.2 for official builds). I have no experience with FFmpeg and I'm not the PPSSPP dev who wrote the video playback code either, but I've done some bisecting and thought maybe some of you know what the current code misses. The [11:31:04 CET] <Akien> issue: https://github.com/hrydgard/ppsspp/issues/11490 [11:31:24 CET] <Akien> TL;DR: Issue triggered by an API change in FFmpeg 3.1 for AVStream: https://github.com/hrydgard/ppsspp/issues/11490#issuecomment-436589320 [11:32:18 CET] <Akien> And this is the code that works well with 3.0.x but produces stuttering in 3.1+: https://github.com/hrydgard/ppsspp/blob/master/Core/HW/MediaEngine.cpp#L245… [11:33:06 CET] <Akien> From what I understand of https://github.com/FFmpeg/FFmpeg/commit/9200514ad8717c63f82101dc394f4378854… I guess the AVStream needs some more configuration now? [11:35:28 CET] <Akien> (This was done in part here, fixing the garbled output, but not the choppiness: https://github.com/hrydgard/ppsspp/commit/7a7c655615a4f607ea2e6bc58dc8ca3e4… ) [11:40:38 CET] <RandomUser8787> hello all. im new to developing with libav libs and im currently having some issues with memory leaks which i dont quite understand. im getting UYVY422 pixel data from a capture device which i copy to a frame. this frame i then run through a filtergraph and then the filtered frame i transfer to the gpu for encoding (nv12). after each frame is encoded i free all frames but im seeing leaks from the filter graph, avcodec_send_frame [11:42:30 CET] <RandomUser8787> i have looked at the ffmpeg filter and vaapi encode code examples but obviously im missing something... [12:58:24 CET] <RandomUser8787> hmm looks like i should have done unref instead of free [13:44:34 CET] <RandomUser8787> or no that didnt really change anything [14:41:10 CET] <zanijwa> Does the precompiled versions of ffmpeg not have https support enabled by default? I'm using youtube-dl and it gave an error 'https protocol not found, recompile FFmpeg with openssl, gnutls or securetransport enabled' [14:41:21 CET] <zanijwa> SO did say to recompile ffmpeg [14:41:59 CET] <zanijwa> So if I do this by myself, should I pass the entire path to my compiled ffmpeg whenever I need to call it? [14:42:34 CET] <zanijwa> Because I prefer automatic updates provided by the Ubuntu package maintainers [14:54:28 CET] <^Neo> good morning friends, is the idea with 4.X now that av_frame_get_best_effort_timestamp / av_frame_get_pkt_duration / av_frame_get_pkt_size is to access the data members directly? [14:56:24 CET] <kepstin> ^Neo: yeah, just access the data members directly. [14:59:14 CET] <ariyasu> zanijwa it does just add it to the whitelist [15:22:10 CET] <santamarta> I am trying to make a long presentation to record in a bluray under linux. It has 21 images and 18 long audios (~1 hr each, but different length each), mostly voice, with some music here and there. The whole audio encoded to .m2ts is ~200M long. The unencoded .jpg images are ~25M. I tried to encode the images to h264, each image is static and most of the audios show only one image, but some of them have two. The result was a 3.7G file. [15:23:07 CET] <santamarta> my question is... There is a way to encode the images in a lower size file that is compatible to dvd or bluray format ? [15:23:25 CET] <durandal_1707> there are some options for still picture [15:23:38 CET] <furq> -tune stillimage [15:23:48 CET] <santamarta> yes, I am using that [15:23:53 CET] <furq> pastebin the full command [15:24:03 CET] <kepstin> the requirements for bluray video specify a fairly short keyframe interval, so the video's gonna be big [15:24:06 CET] <kepstin> no way around that [15:24:09 CET] <furq> how short [15:24:35 CET] <santamarta> it is small: ffmpeg -f concat -safe 0 -i Covers.txt -vf "scale=1920:1080,format=yuv420p" -c:v h264 -tune stillimage portadas2.mp4 [15:25:00 CET] <furq> that's not actually bluray compatible so i'm surprised it turned out that big [15:25:03 CET] <kepstin> I think 1 second gop length [15:25:15 CET] <kepstin> i.e. 30 frames for 30fps [15:26:05 CET] <kepstin> (although many players apparently accept longer values, particularly the ps3 and software players that also do arbitrary h264 high stuff) [15:27:02 CET] <furq> well yeah you need -x264-params bluray-compat=1 [15:27:10 CET] <furq> and maybe also -g 30 but i guess bluray-compat might set that [15:27:13 CET] <furq> or -g 25 rather [15:27:53 CET] <santamarta> thanks, furq [15:27:58 CET] <furq> bear in mind bluray also supports 720p [15:28:06 CET] <furq> so i guess if you care about size then 720p24 is good [15:28:48 CET] <santamarta> I will try... 1080p is important because the images are text, but maybe coud go with more images [15:46:24 CET] <santamarta> could this be solved using a lower framerate ?. Time resolution is not critical at all, a 5 seconds difference causes no problem [15:46:41 CET] <kepstin> I think 24fps is the minimum [15:47:01 CET] <kepstin> but given that the keyframe interval is set based on realtime, that's not likely to make a noticable difference [15:49:30 CET] <kepstin> hmm, it sounds like you can use longer gop when the bitrate is lower. up to 2 seconds if you're under 15mbit/s, which should be true of a video made from mostly still images [15:50:53 CET] <kepstin> if you want the video to fit within a particular size, i'd suggest using a 2-pass bitrate encode with bluray compatible options, and just see if the quality is acceptable [15:51:45 CET] <kepstin> and if you don't actually need blu-ray compatibility, the video can be made a lot smaller :/ [17:01:28 CET] <gh0st3d> Hey everyone... I'm trying to take a video file and create a gif from 1s to 11s that also includes an overlay... I previously got it to include the overlay while creating the gif, but the colors on the overlay were messed up because of the palette thing. For now, I'm making it add the overlay to the video, then creating the palette and the gif [17:01:47 CET] <gh0st3d> However, it definitely was much faster the other way... Is there anyway to generate the palette from 2 files separately? Here's my commands: https://pastebin.com/eRcRY89E [17:02:31 CET] <TheAMM> You'll have to overlay when creating the palette [17:02:50 CET] <TheAMM> You don't have to render the overlay on the video and encode that to a file, you can add the overlay filter before the palettegen [17:03:05 CET] <gh0st3d> Ok, let me try rearranging that and see if I can get it figured out [17:03:05 CET] <TheAMM> The paste isn't loading for me right now so I can't comment on the commands [17:03:43 CET] <TheAMM> but ffmpeg -i in.mkv -vf "overlay=<...>, palettegen" palette.png [17:12:13 CET] <gh0st3d> Oh sweet, got it this time. I thought I tried something similar before but just got a bunch of errors. Is there anyway to compress the gif output with FFMPEG or do you need other software installed to do that? [17:21:27 CET] <kepstin> not sure what you mean by "compress", ffmpeg can generate gif animations directly, but depending on the original content it may be possible to optimize them better manually or with other tools. [19:12:03 CET] <mooniac> is it possible to use two videos of the same length, and make one a smaller inset in one of the corners of the other video? I think that's sometimes called a "mashup"? [19:14:03 CET] <pink_mist> https://www.daniel-mitchell.com/blog/video-in-a-video-ffmpeg/ [19:14:58 CET] <mooniac> woohoo, thanks. [19:15:50 CET] <pink_mist> never tried doing that myself though, so I can't speak to the effectiveness of that blogpost [19:16:14 CET] <mooniac> but it's a good start for me. If it doesn't work, I'll be back here :) [19:22:07 CET] <ChocolateArmpits> mooniac, actually it's called picture in picture "PiP" [19:22:20 CET] <mooniac> thx [19:32:14 CET] <Aerroon> i have a bunch of audio files [19:32:46 CET] <Aerroon> and i'd like to put a static image and stretch it across the entire file to basically make a "video" wher eit plays the audio and shows that one specific image [19:33:04 CET] <Aerroon> what should i do to have the "fastest render time"? [19:33:20 CET] <Aerroon> the normal way of doing it that i figured out i'm basically just rendering out a new video [19:41:29 CET] <pink_mist> no idea, something like: ffmpeg -i pic.png -i audio.mp3 -c:a copy -c:v libx264 -tune stillimage out.mkv perhaps? [19:48:14 CET] <atomnuker> keep in mind that'll make a video [19:48:37 CET] <atomnuker> if you want a single frame you have to encode to a temporary file [19:48:53 CET] <atomnuker> and then use stream copy on that single frame stream along with your audio stream [19:49:28 CET] <atomnuker> some players like vlc won't display it correctly though, but most will, including browsers [19:50:03 CET] <atomnuker> it'll save you a lot of bits though so its worth it [19:50:43 CET] <atomnuker> but if the container supports it you can add the cover art disposition which will make it work with pretty much anything [19:54:27 CET] <Aerroon> i see [19:54:28 CET] <Aerroon> thanks [19:54:45 CET] <Aerroon> the bits bit isn't important for me, i'm more interested in it not taking a long time to encode [19:54:50 CET] <Aerroon> but your solution likely does both [20:37:44 CET] <Aerroon> from tests it seems i'm gonna go with [20:37:47 CET] <Aerroon> ffmpeg -loop 1 -i image.png -i audio.wav -shortest -c:v libx264 -tune stillimage -c:a copy result.mkv [20:38:06 CET] <Aerroon> but this means it'll have to encode it as though it's just a normal video [20:38:20 CET] <Aerroon> any recommendations for a faster video codec? [20:38:38 CET] <Aerroon> ohh, ig uess i could try gpu encoding [20:39:10 CET] <pink_mist> mjpeg? [20:39:56 CET] <Aerroon> that doesn't seem faster [20:42:03 CET] <Aerroon> hmm, it seems the encoding speed isn't even limited by that [20:42:14 CET] <Aerroon> i think it might be the loop 1 argument [20:43:40 CET] <durandal_1707> -loop 1 -- will decode image all over again [20:43:57 CET] <Aerroon> yep, that would be consistent with what i'm getting [20:44:11 CET] <Aerroon> i guess that was the point of the earlier stream copy comment [20:45:06 CET] <mmu_man> plop [20:45:11 CET] <mmu_man> anyone seen my metadata patch? [20:45:34 CET] <durandal_1707> wrong channel for patch talk [20:49:21 CET] <mmu_man> eh [20:59:56 CET] <analogical> I just encoded a sound file to FLAC but the the resulting file become 24-bit although the source file is 16-bit. How do I force FFmpeg to encode to 16-bit FLAC ?? [21:02:34 CET] <dreamseeker> run an ffmpeg -i input_file.flac command and pastebin the output for us [21:02:49 CET] <dreamseeker> er [21:02:57 CET] <dreamseeker> ffmpeg -i input_file.foo I mean [21:03:04 CET] <dreamseeker> whatever extension the input file is [21:09:30 CET] <Aerroon> oof [21:11:07 CET] <Aerroon> speed=1.14e+03x [21:32:10 CET] <Aerroon> >Too many packets buffered for output stream 0:1. [21:32:21 CET] <Aerroon> it's an audio stream copy [21:32:32 CET] <Aerroon> thats 2.5 hours long [21:32:50 CET] <Aerroon> -max_muxing_queue_size 4000 fixes it but i have no idea what that means [21:34:30 CET] <JEEB> Aerroon: ffmpeg.c wasn't for one reason or another capable of initializing the muxer until then [21:35:00 CET] <JEEB> so it makes a queue and sticks the AVPacket there [21:35:38 CET] <JEEB> and the option you set controls the size of that queue [21:35:54 CET] <Aerroon> i see [21:36:21 CET] <JEEB> the better question is why it decides it cannot initialize the muxer (the thing that writes them packets into a container and into a file/stream) [21:49:25 CET] <Aerroon> i don't know either [21:49:43 CET] <Aerroon> because i did demux this audio stream with -c:a copy from another file [21:50:38 CET] <Aerroon> i get the too many packets buffered for output stream 0:1 (which is audio [21:50:42 CET] <Aerroon> from a file i made with [21:50:44 CET] <Aerroon> ffmpeg -stream_loop -1 -i image.mp4 -i audio.m4a -shortest -c:v copy -c:a copy out.mkv [21:51:07 CET] <Aerroon> image.mp4 is a video that consists of one frame [21:51:39 CET] <Aerroon> but the problem appears on the audio stream which demuxed just fine before i put it together with the image [23:04:41 CET] <SneakySnake> Do I really have to do wizardry like this to mix an audio stream with a video while also lowering the original video's volume? https://superuser.com/a/769195 [23:04:55 CET] <SneakySnake> Is there a simpler way? [23:05:11 CET] <SneakySnake> I can't even comprehend what is going on here [23:23:26 CET] <SneakySnake> Okay, I think I figured it out. [00:00:00 CET] --- Fri Nov 9 2018

1 0

ffmpeg-devel.log.20181107
by burek 08 Nov '18

08 Nov '18

[12:00:36 CET] <BtbN> Hm, I wonder if ffmpeg_cuvid.c can just be removed by now. It looks as generic as it gets. [15:41:51 CET] <kwizart> Hello, FYI fsf raised a statement on the fdk-aac license (on Fedora/RH request after a patent disabled fdk-aac package was introduced) [15:41:54 CET] <kwizart> https://www.gnu.org/licenses/license-list.en.html#fdk [15:42:09 CET] <kwizart> in case you have missed it [15:43:03 CET] <kwizart> half related, I've recently found this project https://github.com/yugr/Implib.so [15:43:53 CET] <kwizart> which might allow to dlpen any libraries at runtime (probably once the headers and entries point are added in ffmpeg) [15:44:09 CET] <kwizart> would it be something appropriate for use in ffmpeg ? [15:44:25 CET] <kwizart> (for such cases as fdk-aac decklink or else ?) [15:44:56 CET] <nevcairiel> its questionable this influences the license situation. Some technical mean should not influence how law is interpreted. this is not something ffmpeg s hould consider. if you want to build like that [15:45:18 CET] <nevcairiel> .. you can do that w ithout ffmpeg knowing about it, its supposed to be transparent [15:59:14 CET] <kwizart> yep I think it's just a matter to error as appropriate if end-users request the library but it's not available at runtime [15:59:54 CET] <atomnuker> I wish people would stop trying to find ways to make fdk appear as legal and open source as possible and just use opus [16:00:18 CET] <kwizart> the question is would ffmpeg source tree would be able to have the generated wrapper code or this have to be a downstream use case ? [16:02:00 CET] <gnafu> atomnuker: Hear, hear! [16:03:51 CET] <kwizart> atomnuker, I agree with that too, but the averrage end-user is dumb when it comes to audio codec [16:04:10 CET] <JEEB> also there's the plastic boxen, although the internal AAC encoder is generally speaking OK [16:05:01 CET] <j-b> yes [16:05:09 CET] <j-b> fdk-aac is non-free [16:05:10 CET] <kwizart> atomnuker, and my problem as a distro maintainer is that some scammer repo still advertise ffmpeg built with fdk-aac without fear breaking "good repos" that tries to avoid such licensing issue [16:05:13 CET] <j-b> ffaac is good enough [16:05:19 CET] <j-b> and opus is amazing [16:05:38 CET] <kwizart> j-b, note quite, it's GPL incompatible as stated by fsf, but it might be floss [16:05:47 CET] <j-b> it is not. [16:05:54 CET] <kwizart> (they don't say it's not) [16:06:09 CET] <kwizart> they say it should be avoided when possible, which is good [16:06:32 CET] <j-b> "You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto." [16:07:13 CET] <JEEB> "This is a free software license as far as it goes. It is incompatible with any version of the GNU GPL." <- whatever this means @ the FSF page [16:07:21 CET] <JEEB> "as far as it goes" [16:07:33 CET] <JEEB> that just raises more questions as a wording than anything else [16:07:50 CET] <kwizart> j-b, I don't see that in the current fdk-aac license at all [16:10:06 CET] <kwizart> j-b sorry my bad, it's there, but it's not an issue [16:12:13 CET] <j-b> https://opensource.org/osd [16:15:25 CET] <jamrial> jkqxz: ticket 7538, libva-x11 pkg-config apparently doesn't pull x11 ldflags and cflags [16:16:16 CET] <jamrial> while libva needs to fix it upstream, configure should also make vaapi_x11 depend on xlib, so it may include its ldflags [16:17:41 CET] <kwizart> j-b, maybe ffmpeg/vlc should publish their reading of the fdk-aac license , it would be interesting to compare with the fsf statement wording [16:18:15 CET] <kwizart> but anyway, thx for all advices and sorry for raising that issue again [19:02:40 CET] <cone-228> ffmpeg 03Paul B Mahol 07master:975510758cc7: avfilter/avf_showspectrum: improve magma colors [19:02:41 CET] <cone-228> ffmpeg 03Paul B Mahol 07master:ed58db4237c8: avfilter/avf_showspectrum: add viridis color map [19:02:42 CET] <cone-228> ffmpeg 03Paul B Mahol 07master:0c6d4e7ba4d7: avfilter/avf_showspectrum: add plasma color map [20:57:18 CET] <jkqxz> jamrial: True, but that doesn't appear to be the problem there because it's complaining about libavutil? [20:57:25 CET] <jkqxz> To get libva_x11 you need to have xlib, and xlib gives you those libraries (<http://git.videolan.org/?p=ffmpeg.git;a=blob;f=configure;h=00b5d9795e0daf5c…>). [20:57:29 CET] <jamrial> jkqxz: hwcontext [20:57:52 CET] <jkqxz> Seems to work fine for me. --disable-xlib indeed loses the X flags and libva_x11 too. [20:58:06 CET] <jamrial> nothing is making libavutil depend on (aka, include ldflags from) xlib [20:58:20 CET] <jamrial> or at least nothing obvious [21:05:40 CET] <atomnuker> I think we wanted to get rid of specific vaapi backends eventually and just always used a dri device path [21:06:32 CET] <jkqxz> Getting rid of the X11 backend would be sensible, it's not useful to anyone. [21:07:24 CET] <jkqxz> Ah, I see. VDPAU works because it includes -lX11 explicitly in its own flags, and autodetect pulls that in. [21:07:49 CET] <jkqxz> Whereas VAAPI doesn't have anything making that dependency, and libva-x11 also lacks it. [21:11:30 CET] <jkqxz> Is <https://0x0.st/sIFj.diff> right? It seems to work, but I'm not sure how those configure parts interact exactly. [21:13:00 CET] <jamrial> yeah, that should do it [22:33:45 CET] <atomnuker> BBB: btw gave up on doing adaptation as a separate function, I'll make the whole decoding function in asm [22:34:27 CET] <atomnuker> call overhead was almost 2x the overhead for doing the basic full avx2 adaptation [22:35:43 CET] <atomnuker> with the latest blend avx2 simd decode_symbol_adapt had the most overhead for chimera at 13% the whole decoder [22:38:24 CET] <BBB> yes I saw that too [22:38:25 CET] <BBB> ok [22:39:16 CET] <BBB> this is a good sign btw, now that ec is becoming most runtime [22:39:19 CET] <BBB> it means we did well :) [00:00:00 CET] --- Thu Nov 8 2018

1 0

ffmpeg.log.20181107
by burek 08 Nov '18

08 Nov '18

[00:26:37 CET] <benbro> furq: padding/scaling before format seems to fix it [00:26:39 CET] <benbro> thank you [00:27:02 CET] <benbro> I still don't understand why it sometimes succeeded even with odd width/height [03:36:34 CET] <ZeroWalker> can i trim a video so that the audio starts at the same time as the video (currently it's negative, so audio plays a bit before the video, so i just want to cut out that till the video starts) [03:36:55 CET] <ZeroWalker> so basically, cut audio till the first key frame [03:52:21 CET] <mixfix41> which video editor would you use just ffmpeg? [03:52:31 CET] <mixfix41> maybe use a gui video editor [03:53:06 CET] <frendShot> if he's asking in #ffmpeg, he's probably asking how to use ffmpeg to do that... [03:56:14 CET] <mixfix41> oh ok so just trim the file to the file starts both audio and video with ffmpeg [03:57:51 CET] <mixfix41> yea i do sound a bit brash but i was actually visualizing using kdenlive and all but yeah a quick trim off the beginning should do the work [04:40:26 CET] <mixfix41> ZeroWalker: i think you could use this https://superuser.com/questions/138331/using-ffmpeg-to-cut-up-video for single clips then but later used this https://superuser.com/questions/820747/slicing-video-file-into-several-segm… due to a better file/multiple trims [11:15:16 CET] <mmu_man> plop [11:15:19 CET] <mmu_man> is trac down? [11:25:30 CET] <c_14> works here [11:41:25 CET] <BtbN> It's been insanely slow the last couple days for me [11:42:08 CET] <zerodefect> In what respect? [11:44:36 CET] <zerodefect> Using the C-API and when dealing with sRGB, what do I set the AVFrame's color primaries too (ie. AVColorPrimaries). AVCOL_PRI_BT709 ? [13:03:56 CET] <zerodefect> Hmmm...and if want to convert between bt.709 and bt.601, can I get away with swscale or am I forced to utilize colorspace filter. [13:05:17 CET] <furq> zerodefect: use colorspace or zscale [13:07:02 CET] <zerodefect> Does swscale not make the grade? [13:07:59 CET] <zerodefect> My hesitation with using filters is that the graphs are very static....cannot change input parameters without recreating graph. [13:08:44 CET] <furq> you could also just flag the output as 709 if that's an option [13:09:28 CET] <zerodefect> Not sure what you mean, can you elaborate? [13:10:56 CET] <furq> i mean just not converting colourspace and setting the output primaries etc to 709 [13:11:11 CET] <furq> if that's not an option then you're stuck with using filters afaik [13:12:17 CET] <zerodefect> Ok. I guess I ask this because I note that swscale has a `sws_setColorspaceDetails(..)` function. [13:17:29 CET] <mmu_man> [ogg @ 0x5597871796c0] Error reading attached picture data. 708 != 5120 [13:17:44 CET] <mmu_man> no wonder I can't get a thumbnail inside [13:17:49 CET] <mmu_man> seems to be truncated [13:32:44 CET] <mmu_man> uint8_t line[1024]; [13:32:46 CET] <mmu_man> bleh& [13:34:33 CET] <mmu_man> oh [13:35:35 CET] <mmu_man> & [13:38:22 CET] <mmu_man> \o/ [13:38:30 CET] <mmu_man> en forçant le buffer à 256k ça passe [13:38:40 CET] <mmu_man> sorry [13:38:49 CET] <mmu_man> forcing the buffer size to 256k worked [13:44:42 CET] <mmu_man> hmm aviobuff:ff_get_line & ffmetadec:get_line could use some factoring [14:51:05 CET] <mmu_man> patch sent [14:54:28 CET] <wr> hello, need a help on converting a mp4 video file on the ffmpeg to ogv, how can do it? [15:29:52 CET] <saml> wr, ffmpeg -i a.mp4 -c:v libtheora -c:a libvorbis a.ogv [15:30:37 CET] <wr> ah thanks [15:54:50 CET] <Fenrirthviti> Hey folks, I know this isn't really your problem, but looking for any leads on what might be causing this crash: https://obsproject.com/logs/9OJ_4cLDPVgO6i8h [15:55:02 CET] <Fenrirthviti> NVENC startup crash when using ffmpeg as the output [15:55:51 CET] <Fenrirthviti> Curious if anything similar has been seen with ffmpeg natively or not [16:20:31 CET] <BtbN> Impossible to say without debug symbols, but I'd blame a broken driver. [16:20:48 CET] <BtbN> Or really horribly broken parameters. [16:23:33 CET] <zerodefect> Agreed about debug symbols. c0000005 means `Access Violation` from memory. It's not very happy. [16:24:54 CET] <Mavrik> Feels violated. [16:32:03 CET] <wr> saml, the video just turned to be more low quality than the mp4, any way i increase it? [16:32:46 CET] <relaxed> wr: https://trac.ffmpeg.org/wiki/TheoraVorbisEncodingGuide [16:33:17 CET] <saml> you mean it became retro and better [16:33:35 CET] <wr> saml, too much retro lol [16:45:49 CET] <kepstin> well, you asked for theora, you got retro ;) [16:47:37 CET] <kepstin> why do you want theora anyways? pretty much the only thing that's needed for is obsolete versions of firefox, afaik. [18:49:56 CET] <pi-> I'm experiencing a strange problem. [18:50:07 CET] <pi-> I have a source .caf waveform. [18:50:42 CET] <pi-> If I convert it to .wav with Audacity (which if I remember correctly uses ffmpeg), my DSP code processes it correctly. [18:51:09 CET] <pi-> However if I use `ffmpeg -i X.caf X.wav`, the code now fails to process it correctly. [18:52:07 CET] <kepstin> can you pastebin the output of that ffmpeg command please? [18:52:38 CET] <kepstin> and also what do you mean by "fails to process it correctly?" how/where does it fail? [18:54:43 CET] <kepstin> with that command, ffmpeg will convert the audio to 16-bit little-endian pcm in the wav file, but it should preserve the sample rate and number of channels from the input. [18:56:41 CET] <pi-> kepstin: https://paste.pound-python.org/show/44sRGMU0wu3bZ3eRysnL/ [18:57:43 CET] <pi-> If I examine hexdumps of the generated WAVs (both from audacity and from ffmpeg), they do look strange. Every single 4-cluster ends ??00 or ??ff. [18:58:10 CET] <kepstin> yeah, that's converting from big-endian audio (in the caf) to little-endian (in the wav). If your DSP code expects the opposite endian and doesn't read the wav headers properly that could cause issues [18:58:38 CET] <pi-> I am attempting to extract an ultrasound signal, so 'fails' is difficult to define. [18:58:51 CET] <pi-> It is actually a partial failure. [18:59:05 CET] <pi-> One part of the signal gets recognised successfully, another part doesn't. [18:59:42 CET] <pi-> I wrote my own WAV reader, so it's possible I'm not processing the header right. [18:59:50 CET] <kepstin> how in the world would you get ultrasound stuff from a 48kHz audio file? if it's lowpassed correctly, it won't have anything above 24kHz. [19:00:00 CET] <pi-> well near-ultra [19:00:03 CET] <pi-> 18-20kHz [19:00:11 CET] <kepstin> ah, yeah. [19:00:40 CET] <kepstin> one thing you can try is to add "-c:a pcm_s16be" to the ffmpeg command, to leave the audio in big-endian in the wav file, see if that helps [19:01:23 CET] <kepstin> if it does, then your app is expecting big-endian audio, and it isn't reading the wav headers correctly or isn't doing the required conversion before processing. [19:02:20 CET] <kepstin> oh, nvm, wav doesn't support big endian at all [19:02:23 CET] <kepstin> my mistake [19:02:27 CET] <pi-> the endianness appears to be consistent between ffmpeg and audacity output. [19:02:30 CET] <kepstin> (it's a PC/windows thing) [19:03:05 CET] <pi-> I'm guessing the xx00 and xxff everywhere means small positive and small negative numbers [19:03:37 CET] <pi-> i.e. between -/+ 255 over a total range of -/+ 32768 [19:18:19 CET] <kepstin> yeah, that makes sense. [19:19:50 CET] <kepstin> in the caf file, i'd expect the values to all look like 00xx or FFxx [19:30:39 CET] <GlidingHorse> hi, can I mux HEVC into an mpegts with ffmpeg 4.0? [19:31:03 CET] <GlidingHorse> i'm not asking how ,just asking if it's supported [19:31:13 CET] <durandal_1707> no [19:31:45 CET] <GlidingHorse> :( [19:31:59 CET] <kepstin> pi-: hmm. one thing to note is that ffmpeg usually adds some metadata to the wav file, you want to make sure your wav reader correctly locates the start of the "data" chunk. [19:32:32 CET] <durandal_1707> GlidingHorse: or yes, dunno i thought you ask about av1 [19:32:55 CET] <GlidingHorse> :) [19:33:47 CET] <GlidingHorse> anyway i will try by myself, but wanted to gain some time while ffmpeg downloads [19:34:15 CET] <durandal_1707> it should be possible, i see lines with HEVC in mpegts muxer [19:34:40 CET] <GlidingHorse> excelent :D [20:08:25 CET] <GlidingHorse> indeed, ffmpeg muxed hevc video and e-ac-3 audio into mpegts :D [20:08:35 CET] <GlidingHorse> thanks for the support :D [20:08:37 CET] <GlidingHorse> o/ [20:08:38 CET] <GlidingHorse> \part [00:00:00 CET] --- Thu Nov 8 2018

1 0

ffmpeg-devel.log.20181106
by burek 07 Nov '18

07 Nov '18

[00:02:05 CET] <jamrial> jkqxz: if you're also pushing them to release/4.1 yourself, do the same with a4fb2b1150 and e9d2e3fdaa while at it [00:03:47 CET] <jkqxz> Ok, sure. [00:07:00 CET] <cone-309> ffmpeg 03Mark Thompson 07master:6bdb7712ae02: cbs_av1: Fix header writing when already aligned [00:07:01 CET] <cone-309> ffmpeg 03Mark Thompson 07master:f5894178fb80: cbs_av1: Support redundant frame headers [00:17:37 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:066ff02621ce: configure: Add missing IVF muxer BSF dependency [00:17:38 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:ec1b5216fc65: configure: Add missing V4L2 M2M decoder BSF dependencies [00:17:39 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:af3fccfeff74: cbs_av1: Fix header writing when already aligned [00:17:40 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:10506de9ad1f: cbs_av1: Support redundant frame headers [01:09:27 CET] <cone-309> ffmpeg 03Mark Thompson 07n4.1:HEAD: cbs_av1: Support redundant frame headers [01:25:13 CET] <cone-309> ffmpeg 03Mark Thompson 07n4.2-dev:HEAD: cbs_av1: Support redundant frame headers [02:35:21 CET] <cone-309> ffmpeg 03James Almer 07master:beaa350e2416: avcodec: libdav1d AV1 decoder wrapper. [10:51:55 CET] <cone-871> ffmpeg 03Linjie Fu 07master:87368884a52b: lavc/qsvdec: flush buffered data before reinit [10:51:56 CET] <cone-871> ffmpeg 03Zhong Li 07master:d96ae9d5ea1f: lavc/qsv_overlay: suppress code scan complain [12:40:56 CET] <funman> sys.path.append(os.path.join(sys.MEIPOASS, 'out00-PYZ.pyz')) [12:42:55 CET] <JEEB> what's the best way to figure out which part is blocking with f.ex. ffmpeg.c? [12:43:28 CET] <JEEB> as in, is it filter chain with splits etc, or is it encoding or what [12:43:45 CET] <JEEB> what are good tools to profile such stuff? [12:44:05 CET] <nevcairiel> i would go low-tech and test the invidiual steps individually [12:44:14 CET] <nevcairiel> measure their speed [12:44:41 CET] <JEEB> it's probablematic with live inptus [12:44:50 CET] <JEEB> I could of course dump and check [12:45:09 CET] <JEEB> but then you're not limited in input speed unless I use something like multicat [12:45:22 CET] <JEEB> which outputs MPEG-TS into UDP with speed control [12:45:32 CET] <nevcairiel> of course, but if filtering and encoding is like 10x realtime, its not going to limit you [12:46:27 CET] <JEEB> true. it's just that I have a hunch that I have a case of a filter chain getting stuck or so on one of the outputs or something like that [12:46:42 CET] <JEEB> but yea, I think that makes sense. dump some and see the steps separately [12:46:48 CET] <JEEB> also the overall load is probably one thing [14:06:11 CET] <Compn> ahhh [14:06:16 CET] <Compn> we still don't have qdmc decoder ? [14:06:54 CET] <Compn> no wait we do [14:06:55 CET] <Compn> hmm [14:07:53 CET] <Compn> is my vlc so old it does not have this mapped ? [14:08:45 CET] <durandal_1707> yes [14:09:58 CET] <Compn> durandal_1707 : btw i have imm5 sample (and other samples) for you [14:10:32 CET] <durandal_1707> i accept only money [14:17:43 CET] <Compn> sent you link [14:18:02 CET] <durandal_1707> link is not money, ignored [14:18:21 CET] <Compn> j-b iirc said he would pay you for imm5 decoder [14:18:30 CET] <Compn> there is imm5 sample in there [14:18:44 CET] <durandal_1707> what is in sample? [14:19:06 CET] <Compn> boring video , security cam or dash cam [23:07:03 CET] <durandal_1707> i'm retiring [23:13:18 CET] <durandal_1707> nobody cares, eh! [23:14:35 CET] <jamrial> i do! [23:14:36 CET] <j-b> retiring from? [23:16:14 CET] <Compn> durandal_1707 : retiring from ffmpeg ? [00:00:00 CET] --- Wed Nov 7 2018

1 0

ffmpeg.log.20181106
by burek 07 Nov '18

07 Nov '18

[00:38:52 CET] <petecouture> Building ffmpeg on centos Im getting an error thrown ERROR: libfdk_aac not found but the library was downloaded and installed as rpm which worked for librtmp. Im reading that there maybe an issue with the new version for centos 7 not including the lib into the name. Anyone else run into this? [00:39:58 CET] <pink_mist> doesn't centos require -devel versions of all such libraries if you want to build something that links to them? [00:40:55 CET] <petecouture> @pink_mist thanks I will review. I didnt run into this with the librtmp library. [00:43:41 CET] <pink_mist> I've never actually used centos, so I may well be mistaken [00:43:46 CET] <petecouture> @pink_mist that worked you rock! [00:43:51 CET] <pink_mist> oh good =) [10:28:46 CET] <mrtAkdeniz> Hey there [10:29:17 CET] <mrtAkdeniz> Im trying to use an image as camera source.. Can I do that with ffmpeg? [10:29:46 CET] <mrtAkdeniz> I tried to google it, but probably used wrong keywords.. [10:51:42 CET] <thin> If I want to take a bunch of webm's and supercut random 1-second intervals from all of them into a video, what's my best option [13:02:22 CET] <haasn> ffmpeg: code 3074: invalid colorspace definition [13:02:24 CET] <haasn> huh? [13:03:21 CET] <haasn> oh nvm I'm stupid [14:12:50 CET] <spaam> w [17:20:11 CET] <AlekseyVl89> Hello, huys [17:20:14 CET] <AlekseyVl89> Hello, guys! [17:21:02 CET] <AlekseyVl89> I'm stuck with the problem. ERROR: opus not found using pkg-config [17:21:26 CET] <AlekseyVl89> I'm trying to make ffmpeg for android. I tried on mac os and Ubuntu. And got the same error [17:21:44 CET] <AlekseyVl89> I've installed libopus with apt-get install libopus-dev [17:22:19 CET] <bencoh> what do you mean "for android"? [17:22:23 CET] <bencoh> is that cross-compilation? [17:22:30 CET] <kepstin> you need to build all the dependent libraries (e.g. libopus, etc.) in your cross-compile build environment. You can't use your system libraries (probably x86) for android (probably arm) [17:22:32 CET] <lays147> Hello guys, I am trying to cut a video, however, the cutting is not happening, the param '-c copy' is mandatory? because its apparently giving conflict with the -vf param [17:22:36 CET] <bencoh> if so, then installing a native package won't help [17:22:59 CET] <kepstin> lays147: if you're filtering video, then you can't copy it, because if you copy it it can't be modified by filters. [17:24:28 CET] <AlekseyVl89> bencoh, thank you. [17:24:44 CET] <AlekseyVl89> I managed to build libopus for android as static library [17:24:54 CET] <AlekseyVl89> How can I compile ffmpeg with it? [17:25:45 CET] <kepstin> AlekseyVl89: after installing libopus to your cross environment root, set PKG_CONFIG_PATH to contain the appropriate pkg-config directory in that corss env. [17:26:54 CET] <AlekseyVl89> like this? PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \ --pkg-config-flags="--static" \ [17:27:02 CET] <bencoh> and assuming you're properly generating your sysroot, you might want to have a look at PKG_CONFIG_SYSROOT_DIR and PKG_CONFIG_LIBDIR [17:29:56 CET] <AlekseyVl89> Thanks, guys [20:16:42 CET] <ChocolateArmpits> How is fpsprobesize different from probesize? I have a dv file with messed up audio that doesn't start at beginning, increasing probesize or analyzeduration doesn't help in detecting it initially, however using fpsprobesize helps. [20:26:42 CET] <frendShot> if I use the minterpolate filter to go from a 29.97 fps video to a 60fps one, is it normal for the audio to go out of sync? [20:31:24 CET] <No0n3Left> Do you mean the audio is distorted or that it is not in sync with the video? Assuming it would get a bit out of sync with the video just the framerate has changed and it isnt a multiple of the original, not sure though. [20:32:39 CET] <frendShot> as in, it's out of sync with the video; actions take place at one time in the video, the sound doesn't coincide with them [20:33:44 CET] <frendShot> the sound itself is identical; the minterpolate filter didn't like it when I gave it the video with the audio also muxed in, so I separated the two losslessly, and used the video stream alone with minterpolate [20:33:55 CET] <frendShot> and muxed the audio back in, and that's when it's out of sync [20:33:56 CET] <No0n3Left> Is the video still the same length? [20:34:41 CET] <frendShot> *exactly* the same length? I doubt it; I interpolated from 29.97fps to 60fps... but that shouldn't be that important, right? [20:34:59 CET] <No0n3Left> Do events take place at the same time if you play them back at the same time? [20:35:26 CET] <frendShot> hm? Sorry, I didn't understand [20:35:56 CET] <No0n3Left> Like open the videos at the exact same time and start them. Do they stay in sync, not the audio, but hte actual video itself? [20:36:14 CET] <frendShot> oh I see, the original 29.97 fps and the new 60 fps ones [20:36:16 CET] <frendShot> let me check [20:36:46 CET] <No0n3Left> Cause otherwise it may be something happening when you put the audio back in [20:39:38 CET] <frendShot> heh, the events take place at different times... I think I see what happened; the new interpolated frames occur before the sound [21:01:40 CET] <kepstin> frendShot: the minterpolate filter should preserve video timing. for best results you should have the audio and video in the same file so ffmpeg can manage timestamps for everything. [21:02:02 CET] <kepstin> it sounds like your video originally had an offset between audio and video start, and that was lost when you separated them [21:02:15 CET] <frendShot> kepstin, that's the thing, the minterpolate filter didn't like it when I gave it the video with the audio also muxed in [21:02:17 CET] <saml> if you change framerate, you have to reencode audio, right? [21:02:25 CET] <frendShot> so I had to demux it and then use the filter [21:02:37 CET] <kepstin> frendShot: minterpolate filter has nothing to do with audio [21:02:41 CET] <kepstin> frendShot: it should work fine [21:02:51 CET] <frendShot> well, I understand that, but ffmpeg threw some error about it [21:02:58 CET] <frendShot> I could try and replicate it and paste output here in pastebin [21:02:59 CET] <kepstin> frendShot: we can help you with that... [21:03:05 CET] <saml> usually, when I alter video with -c:a copy, there is a-v sync problem [21:03:11 CET] <frendShot> saml, I see [21:03:22 CET] <saml> so, i'm lazy and do -c:a aac or something [21:03:33 CET] <frendShot> yeah, I used -c:a copy to demux the audio, I used an m4a file since it was an AAC stream from a DVD [21:03:34 CET] <saml> there could be a smarter way [21:03:57 CET] <kepstin> saml: hmm, i'd really expect -c:a copy to keep the offsets/timing correct. [21:04:05 CET] <frendShot> ikr [21:05:13 CET] <saml> hrm yeah i could've done something else. if i experience a-v sync problem with exact command line i used, i'll report [21:16:30 CET] <markweston> how to change L/R audio balance in ffplay or somehow else? [21:18:22 CET] <ChocolateArmpits> markweston, for a single file ? [21:18:23 CET] <ChocolateArmpits> you'd have to use the pan audio filter [21:18:29 CET] <durandal_1707> markweston: stereotools is most user friendly way [21:24:29 CET] <markweston> hey, any way to change L/R audio balance? [21:24:39 CET] <markweston> sorry, wrong channel, I already asked [21:24:43 CET] <frendShot> lol [21:24:54 CET] <frendShot> I was going to say, didn't you see ChocolateArmpits and durandal_1707's answers [21:26:03 CET] <markweston> I just don't want to re-encode the whole movie and OpenBSD doesn't have stereotools in packages, so I'll see if I can do it on OS level with sndio or maybe there's some other media player that can do it on the fly [21:27:13 CET] <kepstin> stereotools is an ffmpeg filter, not a separate package. But yeah, many players will be able to do this, I'm pretty sure mpv can at least. [21:28:51 CET] <durandal_1707> markweston: mpv can do it in fly, get and copy this script: https://github.com/wiiaboo/mpv-scripts/blob/master/audio-balance.lua [21:29:59 CET] <frendShot> woops, wrong button [21:30:07 CET] <frendShot> kepstin, everyone else, https://pastebin.com/raw/1SBsjbrL this is the output [21:30:20 CET] <frendShot> the input is ".\ffmpeg.exe -i .\out2.mkv -filter "minterpolate='fps=60'" output.60fps.mkv" [21:30:25 CET] <frendShot> (Windows PowerShell) [21:30:35 CET] <frendShot> the few lines of errors are at the bottom [21:30:40 CET] <kepstin> frendShot: and your ffmpeg command line? [21:30:50 CET] <frendShot> .\ffmpeg.exe -i .\out2.mkv -filter "minterpolate='fps=60'" output.60fps.mkv [21:31:00 CET] <frendShot> that line had my name in it, so I didn't copy it [21:31:03 CET] <kepstin> oh, the issue is that you used -filter without a stream specifier [21:31:15 CET] Action: frendShot eyes bug out [21:31:15 CET] <kepstin> either use -filter:v to apply it only to video streams, or use -vf (short option) [21:31:21 CET] <frendShot> I see [21:31:36 CET] <markweston> durandal_l1707: thanks man, it's ironic I got the link for the script on #ffmpeg and not on #mpv, where I asked too. The guy there only said "iirc there was a script to do that" but didn't say anything else [21:32:25 CET] <kepstin> frendShot: you'll want to specify some codec options, something like "-c copy -c:v libx264" will copy everything except the video, which will be re-encoded with x264. [21:32:54 CET] <frendShot> I see, and to retain lossless video I use -crf 0 with libx264 right? Even though I guess the file will be huuuuge [21:33:06 CET] <kepstin> frendShot: lossless video is a bad idea [21:33:06 CET] <frendShot> i'm not sure how lossless video will work with framerate change though [21:33:24 CET] <kepstin> use a crf value that gives no quality loss you notice, and that's good enough [21:33:32 CET] <frendShot> right, so transparent [21:33:47 CET] <kepstin> frame interpolation requires making changes to video content, so it is by definition not lossless [21:34:04 CET] <frendShot> yeah, that's what I figured ... so what exactly does -crf 0 here do then? [21:35:07 CET] <kepstin> -crf 0 in 8-bit x264 (the general option for all modes is actually -qp 0) turns on lossless mode, where when you decode the resulting video, you'll get back the exact same bits as were sent to the encoder [21:35:17 CET] <frendShot> i see [21:35:23 CET] <kepstin> this requires huge amounts of space compared to normal lossy video encoding. [21:35:36 CET] <frendShot> yeah, and in the past has given me compatibility issues, too [21:35:58 CET] <kepstin> yes, it requires the decoder support a profile that many hardware decoders and players don't do. [21:36:05 CET] <frendShot> yeah [21:37:17 CET] <kepstin> basically, the only reason to use lossless mode is for certain special archiving tasks where you need to ensure the encoder does not add any artifacts whatsoever, and as a temporary intermediate format to avoid generation loss from multiple encoding/decoding steps. [21:37:36 CET] <frendShot> sweet [21:37:44 CET] <frendShot> all right, lemme try these commands out and report back, thanks [21:50:22 CET] <frendShot> kepstin, thanks, this seems to have done the trick [21:50:38 CET] <frendShot> I guess somewhere along the muxing and demuxing steps, something got misaligned [21:51:06 CET] <aleph-> Hey is there any way at all to get multi core/parallelized encoding with av1? [21:51:48 CET] <kepstin> aleph-: are you a programmer familiar with codecs and threading? maybe you can help! :) [21:52:27 CET] <aleph-> Threading somewhat, codecs not in the slightest. [21:52:43 CET] <aleph-> Taking that as a no as well. :P [21:53:24 CET] <kepstin> av1 isn't really a codec ready for general usage yet, there's a lot more work on the encoder needed. [21:53:35 CET] <aleph-> Aye I figured. [21:53:38 CET] <frendShot> isn't it super slow at the moment? [21:54:03 CET] <aleph-> Indeed. [21:54:13 CET] <kepstin> well, iirc rav1e is only very slow rather than super slow, but iirc it's also rather less efficient? [21:54:27 CET] <kepstin> libaom is super slow, yeah [21:54:46 CET] <aleph-> How is dav1d? [21:55:02 CET] <kepstin> dav1d is a decoder, not encoder [21:55:06 CET] <aleph-> Oh doy. [21:55:33 CET] <aleph-> And I've heard rav1e is rather limited on input type right? [21:55:56 CET] <kepstin> aleph-: most standalone encoders are. Handily, most accept input piped from ffmpeg. [21:56:16 CET] <aleph-> Aye [21:56:49 CET] <kepstin> i mean, if you want to write a standalone tool that accepts arbitrary input, the way to do that is to use the ffmpeg libraries to decode the input :/ [23:08:37 CET] <benbro> [libx264 @ 0x557ffb0db440] width not divisible by 2 (329x308) [23:08:44 CET] <benbro> Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [23:08:59 CET] <benbro> I'm using 'format=yuv420p,scale={0}:-2' [23:09:15 CET] <benbro> shouldn't it handle width 329? [23:09:50 CET] <klaxa> nope [23:10:14 CET] <benbro> klaxa: any workaround? [23:10:28 CET] <pink_mist> a lot of formats require a multiple of 8 or something like that I believe [23:10:46 CET] <klaxa> *2 / 2? [23:10:52 CET] <klaxa> err... [23:10:55 CET] <benbro> ? [23:10:57 CET] <pink_mist> you could add black bars to pad it out to a multiple of 8 [23:11:02 CET] <c_14> yuv420 ist even numbers only [23:11:16 CET] <pink_mist> ah, ok, multiple of 2 for that one then [23:11:20 CET] <c_14> -t [23:11:53 CET] <benbro> no automatic way to handle it? [23:12:09 CET] <c_14> there is [23:12:13 CET] <benbro> what? [23:12:14 CET] <c_14> eeeh [23:12:43 CET] <klaxa> mathematically, integer division by 2 and multiplication by 2 should yield even numbers [23:13:02 CET] <benbro> -vf "pad=ceil(iw/2)*2:ceil(ih/2)*2" ? [23:13:12 CET] <benbro> found this here: https://stackoverflow.com/a/20848224 [23:13:15 CET] <c_14> yes, that [23:13:16 CET] <benbro> but feels dirty [23:13:18 CET] <c_14> but only for the first one [23:13:20 CET] <c_14> the second can stay -2 [23:14:46 CET] <c_14> and you can use your {0} variable for iw [23:15:29 CET] <benbro> my {0} variable is divisible by 2 [23:15:56 CET] <c_14> then that should work [23:16:18 CET] <benbro> checking [23:21:32 CET] <benbro> the error is intermittent [23:21:41 CET] <benbro> sometimes I see the error and sometimes not [23:21:56 CET] <benbro> I don't understand how it's possible [23:21:59 CET] <c_14> log your {0} value, maybe your code is bugged? [23:22:04 CET] <frendShot> then your variable sometimes isn't divisible by two [23:22:12 CET] <frendShot> have your code print out what it is at each step to debug it [23:22:56 CET] <furq> you should probably be doing scale,format and not the other way around [23:23:12 CET] <c_14> oh, yeah [23:23:23 CET] <furq> although you'd get a different error message in that case [23:23:24 CET] <c_14> if your input isn't even that would break it [23:23:32 CET] <c_14> *should [23:23:51 CET] <furq> that's just generally good practice anyway [23:26:25 CET] <benbro> furq: I'll try to switch [23:26:38 CET] <benbro> can it explain why it sometimes fail and sometimes doesn't fail? [23:27:10 CET] <furq> yuv420p resizes the chroma planes to half width/height of the luma plane [23:27:23 CET] <furq> so if the width is an odd number then it can't do that [23:28:01 CET] <benbro> so it should always fail for the same video [23:28:03 CET] <furq> i guess it could just make up a pixel but it's probably better to just get you to do it [23:28:12 CET] <benbro> ok. trying [23:28:21 CET] <furq> yeah if it's intermittently failing on the same source clip then idk what the problem would be [23:29:11 CET] <benbro> furq: ok. trying to scale before format [23:35:37 CET] <frendShot> is there a preferred order when specifying filters? like, if I wanted to upscale and interpolate in the same command [23:38:56 CET] <furq> frendShot: it depends on the filters [23:39:29 CET] <benbro> furq: I have predefined resolutions but if the video width is smaller than all variations I just use it as is [23:39:33 CET] <furq> generally you want to do stuff before scaling so it has more information to work with [23:39:42 CET] <benbro> and in this case it was uneven number [23:39:51 CET] <furq> but if it's really slow (like minterpolate) then you might want to do it afterwards [23:39:52 CET] <frendShot> furq, to be specific, minterpolate='fps=60' and scale=1080:720 [23:39:57 CET] <frendShot> I see [23:40:10 CET] <benbro> furq: should I pad with 1 pixel in this case? [23:40:19 CET] <furq> benbro: pad or scale, it's up to you [23:41:06 CET] <benbro> furq: is it better to add or remove a 1 pixel? or maybe doesn't matter? [23:41:09 CET] <furq> i would normally just crop a pixel off, but it's entirely down to preference [23:41:43 CET] <benbro> crop instead of scale? [23:41:44 CET] <furq> if you don't mind the black bar then padding is the least destructive [23:42:24 CET] <benbro> 'scale=440:-2' if the original is 339px will just stretch one of the pixels? [23:42:54 CET] <furq> if you mean 340 then yeah [23:43:13 CET] <benbro> sorry, I mean 439 [23:43:39 CET] <benbro> I want to add/remove/scale without changing much the size and least cpu overhead [23:44:03 CET] <benbro> I'll choose nicer numbers [23:44:19 CET] <pink_mist> I believe it's better to pad if it's just one single line of pixels [23:44:39 CET] <benbro> if the original video width=499, should I "scale=500:-2", or pad or crop? [23:44:52 CET] <furq> like i said, it's entirely down to preference [23:45:06 CET] <furq> any way of adding or dropping one pixel is fine [23:45:30 CET] <benbro> "scale=500:-2" won't distort a 499px video too much? [23:46:02 CET] <pink_mist> try it and compare to the other options [23:46:27 CET] <furq> you could try pad and fillborders if you want to avoid the black bar [23:46:51 CET] <benbro> 1px black bar probably won't be noticible [23:47:14 CET] <benbro> scale is easier for me in the code because I'll treat all videos the same [00:00:00 CET] --- Wed Nov 7 2018

1 0

ffmpeg-devel.log.20181105
by burek 06 Nov '18

06 Nov '18

[01:10:58 CET] <jkqxz> jamrial: Sent a new version. [01:12:51 CET] <jamrial> jkqxz: can't see it in my inbox yet, but will test it once it's there, thanks [01:32:16 CET] <jkqxz> jamrial: Did you have any thoughts on including some AV1 streams in fate for testing? [01:33:29 CET] <jamrial> jkqxz: the aom testsuite is pretty barebones (just different quantizer setting during encoding), but it's enough for some basic tests [01:34:18 CET] <jamrial> aside from those, we could include some of the small samples i created using Tears of Steel to test parsing things like error resilence and redundant frame headers, timing info using decode model, and such [01:39:30 CET] <cone-223> ffmpeg 03Mark Thompson 07master:34429182b931: configure: Fix av1_metadata BSF dependency [01:41:34 CET] <jamrial> jkqxz: https://pastebin.com/qAxM1zBc [01:42:15 CET] <jamrial> file is unreadable after that. libaom can't read it, and i can't seem to get the ivf demuxer to output anything [01:42:56 CET] <jkqxz> Since it's read in order you lose the first tile group. [01:43:27 CET] <jkqxz> I was looking at trace_headers to show that the redundant frame header was used correctly. [01:43:54 CET] <jamrial> i did an av1_metadata passthrough and it works after your patch. checksum matches [01:44:19 CET] <jamrial> before it would crash and burn [01:44:49 CET] <jkqxz> Yep, your file passes through to be identical. I don't have any others to try, though. [01:45:30 CET] <jamrial> you can create them with aomenc by using error resilence mode with 2+ tile groups [01:45:58 CET] <jamrial> that's the only case where it adds redundant frame headers in between each tile group on every frame except key frames [01:57:16 CET] <cone-223> ffmpeg 03James Almer 07master:89a0d33e3a53: avcodec/cbs_av1: fix decoder/encoder_buffer_delay variable types [02:07:28 CET] <cone-223> ffmpeg 03James Almer 07release/4.1:1c98cf4ddd38: avformat/ivfenc: use the av1_metadata bsf to insert Temporal Delimiter OBUs if needed [02:07:29 CET] <cone-223> ffmpeg 03Mark Thompson 07release/4.1:acd13f12556b: configure: Fix av1_metadata BSF dependency [02:07:30 CET] <cone-223> ffmpeg 03James Almer 07release/4.1:398a70309e4f: avcodec/cbs_av1: fix decoder/encoder_buffer_delay variable types [02:18:06 CET] <nicolas17> can ffmpeg read AV1 without extra libraries? [02:18:12 CET] <jkqxz> If I made the bit comparison come one byte at a time rather than one bit at a time would that be better? [02:20:07 CET] <jkqxz> It could certainly be suppressed by splitting into read/write branches and just doing the necessary bit(read|write)er operations, but I think it's preferable to show something there. [02:21:12 CET] <jkqxz> Wrt refcounting, yes for read only. Again it needs to split the read/write branches out further, so code becomes even more horrible. [02:21:24 CET] <jkqxz> nicolas17: Depends what you mean by "read". [02:24:59 CET] <jamrial> nicolas17: decode? no. Parse, extract, remux? yes [02:47:32 CET] <BBB> nicolas17: for av1, some of us decided to write the decoder outside ffmpeg, so its a library to be linked in. some others are planning to eventually integrate the code directly into ffmpeg, which is fine but not yet done [13:41:32 CET] <dualz> I'm writing a filter that takes two inputs and has two outputs (audio and video). I noticed that filters with two input pads do not receive a filter_frame call back. Upon further reading it says I should use activate() so I messed around trying to get it to work and am dropping frames. I saw an example of a filter with multiple inputs (streamselect) and they use FrameSync, but FS doesn't work with multiple outputs [13:43:21 CET] <dualz> A little guidance on how to use activate() would be appreciated. I can't find a single filter that has two inputs and two outputs for an example [13:45:30 CET] <durandal_1707> dualz: what your filter will do that it needs 2->2 ? [13:46:37 CET] <dualz> for the time being I want a simple pass through [13:47:00 CET] <durandal_1707> framesync as name says, is for receiving synced frames from all inputs, eg, it will buffer to match same time before filtering [13:48:29 CET] <durandal_1707> dualz: afir filter have 2 audio inputs and audio output + video output, but its very specific usecase [13:49:11 CET] <durandal_1707> so doing only passthrough with activate is easy [13:49:40 CET] <dualz> is there a writeup somewhere that'll help? [13:50:04 CET] <durandal_1707> just consume frame from all inputs and output what you want with whatever pts you seem fit [13:51:35 CET] <durandal_1707> look at concat filter [13:51:55 CET] <durandal_1707> and feel free to ask questions if something is not clear [13:52:03 CET] <dualz> so activate { for each input { consume_frame } then for each output filter_frame } ? [13:52:18 CET] <durandal_1707> ff_filter_frame [13:52:27 CET] <dualz> can I call filter frame twice in one activate? [13:54:10 CET] <durandal_1707> dualz: activate does one thing only per call, so better no, best is output one frame and then have state what to push when [13:54:56 CET] <durandal_1707> also you need to signal that you need frames from inlinks [13:55:16 CET] <durandal_1707> and need to handle eof on inlinks and outlinks [13:58:07 CET] <dualz> if activate does one thing, how do I know to consume my audio input or my video input [13:58:50 CET] <dualz> sorry for the noob questions, I just started with ffmpeg [14:00:54 CET] <durandal_1707> dualz: by keeping state in some variable in filter private context [14:01:25 CET] <durandal_1707> and you can consume both video and audio at same time if its availabe [14:01:37 CET] <durandal_1707> just not output both... [14:05:03 CET] <dualz> so I consume both audio and video, I now how a AVFrame* video and *audio when is the best time to output these [14:05:28 CET] <dualz> if I can only output 1 per activate [14:06:30 CET] <dualz> store a reference in my filter context then wait for the next activate? [14:06:32 CET] <durandal_1707> you keep track of consumed frames in context, if you filter out consumed frame, you set it to NULL, so you need to pick new one, if its not NULL you output it [14:15:51 CET] <dualz> durandal_1707: is this the right idea https://gist.github.com/ryanmarin/213606922dff8fd0ae1a513727260483 [14:16:54 CET] <dualz> with this design will there ever be a activate() call where ctx.abuf is null and it'll try to output ctx.vbuf [14:18:46 CET] <durandal_1707> dualz: ff_inlink_consume_frame may return 0 if there is no frame available [14:19:27 CET] <durandal_1707> also your gist have wrong check for null/0 case [14:20:08 CET] <durandal_1707> also you do not need to consume all at once, just when you need it [14:20:28 CET] <dualz> how would i know when i need it [14:23:03 CET] <durandal_1707> dualz: ff_outlink_frame_wanted() [14:23:30 CET] <durandal_1707> you call that that to check if next filter in chain wants frame [14:24:54 CET] <durandal_1707> and if you need frame from inlink you call ff_inlink_request_frame() [14:37:53 CET] <dualz> durandal_1707: https://gist.github.com/ryanmarin/083afed8e909167da24d119d662ab6bf closer? [14:41:58 CET] <durandal_1707> dualz: something like that, missing eof handling and forwarding outlink status to other links [14:42:41 CET] <dualz> will ctx.vbuf ever be null for the ctx.abuf code to run? [14:44:40 CET] <durandal_1707> you can add check for other buf to not request new one if there is one to output [14:45:18 CET] <durandal_1707> AV case may have several small audio frames for each video frame [14:45:59 CET] <durandal_1707> and you can use consume_samples to keep AV in sync [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:4e9cff282476: libavutil: Undeprecate the AVFrame reordered_opaque field [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:882ae091d48b: libx264: Pass the reordered_opaque field through the encoder [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:e7ed9d81bff0: flvdec: Rename FLV_STREAM_TYPE_DATA into FLV_STREAM_TYPE_SUBTITLE [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:d7638d8dfc3c: flvdec: Export unknown metadata packets as opaque data [15:10:39 CET] <dualz> ok so I'm looking at consume samples [15:10:55 CET] <dualz> there's a min and a max argument [15:11:17 CET] <dualz> where do I get these from? [15:12:38 CET] <atomnuker> wbs: mate, 882ae091d48b112004b977314884176841f12cef looks very sketchy [15:12:49 CET] <atomnuker> wouldn't this be better done using side data? [15:13:26 CET] <atomnuker> would avoid the need for a codec cap flag too [15:13:49 CET] <atomnuker> but most importantly you wouldn't be stuffing pointers in random avframe fields [15:14:03 CET] <nevcairiel> this isnt exactly new api, that existed for years [15:14:10 CET] <nevcairiel> also, read ML if you want to review patches =P [15:14:10 CET] <wbs> atomnuker: does the encoder already bring side data along from the AVFrame into the AVPacket? [15:14:32 CET] <wbs> otherwise, yes, that's a bigger and more complete framework for the same thing; reordered_opaque is the old lightweight mechanism [15:14:52 CET] <wbs> and even then, a codec cap would be needed if you wanted a caller to know whether it really works or not for each encoder [15:15:02 CET] <atomnuker> old lightweight? its a hack. [15:15:25 CET] <wbs> feel free to propose patches that does things differently, this has been on the ML for weeks [15:15:26 CET] <atomnuker> you're putting a random pointer in there [15:15:54 CET] <durandal_1707> dualz: just use min = max = ff_inlink_queued_samples [15:15:56 CET] <nevcairiel> its up to the user to specify the content of the field, avcodec does nothing of that sort [15:16:24 CET] <dualz> ok [15:16:39 CET] <atomnuker> wbs: I'd rather see this reverted and done properly [15:16:45 CET] <wbs> atomnuker: what "properly"? [15:16:49 CET] <atomnuker> side data [15:17:26 CET] <wbs> atomnuker: I don't stuff pointers into ints anywhere in that patch, even though that's one thing the caller of libavcodec _can_ do if they want to [15:17:46 CET] <wbs> it's the converse; libavcodec has got an opaque int64_t field you can use. x264 similarly has got a void* you can use [15:18:09 CET] <wbs> to cater for systems where sizeof(int64_t) > sizeof(void*) I keep the int64_t in a buffer in libavcodec and have x264 point to it [15:18:22 CET] <nevcairiel> you dont get to ignore the ML for weeks and then cry for a revert when it hits, you missed your chance, besides, this doesn't add any new API or bad hacks that haven't existed before, reordered_opaque is as old as avcodec [15:19:42 CET] <atomnuker> its a high bandwidth ML, pushing it and saying you can't change it anymore is ridiculous [15:20:42 CET] <wbs> if you contest the use of reordered_opaque here, you can convince nevcairiel and other external users to remove the mechanism altogether. this is just using the existing api for what it was meant for [15:22:39 CET] <atomnuker> sure its meant to be opaque but its a bit of a stretch to put a pointer in there [15:22:50 CET] <nevcairiel> thats the users doing [15:22:57 CET] <nevcairiel> avcodec does nothing l ike that or even suggests it [15:23:26 CET] <wbs> atomnuker: I don't put a pointer in there, where do you claim I'm doing that? [15:24:43 CET] Action: durandal_1707 grabs something... ML is not high bandwidth [15:27:40 CET] <atomnuker> wbs: erm right, nevermind then, the x264 commit message mislead me [15:27:57 CET] <atomnuker> nothing wrong here [15:28:33 CET] <cone-321> ffmpeg 03Mark Thompson 07master:a4fb2b115071: configure: Add missing IVF muxer BSF dependency [15:28:34 CET] <cone-321> ffmpeg 03Mark Thompson 07master:e9d2e3fdaacb: configure: Add missing V4L2 M2M decoder BSF dependencies [15:29:05 CET] <wbs> exactly; if our reordered_opaque would have been 32 bit only, we _could_ have considered squeezing it into x264's void* opaque, but we can't, hence the buffer [16:43:08 CET] <BBB> atomnuker: why would tristans patch be incompatible with simd? arent the two complementary? [16:43:25 CET] <BBB> atomnuker: his patch basically just adds a branch before calling the simd code, which you need anyway [16:46:42 CET] <cone-321> ffmpeg 03Valery Kot 07master:be827e1d38cb: libopenh264enc: Handle sample_aspect_ratio [17:49:41 CET] <atomnuker> BBB: why would you need the branch at all? [17:49:59 CET] <BBB> thats what the header bit does [17:50:08 CET] <BBB> when that header bit is set, the bitstream cdfs are non-per-symbol-adaptive [17:50:29 CET] <BBB> the cpu branch predictor should take care of making that efficient, Id expect [17:50:33 CET] <atomnuker> yes, but you can avoid the branch by just changing the function pointer to a function which does nothing [17:50:53 CET] <BBB> that means an extra call [17:50:57 CET] <BBB> vs. a predictable branch [17:51:00 CET] <BBB> seems unfavourable [17:51:02 CET] <BBB> right? [17:51:21 CET] <atomnuker> I'd disagree in this case [17:51:32 CET] <BBB> the extra call is slower, right? [17:52:02 CET] <atomnuker> optimally we'd like to have no function calls at all which means integrating decoding and updating into a single piece of asm [17:52:46 CET] <atomnuker> yes, its slower, but given how little overhead updating has with simd I'm not sure its even worth doing in pure asm [17:53:10 CET] <atomnuker> e.g. disabling adaptation [17:59:48 CET] <BBB> but the two approaches are sort of mutually incompatible, riht? [18:00:11 CET] <BBB> and youd still need a branch if you wanted the whole function in asm [18:13:03 CET] <atomnuker> for adaptation I'd say it would be unnecessary [18:14:52 CET] <atomnuker> as good as a branch predictor is I'd rather not rely on it unless there's no other choice [18:23:38 CET] <BBB> but an empty function call is worse [18:23:42 CET] <BBB> so what else do you propose then? [18:26:42 CET] <JEEB> rcombs: you make me feel dirty [18:26:50 CET] <rcombs> same [18:26:53 CET] <JEEB> I had to read through dolby documents [18:27:09 CET] <rcombs> the sample file I was looking at when I sent that seems to decode as normal-ass HEVC [18:27:20 CET] <JEEB> yea, it seems to depend on the profile [18:27:26 CET] <JEEB> for which they made additional boxes, of course [18:27:36 CET] <JEEB> I linked both specs (ISOBMFF and the general profiles= [18:27:46 CET] <rcombs> the colors look weird but idk it's an HDR sample, they always try to look ~vibrant~ [18:28:04 CET] <rcombs> it's not like it has a convenient set of color bars I can verify [18:28:24 CET] <rcombs> and even if it is decoding wrong, I'm gonna assert that something is better than nothing here [18:36:13 CET] <Gramner> atomnuker: indirect function calls are branches too as far as the branch predictis is concerned [18:36:39 CET] <Gramner> and indirect branches are predicted worse than normal ones since they can go anywhere, not just two ways [19:10:09 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:febaa63b0f7f: proresenc_anatoliy: Rename a profile name with the correct one [19:10:10 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:cdc487bfc841: prores: Use profile names in the various encoders and decoders [19:10:11 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:aefbb2bf029f: proresdec2: Parse codec_tag and export profile information [19:10:12 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:d37faad0cdbb: mjpeg: Use profile names in the encoder and decoder [22:04:33 CET] <dualz> durandal_1707: I got it working! I just need to test if the audio is falling out of sync or not [22:10:33 CET] <dualz> can anyone tell me when exactly and at what frequency activate() gets called? [22:36:03 CET] <cone-309> ffmpeg 03Ruiling Song 07master:952a299fd390: doc/filters: add document for opencl filters [22:36:03 CET] <cone-309> ffmpeg 03Ruiling Song 07master:a587454fd07c: doc/filters: add tonemap_opencl document. [23:53:58 CET] <durandal_1707> dualz: whenever is needed, there is set_ready to make it more frequent [23:55:11 CET] <michaelni> jamrial, are you done with the av1 fixes ? or there is more to wait for ? [23:56:00 CET] <jamrial> michaelni: two patches by jkqxz on the ml that i already reviewed need to be pushed and backported, and that's it [23:57:34 CET] <michaelni> jamrial, ok no hurry [23:59:32 CET] <jkqxz> Just getting that now. [00:00:00 CET] --- Tue Nov 6 2018

1 0

ffmpeg.log.20181105
by burek 06 Nov '18

06 Nov '18

[00:02:04 CET] <limbo_> furq: If you can suggest some settings, I'm all ears. Burning in subtitles optionally. I currently use mpv to do this (slowly) because everything else I tried fails on embedded fonts. (that aren't also on the system) [00:02:37 CET] <limbo_> Which is most of the re-encoding I do, most other things a PS4 can play. Even without remuxing out of mkv. [00:07:33 CET] <furq> i don't think attached fonts are supported by any of the hardsubbing filters in lavfi [00:07:45 CET] <furq> also i take it that you actually need the styling so you can't just use an external srt [00:07:51 CET] <kepstin> my impression was that ffmpeg can burn in subtitles including fonts if you use the "subtitles" filter. [00:07:56 CET] <kepstin> and give it the mkv file as input [00:08:11 CET] <furq> https://github.com/libass/libass/issues/266 [00:08:52 CET] <kepstin> huh. [00:08:57 CET] <nicolas17> IIRC in closed captioning on TV you get two (or more?) lines that scroll, when a new line appears, what used to be the last line scrolls up to make space [00:09:01 CET] <nicolas17> what subtitle format supports that? [00:09:24 CET] <furq> i've never actually tried it but the subtitles filter doesn't call ass_set_fonts after ass_process_codec_private [00:09:30 CET] <furq> so i assume it doesn't work [00:09:58 CET] <kepstin> nicolas17: you can fake it with most formats if your renderer has good timing, just have one sub say "line1 / line2" then the next "line2 / line3" [00:10:27 CET] <kepstin> nicolas17: in theory webvtt specifies this sort of thing, in practice i've haven't seen it implemented. [00:10:28 CET] <furq> also my favourite kind of bug report is one where one of the main contributors says "this should be easy to fix" and then nothing happens for two years [00:10:52 CET] <nicolas17> I think Apple's video player supports that in webvtt [00:11:26 CET] <furq> also i've seen that faked with srt and it was incredibly annoying [00:12:14 CET] <kepstin> I was playing around with trying that for a web video player caption generation tool, and just doing them normal subtitle style was way less annoying, yeah. [00:12:54 CET] <furq> it makes sense for live subtitling because the timing is screwed up there anyway [00:12:54 CET] <nicolas17> the truth is that I want to overlay a synced chat log on a video ^^ maybe I should render it myself and then overlay [00:12:59 CET] <furq> so you just want everything on screen as long as possible [00:14:41 CET] <furq> you could probably do something like that with ass [00:14:49 CET] <furq> but it sounds easier to just render and overlay [02:33:02 CET] <mixfix41> nicolas17: could subtitle editor work for tha? [02:33:13 CET] <mixfix41> that application does everything i need [02:33:59 CET] <mixfix41> oh with an srt file that is [05:27:09 CET] <mixfix41> its actually difficult for me to use handbrakecli but i think the gui is good [05:28:24 CET] <furq> handbrake is ok but make sure to never use decomb because it's terrible [05:35:04 CET] <mixfix41> heh heh [06:19:11 CET] <KombuchaKip> What is the equivalent to avformat_open_input(), but to open from a file loaded in memory? [06:21:28 CET] <nicolas17> I think that needs "custom IO" [06:21:36 CET] <KombuchaKip> nicolas17: Ok [06:22:04 CET] <nicolas17> documentation for avformat_open_input says "If you want to use custom IO, preallocate the format context and set its pb field." [06:22:14 CET] <xiando> I'm trying to into VP9 encoding and I tried: ffmpeg -i gyongree-171129_downpour.mp4 -threads 8 -c:a opus -b:a 192k -strict experimental -c:v libvpx-vp9 -crf 33 -b:v 0 test.webm <- while it makes a file it only uses one core and takes ages. [06:22:20 CET] <xiando> How can I into multi-core VP9 encoding? [06:22:51 CET] <furq> with difficulty [06:23:00 CET] <furq> i think -row-mt 1 should be all you need these days [06:23:07 CET] <furq> but if you have an older libvpx then it's a huge pain [06:23:40 CET] <KombuchaKip> nicolas17: Like so? https://stackoverflow.com/a/20610535 [06:26:30 CET] <xiando> furq: uhm.. -row-mt 1 did work and now it puts most cores under 20-30% load. instead of 0.64x I get 0.84x on that very low-resolution file. [06:26:47 CET] <furq> yeah libvpx has bad multithreadnig in general [06:28:45 CET] <furq> if it's low resolution then that's probably the reason [06:28:57 CET] <furq> it only works effectively on high res sources [06:29:58 CET] <furq> you might want to try explicitly setting -threads, idk if that's still necessary [06:30:02 CET] <furq> the docs for vpx are pretty bad [06:30:36 CET] <nicolas17> furq: the command he posted had -threads 8 [06:30:51 CET] <furq> so it did [06:30:59 CET] <furq> i guess it's just going to be slow then [06:31:01 CET] <xiando> tried a 1080p, puts a bit more load accross cores but not near 100%, more like 60%. Still, it's an improvement [06:31:15 CET] <xiando> speed is like 0.22x [06:32:06 CET] <furq> you can set -cpu-used if you want to make it fast [06:32:06 CET] <furq> er [06:32:09 CET] <furq> but it'll reduce the quality [06:33:46 CET] <xiando> I must have quality, I can't have blurry kpop [06:34:32 CET] <furq> i take it you're doing this to upload it somewhere [06:34:38 CET] <furq> otherwise i'd just leave it as h264 [06:34:38 CET] <KombuchaKip> nicolas17: Like so? https://stackoverflow.com/a/20610535 [06:35:21 CET] <nicolas17> I guess? I never did it [06:35:52 CET] <xiando> no my tablet can't have files bigger than 4 GB since it can't ext4 the SD card for some reason so I must reduce 6-7GB files to below 4 with quality. So I can use HEVC or VP9 [06:36:01 CET] <xiando> I'm not sharing. [06:36:16 CET] <KombuchaKip> nicolas17: Ok [06:37:11 CET] <nicolas17> exfat? [06:38:39 CET] <xiando> it's one that says it only supports 32 GB SD cards but if I format 64 GB cards as FAT they work fine [06:39:06 CET] <xiando> I guess I could buy a new one but I don't want to [06:39:22 CET] <nicolas17> how big is the SD card? [06:39:30 CET] <xiando> 64GB [06:39:42 CET] <nicolas17> did you try formatting it as exfat? [06:40:03 CET] <nicolas17> it should support files bigger than 4GB [06:40:51 CET] <xiando> I'm fairly sure the card was exfat when I got it. The tablet doesn't "support" cards bigger than 32 GB [06:41:00 CET] <xiando> which is why I had to mkfs.vfat it to make it work [06:41:07 CET] <xiando> it's a thing with older technology [06:41:32 CET] <nicolas17> and i guess you don't want to split it into two video files? [06:41:56 CET] <furq> yeah i would just split it [06:42:00 CET] <xiando> Now there's a simpler solution isn't it [06:42:19 CET] <furq> there's no way to reencode it to <4GB with good quality without spending days [06:42:42 CET] <furq> x265 is quicker than libvpx though fwiw [06:42:47 CET] <furq> since it actually has frame multithreading [06:43:12 CET] <furq> but it's still extremely slow at high quality settings [06:44:55 CET] <xiando> yeah it takes hours with x265, quicker than vp9 but splitting the file took seconds [06:45:08 CET] <nicolas17> uh how did you split it? [06:45:44 CET] <xiando> avidemux-qt, just opened it, selected first half, save, selected second half, saved that, done [06:46:31 CET] <xiando> file operations are quick on raid ssds [06:46:47 CET] <nicolas17> ah nice :D [06:46:51 CET] <nicolas17> I'm still on spinning rust [06:48:00 CET] <xiando> what, there's a big difference for opening programs and everything, just get a SSD as OS disc, seriously. prices are going down and will go down more the coming months due to oversupply [06:48:13 CET] <xiando> we're talking $30 for 240GB [06:54:15 CET] <furq> mkvmerge can split by size [06:54:25 CET] <furq> --split size:4g [06:55:02 CET] <nicolas17> I'd have done like [06:55:25 CET] <nicolas17> ffmpeg -i gyongree-171129_downpour.mp4 -t 1:00:00 -codec copy part1.mp4 [06:55:31 CET] <furq> there's no particularly nice way to do it with ffmpeg [06:55:39 CET] <furq> without working backwards from how long the file is [06:55:55 CET] <nicolas17> yeah, I'd have just split it in half by time :P [06:56:22 CET] <furq> also if you did do it with ffmpeg you should use the segment muxer [08:46:44 CET] <KombuchaKip> Could anyone with knowledge of custom AVIOContext usage kindly provide a code review? I have attempted to implement my own AVIOContext for reading media from memory instead of disk. I just want to make sure this is performed correctly. https://pastebin.com/KA7s4bJs [08:50:09 CET] <squ> what is return type of distance [08:50:58 CET] <squ> it is assigned to pointer to char in one function [08:51:05 CET] <squ> and compared to int64 in another [08:51:51 CET] <squ> if should have space [08:52:17 CET] <squ> pointer to char is compared with 0 [08:58:35 CET] <KombuchaKip> squ: Thank you. I will take a look tomorrow. [09:45:20 CET] <beauty> hello [09:45:27 CET] <beauty> can i connect [10:41:46 CET] <spaam> beauty: i think so. [12:07:39 CET] <gaara4896> quick question, how long does it take for you guys computer to compile ffmpeg library? And what is your spec? [12:16:56 CET] <beauty> two minutes [12:17:00 CET] <beauty> twenty [12:54:12 CET] <skident> Hi there [12:56:42 CET] <skident> I'm trying to use audio filters for audio stream (applying them on the fly). After using some of these filters I got cut sound (sounds like part of audio was lost). [12:57:19 CET] <skident> For example loudnorm work as I described above [12:59:41 CET] <spaam> gaara4896: ~90s on macbook pro with i9 and running with -j12 with only configure without any options. [13:09:52 CET] <gaara4896> spaam: That was fast [13:10:57 CET] <gaara4896> spaam: Do we need to specify -j12? I put only make, and it seems like it not utilizing all the core, and i have a 6 core processor too, and it took me quite a while [13:11:18 CET] <gaara4896> I read online says without putting anything will spawn unlimited worker [13:12:45 CET] <Ke> make -j is unlimited parallelism [13:12:58 CET] <TheAMM> make -j is webscale [13:13:00 CET] <Ke> I use make -j$(nproc) [13:13:29 CET] <Ke> but what is best heavily depends on circumstances [13:13:37 CET] <Ke> might be nproc + 1 [13:15:22 CET] <gaara4896> I change to just -j, and its so freaking fast now... [13:19:27 CET] <Ke> -j is known to cause problems on large compiles, like distro kernels [13:19:49 CET] <Ke> running out of memory etc. [13:22:54 CET] <gaara4896> Ke: Just some thought experiment, if you have threadripper with 4GB ram and doing -j$(nproc+1) might get you out of memory right? Haha [13:24:10 CET] <Ke> gaara4896: depends a lot on, what you are compiling [13:24:26 CET] <Ke> C++ with debug symbols is a lot heavier than C without [13:25:27 CET] <gaara4896> But not people gonna pair 32core processor with 4GB ram I guess [13:43:05 CET] <BtbN> If you have a threadripper and only install 4GB RAM you have failed setting your priorities. [13:43:32 CET] <BtbN> I usually aim for at least 4GB of RAM per core, ideally 8GB+ [14:12:06 CET] <dfhh> hello [14:12:15 CET] <dfhh> hhh [14:12:28 CET] <dfhh> can i connect [16:22:43 CET] <kishmesh> hi. when trying to parse an mp3 file I see a message "header missing" Format mp3 detected only with low score of 1, misdetection possible! [16:22:55 CET] <kishmesh> what could be wrong and how to fix this? [16:24:25 CET] <kishmesh> basically this issue https://superuser.com/questions/827970/ffmpeg-mp3-file-header-restore [16:25:58 CET] <kishmesh> als this bug it seems https://trac.ffmpeg.org/ticket/5689 [16:30:59 CET] <kishmesh> similar with the same file as m4a I get "moov atom not found" [16:43:22 CET] <limbo_> kishmesh: you're missing the metadata section of the file, with mp4 files it's at the end. Either that, or it's not actually the format you think it is. [17:05:02 CET] <kishmesh> limbo_: thank you I figured it out. it was a file format problem. [20:21:13 CET] <puchu> hi [20:21:37 CET] <puchu> how can i convert an audio file (aac 2ch 44100Hz) to aac 2ch 48000Hz [20:23:32 CET] <kepstin> puchu: in general, don't - the lossy transcode will lose quality and that sort of resampling can normally be done by the player if needed. But if you have to, you can use the "-ar" output option in ffmpeg to tell it to resample when re-encoding. [20:24:01 CET] <puchu> kepstin: i want to merge two files with mkvmerge but i get [20:24:10 CET] <puchu> The sample rate of the two audio tracks is different: 48000 and 44100 [20:24:39 CET] <puchu> kepstin: so i have to change it to get i merged. or do you have another idea to force mkv to merge it [20:25:24 CET] <puchu> kepstin and where can i find a would documentation for ffmpeg? the man page seems to miss a lot of stuff [20:25:37 CET] <puchu> like list all available codecs with options [20:27:59 CET] <puchu> ahh found it ffmpeg -codecs [20:29:45 CET] <kepstin> adjusting the sample rate to allow concatenating files is a case where it makes sense [20:30:46 CET] <kepstin> ffmpeg has a builtin aac encoder (named "aac") that works ok, but if you have a build with fdk_aac available it'll probably give better quality at the same size [20:41:16 CET] <puchu> kepstin: thanks [20:41:37 CET] <puchu> kepstin is there a way to list all available options for a specific codec? [20:43:02 CET] <kepstin> there's a bunch of generic options available that apply to many codecs, e.g. bitrate [20:43:11 CET] <kepstin> and some codec specific ones [20:43:19 CET] <kepstin> no general way to get the full list [20:43:50 CET] <kepstin> for aac, just set a bitrate (-b:a) and that's probably enough [20:44:42 CET] <puchu> sucks a bit that there is no way to get a full list [20:44:45 CET] <puchu> thanks for your help [20:45:55 CET] <kepstin> for codec specific options, use " ffmpeg -h encoder=<codecname>" [20:46:07 CET] <kepstin> for general options,... read the ffmpeg man page :/ [20:50:50 CET] <puchu> kepstin: thanks [22:16:14 CET] <No0n3Left> Anyone have suggestions on converting between h264 in a mkv and theora in an ogg? Got the audio part fine, as its just copying an opus stream from the mkv, but the video conversion is quite horrid. Not only did it take 10 minutes for a 4 minute video (which isnt horrid really), it just doesnt render correctly. There are so many visual glitches from the conversion, and lots of laggy frames. The original [22:16:22 CET] <No0n3Left> h264 was 60fps, and mpv says the theora is also 60fps, but it looks more like a 15fps minimum quality jpeg slideshow. I just used the flags -vcodec theora -acodec copy, any suggestions on how to get better quality? [22:17:38 CET] <JEEB> No0n3Left: the theora encoder is notorious for dropping frames when it can't hit your bit rate [22:20:33 CET] <nicolas17> and you didn't set a bitrate [22:20:41 CET] <nicolas17> so maybe it's doing 200kb/s or something stupid [22:20:59 CET] <JEEB> the default in AVCodecContext for bit rate is indeed 200kbps for hysterical raisins [22:21:05 CET] <No0n3Left> It is, what bitrate would you suggest? Dont know much about bitrate [22:21:15 CET] <JEEB> or you just use some quantizer [22:21:20 CET] <JEEB> if theora supports that [22:28:10 CET] <dualz> Using anullsrc and testsrc as my inputs I'm getting very high FPS in my output. I set rate on testsrc to 30 and still no avail [22:28:13 CET] <dualz> https://gist.github.com/ryanmarin/7adbfec209c359296eb5dbfb4df96f9e [23:19:21 CET] <kepstin> yeah, just don't use theora if at all possible [23:19:40 CET] <kepstin> pretty much no reason to nowadays, vp8 (or even 9) is almost always a better choice [00:00:00 CET] --- Tue Nov 6 2018

1 0

ffmpeg-devel.log.20181104
by burek 05 Nov '18

05 Nov '18

[00:09:07 CET] <jamrial> philipl: minor in release branches can't be bumped, since it would clash with master [00:10:31 CET] <jamrial> they have their own exclusive major.minor version that can't be touched. micro is free to use [00:11:02 CET] <JEEB> yeh [00:11:33 CET] <jamrial> michaelni: about the above, maybe point releases should bump micro across the board [00:14:31 CET] <jamrial> as is, there's no clear way to tell the libraries apart between 4.0.2 and 4.0.3, for example [00:14:56 CET] <jamrial> and with the amount of bugfixes each point release gets, it may be a good idea to implement [00:16:58 CET] <nevcairiel> unless that results in an actual functional behavior, in w hich case the change in question should've had a bump, does anyone really care? [00:17:06 CET] <nevcairiel> not t hat bumping hurts anything [00:24:54 CET] <philipl> jamrial: I think it generally makes sense. It basically means treating the so versions like release versions. They always increment on a new release. [00:26:47 CET] <jamrial> philipl: that's what they do. minor is bumped at release time, so the major.minor combination in a release is not used in master [00:27:08 CET] <jamrial> my suggestion is doing the same for point releases in a branch [00:27:32 CET] <philipl> K. So yes, I think it's a good idea :-) [01:05:00 CET] <nicolas17> bah, I wanted to cleanup some code but there's no tests for the functionality [01:06:08 CET] <philipl> If nothing can say you got it wrong, can it really be wrong? [01:06:56 CET] <nicolas17> I mean, I tried removing the feature entirely (support for .Y .U .V files in image2) and all tests passed :P [01:12:43 CET] <philipl> heh [12:54:08 CET] <cone-653> ffmpeg 03Michael Niedermayer 07master:df91af140c55: avcodec/mpegaudio_parser: Consume more than 0 bytes in case of the unsupported mp3adu case [12:54:08 CET] <cone-653> ffmpeg 03Michael Niedermayer 07master:470700bc96c3: avcodec/pictordec: Error out if more than one plane is unused [15:02:50 CET] <cone-653> ffmpeg 03Paul B Mahol 07master:63fbec747753: avfilter/af_afade: remove unused code [19:27:02 CET] <durandal_1707> is there any audio samples available for matching/aligning? [20:22:42 CET] <cone-223> ffmpeg 03Michael Niedermayer 07master:074187d599a2: avcodec/mpeg4videodec: Clear partitioned frame in decode_studio_vop_header() [20:22:42 CET] <cone-223> ffmpeg 03Michael Niedermayer 07master:c1cee0565692: avcodec/cavsdec: Propagate error codes inside decode_mb_i() [20:58:06 CET] <cone-223> ffmpeg 03Michael Niedermayer 07release/4.1:b9875b75837d: avcodec/prosumer: Check for bytestream eof in decompress() [20:58:07 CET] <cone-223> ffmpeg 03Michael Niedermayer 07release/4.1:a163384467dd: avcodec/prosumer: Remove unneeded () [20:58:08 CET] <cone-223> ffmpeg 03Michael Niedermayer 07release/4.1:fd05e20650a5: avcodec/prosumer: Remove always true check in decompress() [20:58:09 CET] <cone-223> ffmpeg 03Michael Niedermayer 07release/4.1:2f04b78b9546: avcodec/prosumer: Simplify bit juggling of the c variable in decompress() [20:58:10 CET] <cone-223> ffmpeg 03Michael Niedermayer 07release/4.1:7d23ccac8d37: avcodec/mpegaudio_parser: Consume more than 0 bytes in case of the unsupported mp3adu case [20:58:11 CET] <cone-223> ffmpeg 03Michael Niedermayer 07release/4.1:bc5777bdab7d: avcodec/mpeg4videodec: Clear partitioned frame in decode_studio_vop_header() [20:58:12 CET] <cone-223> ffmpeg 03Michael Niedermayer 07release/4.1:7ebc27e1fa73: avcodec/cavsdec: Propagate error codes inside decode_mb_i() [21:15:28 CET] <jamrial> michaelni: please don't tag 4.1 yet, i sent a fix for cbs_av1 that needs to be applied first [21:17:37 CET] <jamrial> jkqxz: can you push your fix for redundant header obus? [21:42:06 CET] <michaelni> jamrial, ok [22:54:52 CET] <cone-223> ffmpeg 03Marton Balint 07master:8e5a2495a8da: avformat/ftp: allow nonstandard 202 reply to OPTS UTF8 [23:04:07 CET] <cone-223> ffmpeg 03Marton Balint 07release/4.1:63c1e291ef7b: avformat/ftp: allow nonstandard 202 reply to OPTS UTF8 [00:00:00 CET] --- Mon Nov 5 2018

1 0

ffmpeg.log.20181104
by burek 05 Nov '18

05 Nov '18

[00:00:03 CET] <M6HZ> The first option seems to be wrong: -bsf:v 'extract_extradata,remove_extra=freq=all' doesn't delete them [00:00:25 CET] <M6HZ> And I have troubles with this one: -bsf:v 'h264_metadata=delete_filler' [00:00:38 CET] <M6HZ> Is something missing to activate the filter ? [00:01:06 CET] <JEEB> see the filter_units bsf [00:01:14 CET] <JEEB> that most likely is what you want, it even has examples [00:01:24 CET] <JEEB> just check which NAL unit type the captions are in [00:01:28 CET] <JEEB> and that should be it [00:18:04 CET] <M6HZ> JEEB, it looks like this works without having to re-encode :) : ffmpeg -i file.ts -c:v copy -bsf:v 'filter_units=remove_types=6' -f matroska - | mpv - [00:20:06 CET] <M6HZ> By the way, "just check which NAL unit type the captions are in". Do you know how can I figure this out ? I have found this website, which has been really helpful but it doesn't look very reliable https://fbookmedia.files.wordpress.com/2017/06/facebook-live-closed-caption… [00:21:50 CET] <JEEB> there's a bsf that dumps info on all the packets there [00:22:02 CET] <JEEB> yes, trace_headers [00:22:08 CET] <M6HZ> ok. great. [00:22:48 CET] <JEEB> I highly expect it to be one of the user data ones [00:24:28 CET] <JEEB> and yes, NAL unit type 6 is SEI, btw :P [00:24:41 CET] <JEEB> subtype is probably either 4 or 5 [00:26:28 CET] <M6HZ> according to the website I've linked it would be 4 :) [00:38:07 CET] <M6HZ> JEEB, Ahem. How could you find out which one is the subtitle unit in all the garbage reported by "trace_headers" ? [00:43:32 CET] <M6HZ> Acutally, if you know what you are searching, then you can look at "Supplemental Enhancement Information" and you get "3 nal_unit_type 00110 = 6" [00:43:52 CET] <M6HZ> actually * [00:44:30 CET] <M6HZ> That's a really cryptic output x) [00:45:21 CET] <JEEB> basically it outputs the structure of the packets in the format [00:45:43 CET] <JEEB> and yes, nal_unit_type is the type of the NAL unit (AVC and HEVC are NAL unit based formats) [00:47:19 CET] <JEEB> and yes, looking at the cbs_h264_syntax_template.c "Supplemental Enhancement Information" is the header printed out for SEIs [00:47:51 CET] <JEEB> then it should print out something for the payload as well, since it parses the the payloads as well [00:48:24 CET] <JEEB> it should print out the country code, the extension byte if that's there, and then the payload [00:49:10 CET] <M6HZ> JEEB, it's even more cryptic [00:49:18 CET] <M6HZ> I get things like that: [00:49:21 CET] <M6HZ> [AVBSFContext @ 0x556284646340] 8 last_payload_type_byte 00000100 = 4 [00:49:21 CET] <M6HZ> [AVBSFContext @ 0x556284646340] 16 last_payload_size_byte 01100101 = 101 [00:49:21 CET] <M6HZ> [AVBSFContext @ 0x556284646340] 24 itu_t_t35_country_code 10110101 = 181 [00:49:21 CET] <M6HZ> [AVBSFContext @ 0x556284646340] 32 itu_t_t35_payload_byte 00000000 = 0 [00:49:26 CET] <M6HZ> [AVBSFContext @ 0x556284646340] 40 itu_t_t35_payload_byte 00110001 = 49 [00:49:58 CET] <JEEB> prefer a pastebin site for anything more than one or two lines [00:50:05 CET] <M6HZ> ok [00:50:06 CET] <JEEB> but anyways, that's the type there first [00:50:16 CET] <JEEB> then the payload size [00:50:34 CET] <JEEB> and then you see the t35 values from inside the registered user data sei [01:19:41 CET] <M6HZ> https://www.itu.int/dms_pub/itu-t/opb/sp/T-SP-T.35-2012-OAS-PDF-E.pdf [01:19:48 CET] <M6HZ> JEEB, very interesting [01:20:25 CET] <JEEB> yes [01:20:33 CET] <JEEB> that's the country codes [01:21:46 CET] <M6HZ> Thanks JEEB and nicolas17 for your help. JEEB you seem to have an extensive knowledge about this software :) [01:22:30 CET] <JEEB> I both hate and love this mess of software, and I'm a maintainer at both FFmpeg and mpv :P [01:22:30 CET] <TheAMM> Must be a fan, huh [01:23:03 CET] <M6HZ> JEEB, alright ;) [01:23:23 CET] <JEEB> although I'm mostly just a masochist on the user support channels [01:23:30 CET] <JEEB> otherwise just herping a derp elsewhere on IRC [01:23:45 CET] <nicolas17> in my experience, diving into the source code is more masochistic [01:24:13 CET] <JEEB> depends, you can grep the logs for me exploding here :P [01:24:25 CET] <JEEB> although sometimes I also nowadays leave the channel for a while [01:25:27 CET] <JEEB> nicolas17: but I do agree that I wish I knew less about the insides of discontinuity handling and "vsync" code in ffmpeg.c [01:26:41 CET] <nicolas17> JEEB: diving into the code is how I found this stupid inconsistency https://paste.kde.org/pbemjvxv7 [01:27:14 CET] <nicolas17> if frame09.png is missing, it processes from 1 to 8; if frame10.png is missing, it tries to process 1 to 20 and gives an error when it can't open 10 [01:27:39 CET] <JEEB> oh, the Very Specific modules that someone made for their little use case [01:27:41 CET] <JEEB> yes [01:28:29 CET] <nicolas17> image sequences is very specific and little use case? :P [01:29:02 CET] <JEEB> no, but the module was clearly made by someone for his own little specific use case [01:29:12 CET] <JEEB> just like we have I think like four things to do concatenation [01:29:19 CET] <JEEB> protocol, demuxer, filter? [01:29:40 CET] <nicolas17> https://paste.kde.org/p6ql23grx here's the underlying cause... it uses exponential steps to find the last file instead of checking every file [01:29:59 CET] <nicolas17> so gaps are handled inconsistently depending on if the search happens to check it or not [01:33:16 CET] <Zexaron> A bit old but this is interesting https://twitter.com/ID_AA_Carmack/status/1052237045526089729 [02:30:36 CET] <gaara4896> Was wondering in FFMPEG configure, the --extra-libs=ELIBS add ELIBS [$ELIBS], what is the exact meaning of ELIBS? I can't find the meaning of it [02:34:35 CET] <furq> gaara4896: are you asking where $ELIBS comes from or what you're supposed to put there [02:34:52 CET] <furq> if it's the former then it's just a user set env var [02:34:59 CET] <furq> if it's the latter then it'd be like -lfoo -lbar [02:39:34 CET] <gaara4896> I believe it would be something like libmathlib.so will become -lmathlib right? For adding third party library, is placing the .so file in /lib/ and define --extra-cflags=-Iffmpeg/include and --extra-ldflags=-Lffmpeg/lib sufficient? I tried doing that, but when configure, it always fail and the tail of config.log is always "cannot open shared object file" [02:41:03 CET] <JEEB> yes, extra-ldflags or you make a pc file for your thing and make configure pick it up (and use PKG_CONFIG_PATH env var to point towards the directory it's in) [02:41:47 CET] <JEEB> also of course make sure the solib is for the arch you're building FFmpeg for :P [02:42:17 CET] <JEEB> also generally speaking I wouldn't put my extra dependencies sysroot within the FFmpeg build directory [02:47:49 CET] <gaara4896> What is the different between --extra-ldflags and --libdir? They both seems to do the same thing to me, which is defining the path to the directory where all the solib is placed [02:48:09 CET] <JEEB> extra-ldflags by default is not set [02:48:18 CET] <JEEB> libdir is the directory where libraries installed to [02:48:27 CET] <JEEB> by default it's --prefix + /lib [02:48:38 CET] <JEEB> just like bindir is by default prefix + /bin [02:49:06 CET] <JEEB> I generally for stuff I build locally use a custom prefix like /home/jeeb/ownapps [02:49:28 CET] <JEEB> (by default in many projects /usr/local) [02:49:45 CET] <JEEB> but yea, extra-XXXflags is for building/linking [02:49:58 CET] <JEEB> while libdir is an output (install) parameter for the stuff you built [02:50:00 CET] <JEEB> :P [02:51:45 CET] <JEEB> see ../configure --help |grep libdir [02:51:59 CET] <JEEB> or just one dot if you're in the same dir as the FFmpeg configure script [03:34:14 CET] <gaara4896> The problem I am facing is I had define --extra-ldflags=-L/<fullpath>/ffmpeg/lib, place the .so file inside the ffmpeg/lib, and define --extra-libs=-l<libname>, and also --enable-rpath, however I am still getting "cannot open shared object file: No such file or directory" in the tail of the error, why does it not reference the .so file to the ffmpeg/lib directory? [03:34:34 CET] <gaara4896> The problem occur when I am doing ./configure, not during make [06:37:18 CET] <gaara4896> The problem I am facing is I had define --extra-ldflags=-L/<fullpath>/ffmpeg/lib, place the .so file inside the ffmpeg/lib, and define --extra-libs=-l<libname>, and also --enable-rpath, however I am still getting "cannot open shared object file: No such file or directory" in the tail of the error, why does it not reference the .so file to the ffmpeg/lib directory? [06:37:23 CET] <gaara4896> The problem occur when I am doing ./configure, not during make [10:23:20 CET] <gaara4896> The problem I am facing is I had define --extra-ldflags=-L/<fullpath>/ffmpeg/lib, place the .so file inside the ffmpeg/lib, and define --extra-libs=-l<libname>, and also --enable-rpath, however I am still getting "cannot open shared object file: No such file or directory" in the tail of the error, why does it not reference the .so file to the ffmpeg/lib directory? [10:48:07 CET] <JEEB> because whatever the result is you're getting isn't finding whatever you're needing? :P [10:49:03 CET] <JEEB> if you are not getting through the configure phase then check ffbuild/config.log for the exact command that you can try then running manually. otherwise `make V=1` is what during running the Makefile outputs the full commands [10:49:16 CET] <JEEB> so you can see exactly how the output looks and if it contains what you think it does [10:50:14 CET] <gaara4896> the `make V=1` is for making stage is it? [10:51:32 CET] <JEEB> whatever that means [10:52:08 CET] <illuminated> is ffmpeg coded in c? [10:52:31 CET] <JEEB> yes [10:52:47 CET] <JEEB> there's very few C++ files explicitly for libraries that require C++ interfaces [10:53:54 CET] <illuminated> ok, and what is handbrake coded in? [10:54:06 CET] <JEEB> I have no idea [10:54:08 CET] <illuminated> I understand it's kinda just a gui frontend to ffmpeg [10:54:23 CET] <JEEB> it is an API client for FFmpeg, and then on top of that it has the front-end [10:54:48 CET] <illuminated> I'm learning programming [10:54:56 CET] <illuminated> and I'm looking for projects [10:55:13 CET] <illuminated> thought that writing a simple frontend to ffmpeg would be an interesting project [10:55:22 CET] <gaara4896> I tried to put the last command in the config.log into my console and it works lol, and able to get -l<libname> wihtout a problem, but it get error when doing ./configure <param...> [10:55:38 CET] <illuminated> anyway thanks [10:56:49 CET] <JEEB> gaara4896: so you're not getting further than configure? [10:57:17 CET] <JEEB> gaara4896: then find what command fails ffbuild/config.log [10:57:57 CET] <gaara4896> JEEB: The failing comamnd in config.log works when I run it seperately, yet fail when inside .configure [10:58:52 CET] <gaara4896> https://pastebin.com/6Uwk87bZ, what is the few thing that looks like hash value .c? [10:59:08 CET] <gaara4896> like /tmp/ffconf.R74iH7Iv.c [10:59:12 CET] <JEEB> that's the temporary .c file's name [10:59:21 CET] <JEEB> the script uses mktemp or something to generate them [10:59:49 CET] <JEEB> in this case it's a basic check if you can compile a file that contains a simple main that returns 0 [11:01:24 CET] <gaara4896> I see, no wonder it says C compiler fail test [19:08:42 CET] <a__pi> I'm trying to download something rtsp with Transport: MP2T/H2221/UDP. I tried -rtsp_transport udp but it's not sending this type of transport. Is it possible to use this type with ffmpeg ? [23:06:26 CET] <limbo_> Is there some kind of CLI video encoder interface with a lot of presets for what can play on different platforms, and speed/quality? e.g. encodething -i file.webm --preset ps4 -q high file.mp4 Like handbrake, but faster + CLI. [23:07:49 CET] <ritsuka> handbrake uses the same encodes as all the other open sources programs& so the speed will be mostly the same, and there is a handbrakecli version too [23:40:54 CET] <limbo_> interface is pretty bad though. compared to other linux things. [23:42:43 CET] <furq> do you really need more than one preset these days [23:42:54 CET] <furq> unless it's for 4k, basically everything wants h264 and aac in mp4 [23:43:23 CET] <nicolas17> the only time I ever had to change something for compatibility was "-profile main" for a crappy TV [23:43:34 CET] <furq> yeah the days of using baseline and main are more or less over [23:43:43 CET] <furq> the only real consideration is that you might need to set level 4.1 for hwdec [23:44:00 CET] <nicolas17> furq: I mean in the days of iPod it was more complicated than just main vs high [23:44:30 CET] <furq> ipod video was baseline at a specific resolution wasn't it [23:45:52 CET] <furq> but yeah more or less everything from this decade supports high at 4.1 [00:00:00 CET] --- Mon Nov 5 2018

1 0

ffmpeg-devel.log.20181103
by burek 04 Nov '18

04 Nov '18

[01:17:09 CET] <cone-671> ffmpeg 03Michael Niedermayer 07release/4.0:d2ce6472a77a: avcodec/msrle: Check that the input is large enough to contain a end of picture code [01:17:10 CET] <cone-671> ffmpeg 03Michael Niedermayer 07release/4.0:b8aa7b9a6d86: avutil/integer: Fix integer overflow in av_mul_i() [01:17:11 CET] <cone-671> ffmpeg 03Michael Niedermayer 07release/4.0:fcbd117df307: Changelog: update [01:50:03 CET] <cone-671> ffmpeg 03Michael Niedermayer 07n4.0.3:HEAD: avcodec/prosumer: Simplify bit juggling of the c variable in decompress() [04:20:32 CET] <cone-671> ffmpeg 03James Almer 07release/4.0:dee8f4b01bfe: avcodec/libaomenc: remove AVOption related to frame partitions [11:12:03 CET] <durandal_1707> who put yadif_cuda filter in wrong section of Changelog? [13:01:01 CET] <atomnuker> BBB: wrote checkasm, how do I get the bench time? [13:31:08 CET] <BBB> bench [13:31:15 CET] <BBB> test=<yourtestname> --bench [13:31:27 CET] <BBB> e.g. test=looprestoration_8bpc bench [13:31:49 CET] <BBB> where the name is the one from the table in checkasm.c [13:32:11 CET] <BBB> and then itll report everything that has a report() line under that function cll [13:34:05 CET] <atomnuker> meson test -C build -v test=looprestoration_8bpc bench [13:34:09 CET] <atomnuker> no suitable tests found [13:34:50 CET] <atomnuker> then what's the string in check_func() used for if only the one in report() is used? [13:35:09 CET] <atomnuker> you can give arguments to it, like cdf size, unlike in report [13:35:19 CET] <BBB> ninja && tests/checkasm test=looprestoration_8bpc bench [13:35:41 CET] <BBB> sorry, youre right [13:35:49 CET] <BBB> check_func() is for the per-function perf benchmark (--bench) [13:36:04 CET] <BBB> the report() is only to validate that all tests in that group passed [13:36:30 CET] <BBB> and checkasm.c is for test=<name> selection [13:36:50 CET] <atomnuker> zsh: permission denied: tests/checkasm [13:37:09 CET] <atomnuker> oh damn, forgot to append build dir [13:37:16 CET] <BBB> :) [13:37:41 CET] <atomnuker> did I mention I hate meson and like obscure long bash scripts? [13:37:48 CET] <BBB> o_O [13:37:55 CET] <atomnuker> seems like every project but the simplest that uses meson is fighting against it [13:37:57 CET] <BBB> I guess you did now :) [13:38:25 CET] <atomnuker> people constantly need to add features to fix bugs, do workarounds to make it work [13:38:25 CET] <BBB> j-b selected the build system and he liked this. Im fine with it, it does what it needs to do [13:38:34 CET] <atomnuker> and at the end of the day there's no make backend [13:38:45 CET] <atomnuker> why? make is fast, everyone has it, ninja is some google thing [13:39:06 CET] <atomnuker> but they made it a principle not to have make support [13:39:15 CET] <BBB> I dont care, I really dont. I dont care for build systems. I care for C code and assembly, and algorithms for video compression etc. [13:39:35 CET] <atomnuker> yeah, I know, this is on the side of usability and extendability [13:39:37 CET] <BBB> I ask dumb questions aboutt he build system all the time, and ePirat is knowledgable and answers all my questions [13:39:41 CET] <BBB> so it works fine for me [13:40:05 CET] <BBB> its better than ffmpegs configure, where nobodoy knows how it works, except mru who isnt here and diego whos doing some fork [13:40:38 CET] <BBB> usability o_O whats that? :-p [13:40:48 CET] <BBB> anyway, does checkasm work now? [13:41:07 CET] <atomnuker> configure.sh is alright, I can sort of understand what it does enough to work with it [13:41:08 CET] <BBB> dont forget to build in release mode [13:41:17 CET] <atomnuker> and its also faster than meson and doesn't require ninja [13:41:24 CET] <BBB> (meson configure --buildtype=release -Db_sanitize=none && ninja clean && ninja) [13:41:24 CET] <atomnuker> granted, it doesn't support msvc though [13:41:41 CET] <atomnuker> can't I use debugoptimized? [13:41:43 CET] <JEEB> msvc it does, it doesn't support msvs [13:41:51 CET] <JEEB> otherwise we wouldn't support cl.exe [13:42:17 CET] <BBB> debugoptimized is probably ok, yes [13:42:17 CET] <JEEB> (MSVS is the GUI/IDE, MSVC is the C compiler) [13:42:25 CET] <BBB> release is slightlybetter but itll only be off by a few % [13:42:32 CET] <BBB> dont use debug, its like -O0, results will be useless [13:42:37 CET] <atomnuker> BBB: running the checkasm script and it prints exactly what meson test outputs [13:42:47 CET] <atomnuker> debugoptimized isn't release + debug symbols? [13:43:02 CET] <BBB> I think one is O2 and one is O3? [13:43:17 CET] <BBB> $ tests/checkasm --test=looprestoration_8bpc [13:43:17 CET] <BBB> checkasm: using random seed 4273947412 [13:43:19 CET] <BBB> AVX2: [13:43:20 CET] <BBB> - looprestoration_8bpc.wiener [OK] [13:43:21 CET] <BBB> - looprestoration_8bpc.sgr [OK] [13:43:22 CET] <BBB> checkasm: all 5 tests passed [13:43:24 CET] <BBB> you dont see that? [13:43:40 CET] <atomnuker> nope, I see checkasm: all 1068 tests passed [13:43:53 CET] <atomnuker> ./build/tests/checkasm test=looprestoration_8bpc bench [13:44:03 CET] <BBB> did you copypaste my dashes? [13:44:17 CET] <BBB> my irc client converts them from a real dash to some corrupt counterpart [13:44:22 CET] <BBB> which isnt an actual dash [13:44:24 CET] <atomnuker> ...I did [13:44:30 CET] <BBB> really [13:44:33 CET] <BBB> and still nothing? [13:44:44 CET] <atomnuker> no, I mean I copied them and that's why it didn't work [13:44:47 CET] <BBB> :) [13:44:54 CET] <BBB> sorry, my irc client is & dumb [13:45:07 CET] <BBB> - - [13:45:10 CET] <atomnuker> your dots and quote marks are visibly different here, but dashes not so much [13:52:55 CET] <BBB> you know we have a irc channel for this where other people can also help with these questions, right? just sayin. you dont have to be there if you dont want to, but youd get answers quicker [13:56:28 CET] <atomnuker> yeah, I know, I'm in no hurry and I don't know what else to do after the ec stuff is done [13:58:47 CET] <BBB> theres a long todo list of potential funny optimizations on the wiki [13:58:58 CET] <BBB> https://code.videolan.org/videolan/dav1d/wikis/task-list [13:59:14 CET] <BBB> youd maybe be interested in the more technical ones? [13:59:35 CET] <BBB> e.g. the palette index context optimizations, the hi/lo coef coding context optimizations [13:59:44 CET] <BBB> esp. the second could be pretty big if we do it right [13:59:49 CET] <BBB> brb [16:11:09 CET] <ePirat> atomnuker, yes meson has still some rough edges and lacking support of some things in some areas but it's improving [16:55:44 CET] <cone-077> ffmpeg 03Josh de Kock 07master:8096f52049ac: fate/api-h264-slice-test: don't use ssize_t [16:55:44 CET] <cone-077> ffmpeg 03Josh de Kock 07master:1052578dadf7: fate/api-h264-slice-test: use cleaner error handling [16:58:11 CET] <nevcairiel> January: can you push that to 4.1 as well? [16:58:44 CET] <cone-077> ffmpeg 03Josh de Kock 07release/4.1:5060a615c738: fate/api-h264-slice-test: don't use ssize_t [16:58:45 CET] <cone-077> ffmpeg 03Josh de Kock 07release/4.1:765fb1f224f6: fate/api-h264-slice-test: use cleaner error handling [17:01:50 CET] <jamrial> nevcairiel: why didn't the ssize_t error happen in your fate clients? [17:01:55 CET] <nevcairiel> it does [17:02:10 CET] <nevcairiel> but compile errors in test programs dont get recognized by fate as a failure for some reason [17:02:25 CET] <jamrial> ah [17:02:37 CET] <jamrial> yeah, i've seen that happen before with other tests [17:02:40 CET] <nevcairiel> fatebeta shows the stations as yellow [17:02:42 CET] <jamrial> it's really annoying [17:02:47 CET] <nevcairiel> but for some reason fatebeta was never finished [17:03:27 CET] <jamrial> it's yellow but still says "3660 / 3660" [17:03:33 CET] <jamrial> so not perfect [17:03:41 CET] <nevcairiel> yeah but at l east it indicates something is wrong [18:12:00 CET] <gaara4896> Hello, I was trying to compile ffmpeg, and I face this problem. The config.log looks something like this: https://pastebin.com/DUzgv9W6, but libstegolib.so did exists inside the folder ffmpeg/lib/libstegolib.so, anyone have any idea what might be the cause? [18:12:24 CET] <JEEB> that is not related to upstream as I already told you on #ffmpeg [18:12:45 CET] <JEEB> please do not spam and this channel is specific to FFmpeg development [18:25:36 CET] <cone-077> ffmpeg 03Paul B Mahol 07master:639694eaa554: avfilter/vf_extractplanes: use ff_outlink_get_status() [18:25:37 CET] <cone-077> ffmpeg 03Paul B Mahol 07master:decc008929e3: avfilter/af_amultiply: make use of ff_inlink_queued_samples() [18:25:38 CET] <cone-077> ffmpeg 03Paul B Mahol 07master:4b82b7104120: avfilter/af_amerge: use ff_inlink_queued_samples() [18:25:39 CET] <cone-077> ffmpeg 03Paul B Mahol 07master:c8625e5c6f50: avfilter/af_afade: do not use framequeue directly [18:25:40 CET] <cone-077> ffmpeg 03Paul B Mahol 07master:242f7bf20254: avfilter/split: use ff_outlink_get_status() [18:25:51 CET] <JEEB> -34 [20:36:42 CET] <durandal_1707> SSSSSSSSSSSSSSSPPPPPPPPPPPPPPPPPPAAAAAAAAAAAAAAAAAAAAAAAAAAAAMMMMMMMMMMMMMMMMMMMMMMMMMMMMMM [20:37:53 CET] <atomnuker> where? [20:38:43 CET] <durandal_1707> philipl: did you put cuda yadif filter in wrong section of Changelog? [20:42:38 CET] <durandal_1707> atomnuker: spam of silence [20:50:16 CET] <atomnuker> it is saturday after all [20:51:31 CET] <atomnuker> I have learned to appreciate silence more over the past few days [20:51:47 CET] <atomnuker> there has been constant bombardment here over the past week or more here [20:52:54 CET] <atomnuker> fireworks, explosions, around the evenings, some foreign holiday or something which last a whole 2 weeks or so [20:53:27 CET] <atomnuker> oh and a fucking bird that sings for 5 minutes at 06:36 every morning, wakes me up, and fucks off a few minutes later [20:56:38 CET] <philipl> durandal_1707: did i? [20:58:41 CET] <Compn> lol atomnuker [20:58:47 CET] <Compn> 6:36 bird [21:05:21 CET] <philipl> i thought 4.0 had been cut already. is it not? [21:05:40 CET] <nevcairiel> 4.1 has been cut yes [21:06:04 CET] <nevcairiel> but thats why its in the wrong section [21:06:08 CET] <nevcairiel> because its not in 4.1 =p [21:06:09 CET] <atomnuker> Compn: yes, it thinks its spring, the small idiotic creature, and thus sings as if its the breeding season [21:06:13 CET] <philipl> oh. [21:06:26 CET] <nevcairiel> generally, you always put it into the top section, it gets updated on cuts [21:06:26 CET] <atomnuker> its not as bad as spring though, fuck spring, multiple ones all at once at 05:20 [21:06:48 CET] <philipl> well, that's easy to fix. or do you want me to cherry pick it over? [21:07:01 CET] <nevcairiel> i wouldnt mind having it in 4.1 [21:07:06 CET] <nevcairiel> its not like 4.1 is frozen yet [21:07:17 CET] <philipl> i missed it because when i started, 4.1 hadn't been cut yet. [21:07:33 CET] <philipl> i'll cherry-pick it over later today. [23:46:42 CET] <philipl> nevcairiel: So what do I do about the version.h when porting to the branch? [23:46:59 CET] <philipl> I have a minor bump because of the new filter. Do I just leave it untouched? Increase the micro version instead? [23:47:38 CET] <nevcairiel> do a micro i guess [23:49:32 CET] <philipl> Will do [23:55:18 CET] <cone-077> ffmpeg 03Philip Langdale 07release/4.1:041231fcd632: libavfilter/vf_yadif: Make frame management logic and options shareable [23:55:19 CET] <cone-077> ffmpeg 03Philip Langdale 07release/4.1:67126555fc03: avfilter/vf_yadif_cuda: CUDA accelerated yadif deinterlacer [23:55:20 CET] <cone-077> ffmpeg 03Philip Langdale 07release/4.1:6feec11e489b: avcodec/nvdec: Increase frame pool size to help deinterlacing [23:55:21 CET] <cone-077> ffmpeg 03Philip Langdale 07release/4.1:ebc1c49e417c: avfilter/vf_cuda_yadif: Avoid new syntax for vector initialisation [00:00:00 CET] --- Sun Nov 4 2018

1 0

ffmpeg.log.20181103
by burek 04 Nov '18

04 Nov '18

[00:54:23 CET] <brimestone> Is there an ffmpeg/ffplay wrapper for Swift4 or objective c? [04:44:34 CET] <grosso> hi [04:45:12 CET] <grosso> i'm using ffmpeg to receive a rtsp stream, like this: ffmpeg -i 'rtsp://192.168.1.44:8559/live.sdp -c copy output.mkv [04:46:25 CET] <grosso> the problem is that ffmpeg actually don't save the file until the streaming is stopped from the sending end... it actually don't do anything until streaming is stopped [04:47:38 CET] <nicolas17> that seems unlikely... where would all the stream be stored meanwhile? in RAM? [04:49:04 CET] <grosso> I tryed with several different options: to save a file, to save a sequence of pictures, using segments, re-streaming through udp, etc etc etc... in all cases, it just receives the stream (I see that at the sender end) but don't do anything more until streaming stops [04:49:24 CET] <grosso> yes! it stores it in RAM!! [04:51:45 CET] <grosso> nicolas17... it's so strange.. I can't make it do anything until stop sending stream... then it saves the files from what I think has to be a large RAM buffer [04:52:22 CET] <pink_mist> have you tried streaming so much that it won't fit in RAM? [04:53:20 CET] <nicolas17> how big is the file when you stop it? [04:54:28 CET] <grosso> well, i do no tryed beyond a minute or two... when working with sequences of pictures, it can deliver 1000 pictures from ram [04:55:00 CET] <grosso> that actually is 40 seconds [04:56:08 CET] <grosso> what I think it behaves like the imput was actually a file, rather than a live streaming [04:56:34 CET] <nicolas17> file size I mean [04:58:14 CET] <grosso> file size is easy 4mBytes [04:58:34 CET] <grosso> i don't stressed out [04:58:43 CET] <grosso> just a minute or so [04:59:07 CET] <nicolas17> it can buffer data before writing to output file but 4MB seems too much... [04:59:35 CET] <grosso> it does it.. [05:01:33 CET] <grosso> it don't do anything until streaming stops.. if it could buffer 100MB it will... don't do anything until streaming stops from sender [05:01:55 CET] <grosso> that's the problem [05:02:26 CET] <grosso> I can't make it behaves like if a real live streaming [05:03:09 CET] <grosso> it just behaves like if the input was a file [05:04:15 CET] <nicolas17> input files don't behave like that either [05:04:22 CET] <nicolas17> buffering the entire video in memory makes no sense [05:04:26 CET] <nicolas17> try for longer time? [05:04:53 CET] <grosso> ok, I can try for longer time [05:05:03 CET] <grosso> just wait a moment [05:09:11 CET] <grosso> ok, I just started now [05:09:21 CET] <grosso> tell me when to stop [05:10:05 CET] <nicolas17> maybe like 2 minutes to ensure it exceeds 4MB? [05:10:40 CET] <grosso> ok [05:10:49 CET] <grosso> it is running now [05:11:51 CET] <grosso> maybe it just buffer the last 4mb of video...what is for sure is that it don't saved any file so far [05:12:21 CET] <grosso> so, I stop it? [05:12:42 CET] <furq> grosso: try setting -rtbufsize [05:14:00 CET] <grosso> 8MB [05:15:54 CET] <grosso> rather than limit buffer size, i would like to force it to save files before stream stops [18:08:46 CET] <gaara4896> Hello, I was trying to compile ffmpeg, and I face this problem. The config.log looks something like this: https://pastebin.com/DUzgv9W6, but libstegolib.so did exists inside the folder [18:08:50 CET] <gaara4896> ffmpeg/lib/libstegolib.so, anyone have any idea what might be the cause? [18:09:27 CET] <JEEB> not upstream stuff [18:09:46 CET] <JEEB> please take this to the developers of this specific project [18:10:01 CET] <JEEB> since stegolib is not part of FFmpeg nor is what you're building part of the standard FFmpeg [18:10:27 CET] <JEEB> but what it sounds is that you don't have that stegolib thing in your linker search path [18:10:42 CET] <JEEB> good luck with trying to get support from that project [18:10:46 CET] <gaara4896> I actually wanted to ask CMake or Make specific problem, not ffmpeg, wondering what IRC channel is related to it? [18:13:30 CET] <JEEB> gaara4896: no idea. your best bet is to just ask for guidance with that project you are trying to build [18:17:24 CET] <pink_mist> gaara4896: /msg alis help [18:17:35 CET] <pink_mist> gaara4896: alis is the freenode bot for channel listing and searching [19:09:41 CET] <Filarius> hello, would somebody say me is there will be really big difference for encoding time if I take same raw data file and encode it as 1080p source or 4k source, or 480p source ? [19:10:07 CET] <Filarius> with x264 codec [19:11:18 CET] <furq> yes [19:12:45 CET] <tdr> more data or less data to encode = more or less work to do [19:14:06 CET] <Filarius> just for sure... input file is like just 1Gb raw data( always same for all setings), and only what I change will be making ffmpeg think what resolution it could be, with common settings on x264 [19:14:54 CET] <Filarius> yes, result file will be different at frame count, but in this I only want know what will be with time spent on encoding [19:15:39 CET] <Filarius> furq, and where it will be faster ? [19:17:57 CET] <Filarius> I know I ask strang thing, but I talk about it with buddy and its about using ffmpeg in data procedding [19:18:00 CET] <Filarius> *processing [19:27:30 CET] <ChocolateArmpits> Filarius, in theory then it shouldn't [19:27:51 CET] <ChocolateArmpits> unless there's something memory related due to bigger frames [19:28:56 CET] <ChocolateArmpits> Filarius, you may have to pad it if the data length doesn't stricly match the input settings you will use [19:29:27 CET] <Filarius> yep, padding added [19:30:12 CET] <Filarius> just there some stranger found my old "data to video" project so we came to this question [19:31:26 CET] <ChocolateArmpits> I think there was someone else some time ago who wanted to also lossy compress data as well [19:32:11 CET] <Filarius> a sent days and nights making DCT data coding as I see simple squares is too stupid ideas, and that dude say me what he prefer version what pack data into b/w squares ! [19:32:27 CET] <Filarius> ***i spent days and nights [19:33:22 CET] <Filarius> maybe I was here with same question [19:33:32 CET] <Filarius> 2-3 years ago [19:33:43 CET] <ChocolateArmpits> nah it was a month or so ago [19:34:00 CET] <ChocolateArmpits> definitely in the last few months [19:37:34 CET] <Filarius> well, I have 2 y.o. project on GitHub already made it with Integer Cosinus Transform "stolen" from E264 sources and making error-less coding/decoding best at ~7 bits per 8x8 block yuv420p for case if you want upload it on Youtube (thought testes also like 2 years ago) [19:40:16 CET] <Filarius> thought shortly ago i was back to project with idea to rewrite to make cross platform and I found pure ICT isn't working so best as I thought before, also made myself a little-little more undertanding how actually x264 works [20:10:26 CET] <kerio> how do i shove hevc into rtmp? [20:11:07 CET] <JEEB> you don't [20:11:09 CET] <JEEB> it's not specified [20:11:14 CET] <JEEB> go ask Adobe if you need it [20:11:30 CET] <JEEB> so that it appears in annex E of https://wwwimages2.adobe.com/content/dam/acom/en/devnet/flv/video_file_form… [20:12:06 CET] <JEEB> although adobe really felt like they were going the fragmented mp4 route [20:12:33 CET] <JEEB> and FLV (which is what is transferred over RTMP) went the way of the dodo [22:33:23 CET] <M6HZ> Hello, is there a way to extract CEA 608/708 subtitles from h264 stream over http ? This works, but only for local files: ffmpeg -f lavfi -i movie=file.ts[out+subcc] -map 0:1 -f srt - [22:35:56 CET] <JEEB> I don't see it being any more special unless protocols are blacklisted in the lavfi thing [22:36:11 CET] <JEEB> technically you don't need lavfi for the thing at all, but ffmpeg.c doesn't support it [22:36:34 CET] <JEEB> writing an API client that does caption extraction should be pretty simple [22:36:53 CET] <JEEB> just init lavf+lavc and grab the captions from the video packets' side data [22:38:28 CET] <M6HZ> JEEB, here is what I get if I substitute the file name with a url to a ts file: [Parsed_movie_0 @ 0x555934ccec00] Failed to avformat_open_input 'http' [22:38:48 CET] <JEEB> sounds like you need to escape the : [22:38:51 CET] <nicolas17> that's because : separates parameters in a filtergraph [22:40:53 CET] <M6HZ> you mean like that: \: ? [22:40:56 CET] <nicolas17> yes [22:41:06 CET] <M6HZ> It doesn't change anything. [22:41:47 CET] <JEEB> http://ffmpeg.org/ffmpeg-filters.html#Notes-on-filtergraph-escaping [22:42:07 CET] <JEEB> read up this [22:42:08 CET] <JEEB> :) [22:42:15 CET] <M6HZ> ok [22:42:24 CET] <nicolas17> maybe your shell is interpreting \: and sending : to ffmpeg [22:44:02 CET] <M6HZ> nicolas17, I think you're right. [22:46:48 CET] <M6HZ> The result is different if I double the escape \\: [22:46:54 CET] <M6HZ> Now it looks like the option -referer is ignored. [22:47:33 CET] <M6HZ> Is this option incompatible with -lavfi ? [22:48:20 CET] <JEEB> lavfi makes the lavf context by itself so whatever you set with ffmpeg.c probably ain't gonna get passed [22:49:09 CET] <M6HZ> Wow. Ok. [22:49:46 CET] <JEEB> you could see what that movie filter does [22:49:56 CET] <JEEB> and if it has options for passing lavf options [22:50:12 CET] <JEEB> although to be honest if I was in your shoes I'd probably make a simple API client for the FFmpeg APIs [22:52:48 CET] <M6HZ> JEEB, I would rather avoid that since I would like it to be the more portable and with the fewer dependencies possible. [22:53:31 CET] <JEEB> well FFmpeg with its libs are already your dependencies [22:54:44 CET] <nicolas17> I don't understand why the video can't be passed to -i [22:54:58 CET] <JEEB> because ffmpeg.c doesn't have the logic to handle side data packets [22:55:23 CET] <JEEB> for some reason only lavfi's movie filter has that functionality and generates that extra stream from it if it notices the captions might be there [22:55:43 CET] <JEEB> (well I can understand why kind of, ffmpeg.c is a mess) [22:59:45 CET] <M6HZ> Humm, I guess I'm running out of options. [23:02:20 CET] <nicolas17> curl -H "Referer: http://example.com/" http://example.com/video.mp4 | ffmpeg -f lavfi -i movie=/dev/stdin[out+subcc] [23:02:23 CET] <nicolas17> or would movie=- work? [23:03:07 CET] <M6HZ> Yes, obviously, It would work, but this is a m3u8 stream ... [23:03:57 CET] <nicolas17> HLS? ouch [23:04:17 CET] <nicolas17> hmm [23:04:42 CET] <nicolas17> JEEB: would "ffmpeg -i http://blah -codec copy -" at least *preserve* side data? [23:05:53 CET] <JEEB> hmm, that would still have the side data in the AVPacket I guess [23:05:58 CET] <M6HZ> by "side data" do you mean the subtitles ? [23:05:58 CET] <JEEB> so quite possibly yes [23:06:09 CET] <JEEB> M6HZ: they're part of the video stream as side data, so yes [23:06:19 CET] <M6HZ> Yes, they are preserved [23:06:39 CET] <JEEB> nicolas17: basically I don't see any special handling of SIDE_DATA anything in ffmpeg.c so most likely the AVPackets will just get passed on [23:06:58 CET] <nicolas17> then maybe: [23:07:08 CET] <nicolas17> ffmpeg -referer http://example.com/ -i http://example.com/stream.m3u8 -codec copy -f matroska - | ffmpeg -f lavfi -i movie=-[out+subcc] -map 0:1 -f srt - [23:07:35 CET] <nicolas17> maybe another -f is more appropriate I dunno [23:08:25 CET] <M6HZ> nicolas17, even if I do: ffmpeg -i file.ts -c:v h264 -f matroska - | mpv - I can still get the subtitles. [23:08:51 CET] <JEEB> since they're in the video track yea [23:09:07 CET] <JEEB> unless something touches those AVPackets inappropriately it will pass them through [23:09:36 CET] <nicolas17> yeah but -c:v h264 would unnecessarily re-encode the video, you should use -c:v copy since you're ignoring the actual video anyway [23:10:03 CET] <M6HZ> JEEB, ok, interesting. [23:10:32 CET] <M6HZ> nicolas17, yes, I know that, it's just that it was odd to me. [23:10:59 CET] <JEEB> but yea, this is just a hack from hell tbqh :P but hey, I've already voiced my opinion [23:11:32 CET] <nicolas17> isn't ffmpeg.c a hack from hell? :v [23:11:57 CET] <JEEB> yea, but I recommended a simple lavf+lavc API client [23:12:03 CET] <JEEB> since just doing what he needs isn't too hard [23:12:14 CET] <nicolas17> yeah maybe [23:12:38 CET] <JEEB> grab the side data, make AVPackets from them for the caption decoder, pass them to srt stuff [23:13:40 CET] <nicolas17> I tried writing the equivalent of "ffmpeg -f image2 -i frame%03d.png output.mp4" in C and it was somewhat painful, but I may have done it in the middle of the codecpar transition... [23:14:21 CET] <nicolas17> ffprobe on my program's output file gave different results than on ffmpeg's output and it took me forever to find what differed and it was subtle [23:14:27 CET] <M6HZ> JEEB, Do you think it would also be easy to delete these subtitles once I have extracted them from the video ? [23:14:40 CET] <JEEB> in what sense? [23:15:47 CET] <M6HZ> nicolas17, In the end what differed ? [23:15:49 CET] <M6HZ> JEEB, me ? [23:15:54 CET] <M6HZ> At * [23:16:51 CET] <nicolas17> I don't remember anymore ^^ [23:17:04 CET] <JEEB> ffprobe generally doesn't massage stuff as much as ffmpeg.c does in various if clauses [23:17:14 CET] <nicolas17> there is a filter to strip all sidedata [23:17:15 CET] <JEEB> so it's closer to what the APIs return [23:17:21 CET] <M6HZ> JEEB, I would like to extract the subtitles in a text file, and delete them from the h264 stream, then "merge" them again in a more conventional way. [23:18:33 CET] <JEEB> I can see how that is done but no idea about ffmpeg.c [23:20:39 CET] <M6HZ> nicolas17, sidedata, asidedata [23:22:21 CET] <M6HZ> nicolas17, Great ! [23:23:02 CET] <M6HZ> nicolas17, sadly it means that I have to re-encode the whole stream. [23:24:15 CET] <JEEB> https://www.ffmpeg.org/ffmpeg-all.html#toc-Bitstream-Filters :P [23:26:16 CET] <M6HZ> JEEB, Wow interesting, I've never played with this option. [23:49:37 CET] <M6HZ> JEEB, So it looks like I have the choice between "extract_extradata" + "remove" and "h264_metadata" + "delete_filler", apparently the subtitles are located in SEI messages. [23:50:51 CET] <JEEB> yes, whatever removes SEI messages of a certain type. not going to say if that combo is correct or not since extract_extradata is supposed to just put extradata (decoding init data) into the AVCodecContext's extradata field [00:00:00 CET] --- Sun Nov 4 2018

1 0

ffmpeg-devel.log.20181102
by burek 03 Nov '18

03 Nov '18

[00:51:38 CET] <cone-124> ffmpeg 03Michael Niedermayer 07master:acf4fbddf3fe: doc/APIchanges: Update for 4.1 [00:51:39 CET] <cone-124> ffmpeg 03Michael Niedermayer 07master:324d21164d49: Changelog: next is 4.1 [00:51:40 CET] <cone-124> ffmpeg 03Michael Niedermayer 07master:780d5e30a0a0: Bump minor versions for branching 4.1 [00:52:18 CET] <cone-124> ffmpeg 03Michael Niedermayer 07release/4.1:HEAD: Bump minor versions for branching 4.1 [01:25:41 CET] <cone-124> ffmpeg 03Michael Niedermayer 07master:517573a67088: Bump minor version for master after 4.1 branchpoint [01:25:42 CET] <cone-124> ffmpeg 03Michael Niedermayer 07master:e8ced83f68a7: Changelog: Add new version next tag [01:25:43 CET] <cone-124> ffmpeg 03Michael Niedermayer 07master:f4dd2db91301: RELEASE: bump to 4.1.git [02:44:07 CET] <cone-124> ffmpeg 03Michael Niedermayer 07release/4.1:3c7e97343065: Update for 4.1 [02:44:08 CET] <cone-124> ffmpeg 03Michael Niedermayer 07release/4.1:1665ac6a4461: RELEASE_NOTES: Based on the version from 4.0 [02:56:44 CET] <cone-124> ffmpeg 03James Almer 07master:2d2af23349ca: avformat/ivfenc: use the av1_metadata bsf to insert Temporal Delimiter OBUs if needed [02:58:09 CET] <cone-124> ffmpeg 03kjeyapal(a)akamai.com 07master:d6d407d2d758: avformat/dashenc: Disable writing CODECS tag for HEVC streams [04:09:17 CET] <philipl> BtbN: I have posted an updated patchset, mostly just in case michaelni wanted to see it after updates. [09:34:10 CET] <cone-122> ffmpeg 03Paul B Mahol 07master:97e9e3504d36: avfilter: add tpad filter [09:40:35 CET] <BradleyS> :) [11:26:38 CET] <cone-122> ffmpeg 03Paul B Mahol 07master:4620e4c7ad2f: avfilter/f_loop: make sure duration of frame is never zero [12:07:54 CET] <atomnuker> BBB: had the time to play with the code? [12:16:50 CET] <BBB> atomnuker: not yet, finishing SGR first& one thing at a time [12:17:04 CET] <BBB> av1 = big codec, takes some time to get to everything [12:35:08 CET] <cone-122> ffmpeg 03Paul B Mahol 07master:ef1aadffc785: avfilter/f_loop: switch to activate [14:50:22 CET] <atomnuker> BBB: did a minor update on the PR, using pand like you suggested [14:50:42 CET] <atomnuker> are 3 argument instructions more expensive to encode than 2 argument versions? [14:53:53 CET] <BBB> I dont think so [14:54:05 CET] <BBB> 2-arg is automatically expanded into 3-arg when using vex encoding, no? [14:54:14 CET] <BBB> at least thats what the lldb disassembly suggests [14:58:08 CET] <atomnuker> also ALIGN() definitely does not work, none of the cdfs are actually aligned, mova segfaults [15:04:38 CET] <BBB> is the cdf itself algined? [15:05:05 CET] <BBB> hm& it should be [15:05:06 CET] <BBB> it is a ref [15:05:25 CET] <BBB> cdf.c:4051 is not though [15:05:29 CET] <BBB> static CdfThreadContext cdf_init[4] = { .. [15:05:35 CET] <BBB> that needs to be aligned also [15:17:46 CET] <jamrial> it's complaining about seta on win32 [15:18:05 CET] <jamrial> probably x86_32 in general [15:27:15 CET] <atomnuker> yeah, is 32 bit windows not able to use byte regs or something? [15:27:43 CET] <BBB> 32bit? only a few regs have bye equivalents [15:27:51 CET] <BBB> for 64bit I thought they all did [15:27:55 CET] <BBB> but I dont recall exactly [15:29:01 CET] <jamrial> also, it's still slower than what gcc compiles [15:29:42 CET] <nevcairiel> A/B/C/D have byte regs [15:30:09 CET] <BBB> Ill play around with this a bit when I have time [15:30:29 CET] <BBB> it should be faster when you remove the extra calls [15:30:33 CET] <nevcairiel> but that comes back to comment from yesterday, in x86inc.asm which register is behind which name varies between platforms [15:35:27 CET] <nevcairiel> although r2 or r3 should map to a register with a byte component on all platforms [15:35:36 CET] <nevcairiel> if oyu can use one of those two [15:45:34 CET] <BBB> if you use avx2, just make it 64bit only [15:45:39 CET] <BBB> no reason to use avx2 on 32bit platforms [15:45:50 CET] <BBB> lets move people off 32bit [15:46:14 CET] <nevcairiel> you have more choices then for sure, but still like 4 legacy registers left that dont have one :D [15:46:45 CET] <nevcairiel> or 3 really, since you dont use the 4th [15:48:00 CET] <atomnuker> I could do that I suppose but it'll cost 1 more vector reg [15:57:28 CET] <Gramner> up to 6 vector regs it doesn't matter. e.g using two or six is all the same. above 6 becomes slower on win64, and above 8 becomes 64-bit only [16:02:11 CET] <atomnuker> does that apply for all platforms? [16:02:40 CET] <atomnuker> also does it apply to gprs [17:08:06 CET] <Gramner> atomnuker: volatile registers per platform: x86-32: 3 gpr, 8 vec. win64: 7 gpr, 6 vec. unix64: 9 gpr, 16 vec [17:08:33 CET] <Gramner> anything more than that costs some performance since registers have to be saved and restored [17:09:57 CET] <Gramner> avx-512 adds another 16 volatile vector regs on x68-64, so you can use up to 22 on win64 without having to save stuff [17:10:12 CET] <nevcairiel> callee saved vector registers are so weird [17:10:29 CET] <Gramner> yes. it makes no sense because they're mostly used in leaf functions [17:10:47 CET] <BBB> atomnuker: I have a fix for you [17:10:52 CET] <Gramner> I could _maybe_ understand having like 1 or 2 being callee-saved but certainly not the majority [17:11:05 CET] <BBB> I can get it down from 14.738 to 14.661 in decoding time for Chimera first 1k frames [17:11:12 CET] <BBB> atomnuker: I havent tuned anything yet, but it works at least [17:11:17 CET] <BBB> and it appears to not be slower anymore [17:11:45 CET] <BBB> I bet you can bring it down further by removing the msac_update_cdf for the boolean one and instead simply having a special version that updates the boolean one without copying [17:11:46 CET] <BBB> anyway [17:12:56 CET] <BBB> atomnuker: https://code.videolan.org/rbultje/dav1d/commits/atomnuker-cdf-update-simd [17:13:11 CET] <BBB> atomnuker: I threw out half of your changes, please dont be angry, was just trying to boil it down to something simple for the proof of concept [17:13:19 CET] <BBB> so all changes to cdf.c are gone [17:13:26 CET] <BBB> and the wrapper for the avx2 function is also gone [17:13:49 CET] <BBB> this works for me, generates same md5, and is faster (a little). I think with some tuning you can speed it up a lot more [17:15:42 CET] <BBB> atomnuker: one thing that may be interesting is to try and have a version for n_syms == 16 and then another for n_syms < 16 which writes the counter etc. all in avx2 instead of gpr [17:15:48 CET] <BBB> since then its just one read and one write [17:15:52 CET] <BBB> not sure how useful that would be though [17:16:39 CET] <BBB> atomnuker: lmk if you want to discuss or whatever [17:17:55 CET] <BBB> atomnuker: but most of it shouldbe fairly obvious [18:13:51 CET] <nicolas17> when using a big image sequence as input, often I see neither CPU nor I/O are saturated, because ffmpeg reads one file (CPU idle) and then decodes it (I/O idle) [18:13:53 CET] <nicolas17> if I do "cat *.jpg > /dev/null" in the background, overall performance improves [18:13:54 CET] <nicolas17> would it make sense to add pre-fetching to ffmpeg itself? [18:14:43 CET] <philipl> BtbN: https://github.com/philipl/nv-codec-headers/commit/576d39d4bde190989925b290… [18:14:59 CET] <philipl> Ticket #7530 is a valid complaint. [18:15:46 CET] <nevcairiel> that function is CUDA 9.2, which matches the documented driver requirement for the 8.2 branch [18:16:20 CET] <nevcairiel> if he gets past that function itll probably get stuck on some other function [18:16:24 CET] <nevcairiel> this is just the first [18:21:50 CET] <philipl> nevcairiel: technically, they say CUDA 8.0 is the base requirement. [18:22:04 CET] <philipl> Now, the driver requirements for Video SDK 8.2 imply the presence of this function. [18:22:08 CET] <nevcairiel> then i dont know where the driver versions in the README file come from [18:22:34 CET] <philipl> Those come from the Video SDK stated requirements. [18:22:46 CET] <nevcairiel> in any case, those are high enough for your function [18:23:02 CET] <philipl> Yes. But it should be made optional or not included for the older SDK releases. [18:23:09 CET] <philipl> Header releases, rather. [18:23:22 CET] <nevcairiel> its only useful for the CUDA 10 interop anyway, isnt it? [18:23:26 CET] <nevcairiel> so just not backport it [18:23:27 CET] <philipl> So the 8.0 and 8.1 branches shouldn't have it. [18:23:28 CET] <philipl> Right. [18:23:32 CET] <philipl> BtbN: did backport it. :-P [18:23:36 CET] <nevcairiel> ah [18:23:38 CET] <nevcairiel> oh well [18:23:39 CET] <philipl> So I guess the right answer is to revert on the old branches. [18:24:17 CET] <philipl> but the person doing the ffmpeg build might use the latest headers and this shouldn't break people not using vulkan interop [18:24:25 CET] <philipl> so I think making it optional on the 8.2 headers is correct. [18:24:33 CET] <philipl> The same way the rest of the vulkan interop functions are optional. [18:24:42 CET] <nevcairiel> not that it hurts anything [18:24:47 CET] <nevcairiel> but i dont think it'll fix that guy [18:24:54 CET] <philipl> It can fix him. [18:25:10 CET] <philipl> Well, maybe. It's confusing. [18:25:21 CET] <philipl> I'd rather not try and work it out and just make it optional so the vulkan stuff is consistent. [18:25:22 CET] <nevcairiel> if he is u sing the 8.2 headers, those would require a new driver anyway [18:25:36 CET] <nevcairiel> but yeah just making it optional would be f ine [18:25:57 CET] <philipl> It's fuzzy. if he tried to use certain nvenc features, it would fail on old drivers. nvdec would work. [18:26:24 CET] <BtbN> philipl, only mpv is using it so far, is it? [18:26:26 CET] <nevcairiel> does ffmpeg still build w ith 8.0/8.1? [18:26:43 CET] <BtbN> nevcairiel, it should, I didn't do a test build in a while though [18:27:05 CET] <atomnuker> BBB: this is nuts [18:27:12 CET] <BBB> ? [18:27:45 CET] <philipl> BtbN: yeah, only mpv [18:28:08 CET] <atomnuker> BBB: its very clever [18:28:19 CET] <BBB> atomnuker: so, few suggestions for improvements [18:28:25 CET] <philipl> BtbN: It already does a null check for the vulkan functions so I don't even have to add a guard to make this one optional. [18:28:43 CET] <BBB> atomnuker: first of all, remove the shift using [rsp], instead use pmulhuw using a constant vpbroadcastded using r4q [18:29:02 CET] <BBB> e.g. times 2 dw 4096, times 2 dw 2048, times 2 dw 1024, times 2 512 [18:29:11 CET] <BBB> and then [offset+r4q*4] [18:29:46 CET] <BBB> then secondly, I think theres merit in having a jumptable and doing 4 versions (maybe even 5) for this function [18:29:54 CET] <BBB> one for <8, one for ==8, one for <16, one for ==16 [18:30:08 CET] <BBB> the first one does xmm registers only and can do the counter in the simd code [18:30:28 CET] <BBB> (pcmpgtw pw_32, etc., then or using fixed 0xff psrldqed by 2 ored into the final paddw register) [18:30:30 CET] <BBB> same for <16 [18:30:37 CET] <BtbN> philipl, it's a CUDA function though [18:30:38 CET] <BBB> then for ==8 and ==16, do the same as you do now in xmm/ymm registers [18:30:50 CET] <philipl> BtbN: I mean the interop cuda functions. [18:30:50 CET] <BBB> ad maybe a amaybe a fifth one for ==1 (which would be the boolean one) [18:30:58 CET] <BtbN> philipl, ok. [18:31:02 CET] <BBB> mc shows how to do jumptables [18:31:04 CET] <BtbN> I'll do another round of releases then [18:31:47 CET] <BBB> atomnuker: and please add a checkasm test, this is easily testable :) [18:32:02 CET] <BBB> that also allows exploring performance gains per n_symbol number etc. [18:34:52 CET] <BtbN> philipl, cuDeviceGetAttribute is around for forever, right? [18:35:06 CET] <BtbN> Oldest reference I can find is it being in CUDA 8.0, but it's probably even older [18:35:27 CET] <philipl> BtbN: CUDA 8.0 is old enough to equal forever. All SDK releases we support say they require CUDA 8 [18:36:24 CET] <BtbN> true [18:38:57 CET] <philipl> michaelni: do you want to re-review the yadif refactoring, or are you happy for me to push? [18:39:18 CET] <nevcairiel> bonus p oints if you sneak it into 4.1 before it freezes [18:39:24 CET] <philipl> woo [18:39:52 CET] <philipl> I can sneak in all sorts of things if I skip the review step. :-) [18:45:55 CET] <BBB> atomnuker: Im also guessing that when you use cmov and clz, you can prevent all the conditional code in the beginning [18:47:55 CET] <BBB> atomnuker: something like clz bla, word [cdfq+nsymsq*2], cmp bla, 28 cmovle bla, 28 (?) [18:48:30 CET] <BBB> atomnuker: and then you can use that to calculate the offset into the pmulhuw table [18:48:50 CET] <BBB> atomnuker: I know, a little crazy, but it sounds doable right? [18:49:48 CET] <BBB> but please make a checkasm test so we can explore speed gains from each of these on various n_symbols numbers [18:49:57 CET] <BBB> otherwise it gets fairly nonsensical [18:51:05 CET] <philipl> BtbN: so want me to push? [18:51:17 CET] <BtbN> push? [18:51:52 CET] <BtbN> The nv-codec-header stuff is already done [18:52:45 CET] <BtbN> Or you mean yadif? [18:52:49 CET] <nevcairiel> you didnt push 8.0 :( [18:52:57 CET] <BtbN> I'm pretty sure I did [18:53:07 CET] <nevcairiel> oh you pushed a tag [18:53:09 CET] <nevcairiel> but not the branch [18:53:11 CET] <BtbN> oh [18:53:23 CET] <BtbN> fixed [18:53:38 CET] <philipl> BtbN: I meant nv-codec-header. but you work fast :-) [18:54:55 CET] <philipl> I want to push yadif but michaelni was not explicit as to his level of satisfaction. [18:55:32 CET] <BtbN> I typoed the commit... well, too late now [18:55:40 CET] <philipl> You can force push it. :-) [18:55:49 CET] <BtbN> Pretty sure force pushes are blocked [18:57:03 CET] <philipl> only one way to be sure. [18:57:30 CET] <BtbN> It would also mean re-doing the whole releases [18:57:32 CET] <BtbN> not worth it [18:59:20 CET] <michaelni> philipl, i think its ok to push the yadif change, sorry that my reply was unclear [19:10:31 CET] <nicolas17> guess I should post to the mailing list instead... [19:11:13 CET] <tmm1> i wish yadif had neon simd [19:11:25 CET] <nevcairiel> write some [19:11:26 CET] <nevcairiel> :D [19:11:35 CET] <nicolas17> tmm1: compiler autovectorizer not smart enough yet? :) [19:11:56 CET] <nevcairiel> never [19:12:01 CET] <nicolas17> :P [19:12:32 CET] <tmm1> i wrote a simple blend deint using neon intrinsics [19:12:37 CET] <tmm1> yadif is 100x more complicated [19:13:02 CET] <nevcairiel> well you have the x86 simd to steal a concept from [19:13:25 CET] <nicolas17> does ffmpeg "support" having an image2 sequence with gaps? or is the result undefined? [19:16:40 CET] <nicolas17> if I have frame{1,2,3,4,5,6,8,9,10}.png, and I use ffmpeg -i %d.png, is it supposed to transcode frames 1-6, or transcode 1-10 and skip the missing 7, or give an error because of missing 7, or "there's no guarantees, don't do that"? [19:17:29 CET] <durandal_1707> nicolas17: error, also wrong channel [19:23:24 CET] <philipl> michaelni: thanks! [19:25:09 CET] <JEEB> can someone note if this one is OK so I can push both https://patchwork.ffmpeg.org/patch/10574/ (which was LGTM'd) and it [19:25:15 CET] <JEEB> asdf [19:25:20 CET] <JEEB> https://patchwork.ffmpeg.org/patch/10583/ [19:25:25 CET] <JEEB> is the first one I meant, which wasn't ack'd [19:25:30 CET] Action: JEEB is clearly tired [19:26:14 CET] <JEEB> or should I add the dts there in that message as well? :V [19:26:30 CET] <JEEB> since the offset by itself might not be useful possibly [19:27:56 CET] Action: durandal_1707 wonders when users will appear wondering what new message means [19:28:40 CET] <JEEB> it's not a new message, but generally if you hit that you're going to get A/V desync [19:28:55 CET] <JEEB> at least in most cases I've had that happen it means that the vsync logic in ffmpeg.c is going to crash town :P [19:29:08 CET] <JEEB> either due to weirdness in the input or otherwise [19:30:10 CET] <JEEB> I have a patch that stops that intra stream discontinuity check from triggering if a stream of the "opposing" type (video if audio, and vice versa) has already reached the jumped-to point [19:32:29 CET] <nicolas17> durandal_1707: I ask what it's *supposed* to do because the actual behavior is certainly weird https://paste.kde.org/pbemjvxv7 [19:33:41 CET] <cone-675> ffmpeg 03Philip Langdale 07master:598f0f39271d: libavfilter/vf_yadif: Make frame management logic and options shareable [19:33:41 CET] <cone-675> ffmpeg 03Philip Langdale 07master:d5272e94ab22: avfilter/vf_yadif_cuda: CUDA accelerated yadif deinterlacer [19:33:41 CET] <cone-675> ffmpeg 03Philip Langdale 07master:1b41115ef708: avcodec/nvdec: Increase frame pool size to help deinterlacing [19:34:24 CET] <durandal_1707> nicolas17: gaps are not supported, unless you use glob pattern [19:34:25 CET] <nicolas17> find_image_range is doing some exponential search for the last file in the sequence [19:36:30 CET] <nicolas17> if find_image_range happens to skip over the gap, then the encoding process will give an error when it can't open the missing file; otherwise, it will successfully encode up to the missing file [20:26:26 CET] <BtbN> philipl, https://travis-ci.org/FFmpeg/FFmpeg-Coverity/builds/449984694#L1824 [20:26:28 CET] <BtbN> oO [20:39:22 CET] <BtbN> Why? The configure check looks right to me. [20:40:00 CET] <BtbN> hm, cuda_sdk _is_ enabled on there [20:40:05 CET] <BtbN> how [20:41:04 CET] <BtbN> philipl, seems like it doesn't build on older cuda SDKs, namely on the one Debian Stretch uses. [20:51:42 CET] <BBB> atomnuker: https://code.videolan.org/videolan/dav1d/merge_requests/289 for you [20:53:37 CET] <atomnuker> BBB: just write that into the msac_decode_symbol_adapt function [20:53:47 CET] <atomnuker> I'm not sure if a branching asm code would be worth it [20:54:04 CET] <BBB> maybe I should move that function from header to .c file [20:54:14 CET] <BBB> otherwise well blow up the binary size of decode.c by a lot [20:54:44 CET] <atomnuker> yeah, compiler will inline msac_decode_bool anyway so you save a function call [21:01:32 CET] <philipl> BtbN: I'll look. [21:02:13 CET] <BBB> done [21:05:19 CET] <durandal_1707> anybody against audio match/align report filter? [21:05:21 CET] <BtbN> philipl, I sent new docker images on the way with a more up to date CUDA SDK (9.2 iirc) [21:08:02 CET] <atomnuker> BBB: no need for a function for msac_update_bool_cdf, just make the msac_decode_bool_adapt do the adaptation [21:09:22 CET] <atomnuker> also I don't know the coding style though and through but I don't think msac_update_cdf needs the msac_ prefix as its no longer exposed [21:09:27 CET] <BBB> you dont think its better self-documentation? [21:09:28 CET] <atomnuker> other than that lgtm [21:09:35 CET] <BBB> oh right msac_ prefix should go, yes [21:09:46 CET] <atomnuker> its fine, _adapt implies it does adaptation [21:10:01 CET] <BBB> right [21:12:23 CET] <BBB> done [21:13:42 CET] <durandal_1707> michaelni: you havent pushed prosumer changes, they need to go to 4.1 too [21:16:38 CET] <atomnuker> BBB: yeah, looks fine [21:18:56 CET] <durandal_1707> JEEB: that log message patch should be fine [21:19:58 CET] <JEEB> durandal_1707: I only poked because 2/2 never got any straight acks even though I could take the "you should also raise teh level for that other one, too" as someone having a positive opinion on it [21:29:29 CET] <philipl> BtbN: apparently that initialization syntax for built-in vectors is new... [21:29:55 CET] <philipl> I don't want to change it because the old way requires you to know the exact type (so uchar2 vs ushort2 becomes a problem) [21:36:52 CET] <BtbN> yeah, it's probably fine [21:36:56 CET] <BtbN> what syntax exactly? [21:37:13 CET] <philipl> uchar2 foo = { 1, 2 }; [21:37:29 CET] <philipl> I would need to do: uchar2 foo = make_uchar2(1, 2); [21:42:19 CET] <nicolas17> durandal_1707: audio align? that sounds like something I could use... [21:44:12 CET] <durandal_1707> nicolas17: yes, it would report best match between two (or more?) tracks so one could align them for example when mixing, it would not do actual aligning [21:45:41 CET] <BtbN> philipl, can probably do some C++ template magic [21:45:59 CET] <philipl> BtbN: well, the point is my current syntax does do C++ magic. [21:46:07 CET] <BtbN> more [21:46:49 CET] <BtbN> You can make a make_vector<T> and specifically implement it for the types you use [21:47:03 CET] <BtbN> and then use that in your template [21:48:53 CET] <philipl> Does that exist in cuda? [21:48:58 CET] <philipl> Oh, I see. [22:02:24 CET] <nicolas17> durandal_1707: I think the audio I need to align like that would be a bitwise match, so I probably have easier ways to do it, but your filter still sounds interesting :) [22:03:45 CET] <durandal_1707> nicolas17: well it will report score, so if its inf you know its bitexact :) [22:11:09 CET] <philipl> BtbN: https://github.com/philipl/FFmpeg/commit/9df533144fcead8b416ba928d5aefb8de5… [22:15:23 CET] <BtbN> it builds on CUDA 10/VS2017 at least [22:35:12 CET] <philipl> BtbN: https://github.com/philipl/FFmpeg/commit/8e50215b5e02074b0773dfcf55867654ee… [22:35:15 CET] <philipl> even simpler. [22:35:21 CET] <philipl> Obvious in retrospect. [22:49:32 CET] <cone-671> ffmpeg 03Philip Langdale 07master:8e50215b5e02: avfilter/vf_cuda_yadif: Avoid new syntax for vector initialisation [22:56:04 CET] <philipl> BtbN, nevcairiel: Well, with that out of the way, we're back to pix fmts. Yay! [23:47:51 CET] <cone-671> ffmpeg 03Michael Niedermayer 07master:9acdf17b2c30: avcodec/prosumer: Check for bytestream eof in decompress() [23:47:52 CET] <cone-671> ffmpeg 03Michael Niedermayer 07master:506839a3e9cc: avcodec/prosumer: Remove unneeded () [23:47:53 CET] <cone-671> ffmpeg 03Michael Niedermayer 07master:1dfa0b6f36d2: avcodec/prosumer: Remove always true check in decompress() [23:47:54 CET] <cone-671> ffmpeg 03Michael Niedermayer 07master:66425add270c: avcodec/prosumer: Simplify bit juggling of the c variable in decompress() [00:00:00 CET] --- Sat Nov 3 2018

1 0

ffmpeg.log.20181102
by burek 03 Nov '18

03 Nov '18

[04:43:35 CET] <grosso> hi [04:44:44 CET] <grosso> any chance to get ffplay utility in android? [07:08:00 CET] <grosso> ffplay for android? [08:02:13 CET] <Mavrik> That would be pretty silly to have. [10:43:09 CET] <friki> grosso: I have a Debian chroot with xorg+vnc server in my rooted Android (LineageOS). I've tested ffmpeg and ffplay, both work fine even to encode video. Probably it isn't what you are looking for ;-) [11:50:30 CET] <kingmanor> is there a reason the static builds are compiled with gcc 6 vs gcc 8? [12:13:51 CET] <relaxed> kingmanor: I encountered too many build issues with 8, if you're asking about https://www.johnvansickle.com/ffmpeg/ [12:23:17 CET] <kingmanor> yeah i guess i am as that is the main one linked on the download page. yes ubuntu ffmpeg is built on gcc8 but it is 4.0.2 release i was looking for latest. [12:24:01 CET] <kingmanor> its ok im gonna try to build it myself [12:26:54 CET] <relaxed> kingmanor: if you haven't seen it, check out https://trac.ffmpeg.org/wiki/CompilationGuide [14:16:54 CET] <zivanovicb> we have to transcode one video which is 1080p to 1080p,720p,480p, all in mp4,webm & hls, which would be 9 different outputs i guess [14:17:11 CET] <zivanovicb> is it better to do all this in one ffmpeg instance using multiple outputs or break it down in their own separate ffmpeg instances? [14:18:18 CET] <zivanovicb> its also really important that original resolution(1080p) finishes as fast as possible [15:37:16 CET] <Shibe> hi, what is wrong with this command? sudo ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -vcodec h264_vaapi -f pulse -ac 2 -i 1 -crf 0 -preset ultrafast -acodec pcm_s16le output.mkv [15:37:22 CET] <Shibe> Option vf (set video filters) cannot be applied to input url 1 -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to. [15:41:55 CET] <relaxed> try, ffmpeg -f kmsgrab -i - -f pulse -ac 2 -i 1 ... [15:44:59 CET] <Shibe> relaxed: sudo ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -vcodec h264_vaapi -f pulse -ac 2 -preset ultrafast -acodec pcm_s16le -i 1 output.mkv [15:45:04 CET] <Shibe> Option vf (set video filters) cannot be applied to input url 1 -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to. [15:45:55 CET] <durandal_1707> move -vf before output.mkv [15:47:57 CET] <relaxed> Shibe: your inputs go first in the command [15:50:59 CET] <Shibe> relaxed: 'sudo ffmpeg -f kmsgrab -vcodec h264_vaapi -f pulse -ac 2 -preset ultrafast -acodec pcm_s16le -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' output.mkv' like this? [15:58:30 CET] <relaxed> Shibe: ffmpeg <demuxer/format 1 options> -i <input 1> <demuxer/format 2 options> -i <input 2> <filters & encoding options> <output> [15:59:20 CET] <relaxed> your last command has no inputs (-i) [16:16:44 CET] <Shibe> relaxed: works perfect, thanks! [20:22:29 CET] <teratorn> is there a way to select [*] (all) audio and video stream pairs from the input list in an ffmpeg filtergraph? [20:24:33 CET] <durandal_1707> teratorn: huh, you can only pick them, there is no * thing [20:25:30 CET] <furq> teratorn: if it's the first filter in the list then maybe don't set an input label [20:25:43 CET] <furq> that'll give you all the input streams in the order they were specified [20:31:42 CET] <teratorn> furq: i was thinking that ^ [20:31:55 CET] <teratorn> would be nice to have some simple patterns in "label expansion" :) [20:32:18 CET] <furq> you could also do [0][1]... if you know how many inputs you have [20:42:49 CET] <grosso> hi [20:44:18 CET] <grosso> can I receive a rtsp stream with ffmpeg and save to sequence of images realtime? [20:46:18 CET] <edgy> Hi, In https://trac.ffmpeg.org/wiki/Encode/H.264 "It is not required if you are going to use a video service such as YouTube. YouTube recommends using faststart" Isn't there a contradiction here? I thought the phrase should say "It is required" [20:50:48 CET] <edgy> Also the channel title says 4.0.1, shall I tell you that 4.0.2 is released? ;) [20:52:53 CET] <HarveyPwca> An odd thing is happening when I am using ffmpeg. After ripping a film from a dvd and combining the vob files into a single file I use ffmpeg (cli) to select the video, audio and subtitles desired. ffmpeg starts out really well at 16x but then slooooowwwws down to <1x. Never used to do this. Anyone know how I can figure out how/why this is happening? [20:55:45 CET] <HarveyPwca> Generally the command I'm using is: ffmpeg -i [film name] -map 0:0 -c:v copy -map 0:1 -c:a copy -map 0:2 -c:s copy [new file] [20:56:41 CET] <HarveyPwca> Depending on the film the 0:x for desired language/subtitles changes. [20:57:09 CET] <HarveyPwca> So the above is just a general command. [20:57:33 CET] <durandal_1707> you sure it doesnt start using swap? [20:58:12 CET] <HarveyPwca> Never looked before. But looking just now it seems to be. [21:18:17 CET] <HarveyPwca> It seems to be using very little of the system memory and none of the swap space. [21:18:52 CET] <HarveyPwca> System memory fluctuates between 11% - 16%. [21:19:56 CET] <HarveyPwca> Gawd it's slow. It's taken 45 minutes to process a lousy 10 minutes of film. [21:20:13 CET] <durandal_1707> HarveyPwca: pastebin full uncut ffmpeg output [21:22:10 CET] <edgy> any help? [21:23:20 CET] <durandal_1707> edgy: no, i see no contradiction [21:23:40 CET] <furq> edgy: it'll start processing faster if you use faststart, but it'll work fine either way [21:24:08 CET] <furq> although i'd just use mkv in any case because that doesn't require a second pass [21:24:33 CET] <edgy> furq: ah! thanks! [21:24:52 CET] <edgy> furq: you mean mkv instead of mp4? [21:24:55 CET] <furq> yeah [21:25:07 CET] <furq> faststart mp4 with ffmpeg has to copy the entire file after writing it [21:25:38 CET] <edgy> furq: but youtube recommends mp4 not mkv [21:25:48 CET] <furq> yeah youtube recommends a lot of insane things that don't matter [21:25:49 CET] <edgy> furq: I don't know why though [21:25:57 CET] <edgy> furq: ;) [21:26:17 CET] <furq> probably half of the things on that youtube recommendation page are either irrelevant or wrong [21:27:04 CET] <edgy> furq: that's really strange! Can you please give me the command you recommend for youtube? I will trust you better than google ;) [21:27:13 CET] <furq> what are you using now [21:28:04 CET] <furq> i checked again and they fixed a lot of the stupider recommendations on there [21:28:16 CET] <furq> 2 consecutive B frames [21:28:16 CET] <furq> Closed GOP. GOP of half the frame rate. [21:28:20 CET] <furq> these are the only bits i disagree with [21:28:37 CET] <furq> just keep both of those at the x264 defaults [21:29:06 CET] <edgy> furq: may be something like ffmpeg -i input -c:v libx264 -preset slow -profile:v high -crf 18 -coder 1 -pix_fmt yuv420p -movflags +faststart -g 30 -bf 2 -c:a aac -b:a 384k -profile:a aac_low output [21:29:10 CET] <HarveyPwca> durandal_1707: Here is the pastebin link, I hope... https://pastebin.com/eYpch2Rs [21:29:19 CET] <furq> yeah just get rid of -g 30 -bf 2 [21:29:24 CET] <furq> and also 384k aac is absolutely insane [21:29:29 CET] <furq> they got rid of that recommendation now [21:29:45 CET] <furq> honestly you shouldn't reencode the audio at all [21:30:31 CET] <furq> get rid of -profile high, let x264 set that automatically (it'll probably be high anyway) [21:30:34 CET] <furq> and -coder 1 [21:30:38 CET] <edgy> furq: not sure whether -g 30 or -g 15 maybe? [21:30:38 CET] <edgy> furq: I copied this from somewhere [21:30:44 CET] <furq> just get rid of it entirely [21:31:01 CET] <furq> x264 has sensible defaults for all of this stuff [21:31:30 CET] <furq> and then just use -c:a copy and get rid of all the other audio stuff [21:32:49 CET] <edgy> furq: thanks, though this stuff is a bit difficult for me but if i might ask, why do you recommend removing -bf 2? [21:33:54 CET] <furq> there's just no reason to manually set bframes when youtube is reencoding anyway [21:34:17 CET] <furq> as a general rule you shouldn't touch any x264 option other than preset, tune and crf unless you know what you're doing [21:35:04 CET] <furq> basically i would do -i foo.mp4 -c:v libx264 -c:a copy -preset slow -crf 18 -pix_fmt yuv420p bar.mkv [21:35:07 CET] <furq> brb [21:36:24 CET] <edgy> furq: Thanks! what's tune and what's brb? execuse my english [21:37:09 CET] <HarveyPwca> edgy: brb = be right back [21:38:02 CET] <edgy> HarveyPwca: thanks and tune? [21:38:10 CET] <HarveyPwca> Not a clue. [21:39:01 CET] <relaxed> edgy: it's detailed here, https://trac.ffmpeg.org/wiki/Encode/H.264 [21:39:34 CET] <durandal_1707> HarveyPwca: how big is input file? [21:39:53 CET] <edgy> relaxed: thanks! [21:39:59 CET] <HarveyPwca> 4.7Gb [21:42:45 CET] <HarveyPwca> I should also note that this situation only occurs when I'm doing filename.vob to newfilename.vob. If I'm doing filename.mkv to newfilename.mkv (or *.mp4 or *.mkv to *.mp4) there's no slowdown at all. Zips right through these similarly sized files in ~20min or less. [21:44:11 CET] <HarveyPwca> in.vob to out.vob can take 4 or more hours. [21:44:36 CET] <furq> not that i think that's intended behaviour, but why would you copy it to another vob [21:45:44 CET] <relaxed> HarveyPwca: you're trying to create an SVCD? [21:45:51 CET] <HarveyPwca> furq: I'm removing extraneous audio and subtitle streams by selecting only those streams I want to keep. [21:46:14 CET] <durandal_1707> HarveyPwca: does same happens if you instead of final vob use -f null - ? if no than vob muxer is doing something silly [21:47:21 CET] <HarveyPwca> What is "-f null -"? [21:47:41 CET] <furq> well it's not the vob muxer for starters [21:47:42 CET] <HarveyPwca> Or did you mean "-f null"? [21:48:48 CET] <nicolas17> "-f null" picks the null muxer which does nothing and sends data nowhere, but ffmpeg still wants a filename, "-" is enough to satisfy it [21:50:08 CET] <HarveyPwca> Ah...just looked up the null command and read it a bit. Will try that now. Give me a couple of minutes to respond. [21:52:45 CET] <relaxed> also try -f dvd [21:53:04 CET] <furq> or just remux to mkv instead of vob [21:53:13 CET] <HarveyPwca> Zowie! That sucker screams speed wise. Gets up into the 200x+ speeds. However, after a few seconds I get this: [21:53:34 CET] <HarveyPwca> [null @ 0x55e076b50260] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 6474240 >= 6474240 [21:55:24 CET] <HarveyPwca> The value * >= * increments with same message. [21:56:31 CET] <HarveyPwca> furg: Never thought to just remux. I'll try that now. Let you know in a moment the results. [21:57:16 CET] <nicolas17> you *are* doing a remux, just into the same format as the input :P [21:57:36 CET] <furq> yeah there's no point using vob as the output [21:57:51 CET] <furq> especially if it's bigger than 1GB because then it's an entirely worthless format [21:57:56 CET] <HarveyPwca> Yes. I'm seeing that now. [21:57:57 CET] <furq> given that you can't put it back onto a dvd [21:58:56 CET] <HarveyPwca> Well, actually the file gets smaller than the original because I'm negatively discarding the unwanted audio streams. [21:59:19 CET] <nicolas17> you won't turn 4.7GB into 900MB by discarding audio :) [21:59:20 CET] <HarveyPwca> Most of the time the file starts at 4.7Gb and ends up being half that or less. [22:01:24 CET] <tdr> that would have to be a lot of audio streams and channels [22:01:50 CET] <nicolas17> or long video with really bad video quality [22:01:57 CET] <HarveyPwca> Okay. The .vob to .mp4 finished in record time. Video and Audio are correct but waaayyyy off in being synced. [22:03:17 CET] <HarveyPwca> I think the .vob files are seriously borked somehow. Will just go back and re-rip the film. Thanks for all the help everyone. :-) [22:38:20 CET] <grosso> hi [22:38:32 CET] <grosso> can I ask a question? [22:38:41 CET] <tdr> dont ask to ask [22:38:46 CET] <grosso> ok [22:39:10 CET] <grosso> i'm trying to receive a rtsp stream [22:39:38 CET] <grosso> so far I can receive it, my problem is just the "output" part of the command line [22:40:11 CET] <grosso> no mather what I put as "output" the command never writes it in real time [22:41:04 CET] <grosso> so, if i set to output a sequence of images, it don't write anything, then when the streamer stops, it begins to write the images [22:43:44 CET] <grosso> what command should I use to write a video file or a sequence of images while stream is coming? [22:44:06 CET] <grosso> i.e. "realtime" [22:44:11 CET] <tdr> you should prob paste what you're using now so folks can see what may be wrong [22:44:21 CET] <grosso> ok,here I go [22:46:12 CET] <grosso> ffmpeg -rtsp_flags listen -i rtsp://192.168.1.44:8559/live.sdp test.mp4 [22:47:40 CET] <grosso> or ffmpeg -rtsp_flags listen -i rtsp://192.168.1.44:8559/live.sdp -r 25 img-%02d.png [22:49:43 CET] <nicolas17> are you sure your CPU and disk can write 25 PNG images per second? [22:49:45 CET] <relaxed> ffmpeg -i 'rtsp://192.168.1.44:8559/live.sdp' -c copy output.mkv [22:49:59 CET] <grosso> no, I'm not shure [22:51:13 CET] <grosso> i don't need specifically 25 png per second, what I really need is any way to display that output in the screen of an android device, so no ffplay [22:52:02 CET] <relaxed> install mpv on the android device and play them stream using it [22:52:17 CET] <nicolas17> you could have started at that [22:52:19 CET] <nicolas17> http://xyproblem.info/ [22:52:36 CET] <relaxed> it's called mpv-android in the playstore [22:54:22 CET] <grosso> ok, I will try it... but, any players that I have tested, including vlc don't have that "listen" option so can't receive my stream [22:57:01 CET] <relaxed> is that necessary? Can you start the stream, then play it? [22:57:05 CET] <grosso> so you think mvp-android should receive this stream? so far only ffmpeg and ffplay have worked for me [22:57:18 CET] <nicolas17> where would that ffmpeg process run anyway? [22:57:45 CET] <relaxed> mpv uses ffmpeg's demuxers/codecs, so it should be fine [22:58:06 CET] <nicolas17> relaxed: yeah but can you tell mpv-android to "listen"? [22:58:25 CET] <relaxed> paging JEEB [22:59:44 CET] <grosso> the parameter "listen" tells the player act as a server... it is absolutely necessary since the sending end expects a server [22:59:52 CET] <relaxed> gotcha [23:05:59 CET] <grosso> nicolas17 i'm running ffmpeg in android... there are lots of compilations, but what don't exist is ffplay which is exactely what I need... so what I'm trying to do is receive with ffmpeg and output to wathever that I can then see on android... a video or a sequence of pictures, even raw images... but it has to be "realtime" that is, i need to see it as it comes to ffmpeg [23:09:24 CET] <grosso> I will try this: ffmpeg -i 'rtsp://192.168.1.44:8559/live.sdp -c copy output.mkv [23:28:23 CET] <grosso> that command actually works, but... 1) no audio (original audio is AAC 1 channel) 2) I still have this problem: file is generated when sending end stops, rather than been generated "realtime" [00:00:00 CET] --- Sat Nov 3 2018

1 0

ffmpeg-devel.log.20181101
by burek 02 Nov '18

02 Nov '18

[00:07:04 CET] <atomnuker> BBB: I'm getting a stack smash abort, and I've already wrapped all cdfs in cdf.h using ALIGN(, 16); [00:07:19 CET] <atomnuker> is there somewhere in the decoder where the CDF isn't taken from that array but is on stack? [00:12:19 CET] <atomnuker> seems to be msac_decode_bool_adapt() [00:27:32 CET] <atomnuker> ALIGN doesn't seem to be working as I'd expected, my code only works if I create a temporary cdf and then copy the updated one back [00:27:54 CET] <atomnuker> otherwise I get a desync [01:10:16 CET] <BBB> atomnuker: not quite sure, but yes, bool uses one on the stack and probably shouldnt use the update_cdf anyway (it can use an optimized update which is purely boolean) [01:36:37 CET] <cone-330> ffmpeg 03Philip Langdale 07master:4a976200d785: avcodec/vdpau: Initialise driver version variable explicitly [01:55:55 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:ea160f02137e: avutil/pixfmt: Document chroma plane size for odd resolutions [01:55:56 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:31262860ccd3: swresample/swresample: Fix input channel count in resample_first computation [01:55:57 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:922a29142e77: avcodec/diracdec: Prevent integer overflow in intermediate in global_mv() [01:55:58 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:d5c48e8536d0: avcodec/dirac_dwt_template: Fix several integer overflows in horizontal_compose_daub97i() [01:55:59 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:ee5b42ce2f2f: avcodec/diracdec: Change frame_number to 64bit as its a 32bit from the bitstream and we also have a -1 special case [01:56:00 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:c07d568603d3: avcodec/diracdec: Check slice numbers for overflows in relation to picture dimensions [01:56:01 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:c1a3212d45a9: avcodec/diracdec: Check bytes count in else branch in decode_lowdelay() too [01:56:02 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:1e1c02c51b3b: avcodec/qtrle: Check remaining bytestream in qtrle_decode_XYbpp() [01:56:03 CET] <cone-330> ffmpeg 03Nikolas Bowe 07release/3.3:e2d052d5c832: lavc/svq3: Fix regression decoding some files. [01:56:04 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:0026d007428a: avformat/flvenc: Check audio packet size [01:56:05 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:0678985c14dd: avcodec/aacpsdsp_template: Fix integer overflow in ps_stereo_interpolate_c() [01:56:06 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:0ea1b62b9ac5: avcodec/mpegaudio_parser: Initialize poutbuf* [01:56:07 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:488c246bf565: avcodec/shorten: Check verbatim length [01:56:08 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:d16d851238f4: avcodec/shorten: Fix integer overflow in residual/LPC combination [01:56:09 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:298ca73ab14b: avcodec/shorten: Fix signed 32bit overflow in shift in shorten_decode_frame() [01:56:10 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:907699d1207f: avcodec/scpr: Check for min > max in decompress_p() [01:56:11 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:44ebea8657e2: avformat/rmdec: Fix EOF check in the stream loop in ivr_read_header() [01:56:12 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:24add16a3236: avformat/mlvdec: read_string() received unsigned size, make the argument unsigned [01:56:13 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:b80d7d4a0920: avformat/nsvdec: Do not parse multiple NSVf [01:56:14 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:dc8ec1a5bc85: avcodec/snowdec: Fix integer overflow with motion vector residual [01:56:15 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:347c50b0d4e7: avcodec/vb: Check for end of bytestream before reading blocktype [01:56:16 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:66331499a7d2: avcodec/hq_hqa: Check remaining input bits in hqa_decode_mb() [01:56:17 CET] <cone-330> ffmpeg 03Michael Bunk 07release/3.3:7266a6d23c89: examples: Fix use of AV_CODEC_FLAG_GLOBAL_HEADER [01:56:18 CET] <cone-330> ffmpeg 03Dale Curtis 07release/3.3:1acec9bbf55b: avformat/mov: Error on too large stsd entry counts. [01:56:19 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:78a1db00c4c4: avcodec/indeo4: Check dimensions in decode_pic_hdr() [01:56:20 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:5c91bb1881a8: avcodec/ra144: Fix undefined integer overflow in add_wav() [01:56:21 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:fa5498cb4dae: avcodec/h264_refs: Document last if() in ff_h264_execute_ref_pic_marking() [01:56:22 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:46875255bc95: avcodec/dvdsubdec: Avoid branch in decode_run_8bit() [01:56:23 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:3c3b437d7947: avcodec/shorten: Fix bitstream end check in read_header() [01:56:24 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:bf4026fcbee8: avcodec/zmbv: Update decomp_len in raw frames [01:56:25 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:7fb45ccf4ef5: avcodec/zmbv: Check that the decompressed data size is correct [01:56:26 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:43a80efee600: avcodec/mpeg4videodec: Fix undefined shift in get_amv() [01:56:27 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:6177ed2a9b70: avcodec/dvdsubdec: Sanity check len in decode_rle() [01:56:28 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:a9ce7cfc5949: avcodec/unary: Improve get_unary() docs [01:56:29 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:1e59d05e48f9: avformat/utils: Fix integer overflow in discontinuity check [01:56:30 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:a1f4fe165a40: avformat/utils: Never store negative values in last_IP_duration [01:56:31 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:caddb7890aa6: avcodec/ra144: Fix integer overflow in add_wav() [01:56:32 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:65710b7e6c89: avcodec/h264_cavlc: Check mb_skip_run [01:56:33 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:947f7af04821: avcodec/mpeg4videodec: Fix typo in sprite delta check [01:56:34 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:98444e0222ed: avcodec/jpeg2000dec: Fix off by 1 error in JPEG2000_PGOD_CPRL handling [01:56:35 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:2405af5081a9: avcodec/msrle: Check that the input is large enough to contain a end of picture code [01:56:36 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:72e1c43d6df5: avutil/integer: Fix integer overflow in av_mul_i() [01:56:37 CET] <cone-330> ffmpeg 03Michael Niedermayer 07release/3.3:6fe9f36dd994: Update for FFmpeg 3.3.9 [02:14:54 CET] <Gramner> fwiw shrx is faster than shifting by cl [02:16:13 CET] <Gramner> shr r,cl is 2/2 (3 µops), shrx r,r,r is 1/0.5 (1 µop) [02:17:47 CET] <Gramner> and no, that doesn't make any sense. just intel things. [02:28:02 CET] <jamrial> worth the extra byte then [02:41:30 CET] <BBB> hm& bmi2 [02:41:33 CET] <BBB> whats that [02:41:36 CET] <BBB> do I have that? [02:41:52 CET] <BBB> Ive always wanted a shift instruction that did not need cl [02:42:08 CET] <Gramner> bmi2 == avx2 in practice [02:44:41 CET] <Gramner> as in all cpus with avx2 have bmi2, and all cpus with bmi2 has avx2 [02:48:28 CET] <Gramner> since it uses vex encoding those budget cpus with AVX fused off for marget segmentation reasons also lack bmi [03:45:45 CET] <nicolas17> when using a big image sequence as input, often I see neither CPU nor I/O are saturated, because ffmpeg reads one file (CPU idle) and then decodes it (I/O idle) [03:46:16 CET] <nicolas17> if I do "cat *.jpg > /dev/null" in the background, overall performance improves [03:46:24 CET] <nicolas17> would it make sense to add pre-fetching to ffmpeg itself? [04:12:47 CET] <nicolas17> https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/HEAD:/libavformat/img2dec.c "reading blob failed"? [04:14:40 CET] <nicolas17> https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/img2dec.c#L130 is this doing a binary search? [04:39:39 CET] <philipl> Because I like to double down on bad life choices, I'm now trying to implement bwdif in cuda. It seems mostly working but I clearly see small artifacts in sample playback that aren't in the CPU version. Can't see any errors in the algorithm at this point. [04:40:22 CET] <nicolas17> looks like it's not quiiite a binary search but similar idea [04:40:31 CET] <nicolas17> I'm not sure if it's wise, what if there are gaps in the sequence? [11:15:05 CET] <durandal_1707> is there tool to display raw bayer pixels, you just specify dimensions and pattern? [13:52:33 CET] <cone-316> ffmpeg 03Paul B Mahol 07master:d98fb558be9c: doc/filter: add some loop filter trivial examples [15:56:37 CET] <cone-316> ffmpeg 03Michael Niedermayer 07release/3.4:cb74b3ff27d5: Update for 3.4.5 [18:16:57 CET] <atomnuker> BBB: opened a pr for the wip simd [18:17:40 CET] <atomnuker> help on why it corrupts other CDFs welcome, not even ALIGN(, 128) helps [18:27:10 CET] Action: durandal_1707 *cries* [18:29:10 CET] <atomnuker> is it that cold over there already for electricity bills to do that? [18:29:33 CET] <durandal_1707> how did you guess? [18:34:09 CET] <atomnuker> s-same -_- [18:45:07 CET] <kurosu_> atomnuker, tried your dav1d msac branch, it is slower (than pure C) :( [18:45:31 CET] <kurosu_> haven't looked at your branch, but msac_update_cdf is already vectorized [18:45:36 CET] <kurosu_> bbl [18:45:43 CET] <durandal_1707> clang? [18:52:02 CET] <kurosu_> gcc 8.2/win64/msys2 [18:52:37 CET] <kurosu_> I wonder what would happen if I specified -mavx2 or stuff; I have already seen bmi insns in bitstream functions, in that case [18:53:19 CET] <jamrial> how did you bench? atomnuker didn't add a checkasm test yet [18:53:37 CET] <kurosu_> just runtime on a sequence [18:55:14 CET] <kurosu_> 9.7s -> 10.5s [18:57:29 CET] <jamrial> kurosu_: yeah, it is slower for me as well [18:58:04 CET] <jamrial> 59fps to 56fps on this one 1080p sample [18:58:37 CET] <BBB> pand m1, m2 [18:58:37 CET] <BBB> pxor m2, [inv_16bit] [18:58:38 CET] <BBB> pand m0, m2 [18:58:40 CET] <BBB> atomnuker: pandn [18:59:01 CET] <BBB> pandn m0, m2, m0 is the same as pxor m2, 0xfffffff.. and then pand m0, m2 [18:59:11 CET] <BBB> memory access is expensive [19:01:12 CET] <BBB> you could also consider a mask to only update the bits within n_syms [19:01:18 CET] <BBB> that way you dont need the weird alignment [19:04:17 CET] <jamrial> psrlw using memory seems to read 16 bytes from it, and i'm not sure if that's what you inteded [19:05:23 CET] <cone-796> ffmpeg 03BIGLER Don (Framatome) 07master:cb74c33106d0: avdevice/decklink_dec: fix codec_tag of RGBA formats [19:05:23 CET] <cone-796> ffmpeg 03Anton Platov 07master:295fd12d17f3: avdevice/libndi_newtek_dec: add extra_ips option to libndi_newtek allowing use remote network sources [19:09:51 CET] <BBB> atomnuker: want me to play around with that? [19:10:09 CET] <BBB> atomnuker: and the C code works, right? [19:13:08 CET] <atomnuker> kurosu_: yeah, it does 2 function calls instead of one and 2 memcpys [19:14:40 CET] <BBB> thats why I suggested inline sse2 [19:14:43 CET] <BBB> so the function call is gone [19:15:32 CET] <atomnuker> well, the functions will get bigger once updating is done at the same time as decoding [19:15:48 CET] <atomnuker> so inlining won't really be doing good for the size at that point [19:16:13 CET] <atomnuker> BBB: go and play with it, I tried pandn and it didn't work right away [19:17:18 CET] <BBB> sure :) [19:33:27 CET] <thardin> does ffmpeg needs hosting for things? [19:33:53 CET] <thardin> local computer club has some 13 Gbps of unused bandwidth to spare for free software projects [20:02:21 CET] <cone-796> ffmpeg 03Anton Platov 07n3.4.5:HEAD: avdevice/libndi_newtek_dec: add extra_ips option to libndi_newtek allowing use remote network sources [20:14:56 CET] <durandal_1707> tpad is 411th filter [20:23:46 CET] <philipl> BtbN: time to pester you about the yadif filter :-) [20:24:39 CET] <BtbN> I haven't forgotten it, just insanely short on time at the moment [20:26:02 CET] <BtbN> philipl, I'm also not sure who is responsible for yadiff to ok those changes. [20:26:40 CET] <philipl> michaelni presumably? [20:26:41 CET] <BtbN> it's just moving a bunch of functions without changes, so should be ok in general [20:26:51 CET] <philipl> It seemed wasteful to duplicate them. [20:27:01 CET] <philipl> I can also consolidate bwdif to reuse them with a small addition. [20:27:06 CET] <BtbN> yeah, it's the right approach, just needs to be okayed [20:27:35 CET] <JEEB> &41 [20:27:42 CET] <durandal_1707> michaelni: why cant release be named Carl? you are not friendly and ignore lots of votes [20:29:11 CET] <jamrial> durandal_1707: because they are troll votes [20:30:22 CET] <BtbN> philipl, I'm starting to wonder if we should put all those CHECK_CU stuff in some central place. [20:30:35 CET] <BtbN> But that's something for another patch at some point [20:30:53 CET] <philipl> Yeah. It can be done - just a little tedious to ensure naming is consistent, etc. [20:31:20 CET] <durandal_1707> jamrial: troll votes? you can not be so serious [21:07:35 CET] <durandal_1707> it is 11, 2018 and there are still AVERROR_EOF bugs in ffmpeg code [21:09:03 CET] <philipl> There is no end to EOF. [21:10:43 CET] <gnafu> EOEOF [21:10:57 CET] <gnafu> "There was a coder, had a bug. EOEOF." [23:11:06 CET] <cone-124> ffmpeg 03Marton Balint 07master:6a034adf7516: avformat/ftp: return AVERROR_EOF for EOF [23:19:55 CET] <cone-124> ffmpeg 03Marton Balint 07release/4.0:00e7e4b18806: avcodec/libx264: remove FF_CODEC_CAP_INIT_THREADSAFE flag [23:19:56 CET] <cone-124> ffmpeg 03Marton Balint 07release/4.0:ebc01c8f6dfc: avformat/ftp: return AVERROR_EOF for EOF [23:27:07 CET] <akravchenko188> jkqxz: Hello. I have sent updated patches for hwcontext_amf & amfenc. Could you please review them and apply if they are ok? [23:55:30 CET] <philipl> BtbN: heh. That's a landmine of a function to add :-) [23:56:05 CET] <BtbN> Yeah, I definitely do not want the full enum. Most of that is useless [23:56:16 CET] <BtbN> CU_DEVICE_ATTRIBUTE_ECC_ENABLED... [00:00:00 CET] --- Fri Nov 2 2018

1 0

ffmpeg.log.20181101
by burek 02 Nov '18

02 Nov '18

[02:51:30 CET] <p1nky> looking at this .. https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20inp… [02:51:43 CET] <p1nky> anyone know of a way to create a dynamic version using an mpeg-ts as input ? [02:52:26 CET] <p1nky> dynamic meaning not needing to specify how many inputs there are.. just all programs on the TS in their own box and as many boxes as needed within a given size where they all fit [02:53:28 CET] <p1nky> it would also be neat to do the same but be able to just extract one frame out of each and write a png [02:55:38 CET] <nicolas17> hmm programs from a ts... [02:56:15 CET] <nicolas17> I mean you can write a script to generate an appropriate filtergraph if you tell it how many inputs you have [02:56:48 CET] <nicolas17> but I never had to deal with multi-program transport streams [02:56:53 CET] <p1nky> yeah i was thinking if i had to do that i would .. probably having to run something to peek at the ts first and get that info [02:56:56 CET] <nicolas17> so I don't even know how to tell how many programs are there [02:57:33 CET] <p1nky> i guess i can start with them static .. the examples all show multiple input files [02:57:43 CET] <nicolas17> also what do you do if the number of programs is a prime number? :) [02:57:54 CET] <p1nky> i am hoping that my one input can be the ts and i can reference them [02:57:57 CET] <p1nky> hah [02:58:21 CET] <p1nky> probably just round to an even # [03:01:35 CET] <poutine> p1nky, I know ts duck allows for some stuff like that [03:02:50 CET] <p1nky> oh neat, i ran across that the other day [03:02:55 CET] <p1nky> but didn't think it did stuff like mosaics [03:10:14 CET] <p1nky> oh i guess you were saying to inspect the ts [03:10:29 CET] <p1nky> i wonder how you even reference a ts program in a filtergraph [03:11:56 CET] <nicolas17> looks like stream specifiers have program_id [03:12:23 CET] <nicolas17> in the wiki example, [0:v] is a stream specifier, meaning "video stream of input file 0" [03:12:46 CET] <p1nky> i am trying [0:v:0] [03:12:59 CET] <p1nky> working on an output issue now :) trying to multicast the output [03:13:57 CET] <nicolas17> weird, docs say it should be v:0 to get the video stream of input 0, maybe it supports 0:v for compatibility? [03:14:15 CET] <nicolas17> anyway [03:14:23 CET] <nicolas17> for programs, it seems to be p:program_id[:stream_type[:stream_index]] [03:22:20 CET] <p1nky> ooh .. not sure if this is it or not https://ffmpeg.org/ffmpeg-filters.html#toc-xstack [03:24:13 CET] <garyserj> I have an mkv file, video(AVC) / Audio(Opus). How do I convert it to mp4 ? [03:30:06 CET] <ariyasu> ffmpeg -i input.mkv -vcodec copy -acodec copy output.mp4 [03:31:10 CET] <nicolas17> AVC is h264? [03:32:08 CET] <garyserj> Is -c copy the same as -acodec copy -vcodec copy ? [03:33:13 CET] <furq> yes [03:33:18 CET] <furq> and subtitles as well [03:33:48 CET] <furq> but you probably want -c:v copy -c:a aac [03:36:06 CET] <garyserj> thanks yeah that worked well, that audio codec. [10:59:18 CET] <termos> I want to create HLS segments with epoch time and segment numbers in the filenames. Tried using second_level_segment_index hls flag but %d is then starting to count from epoch time for when I started transcoding. Any way to make this number start at 0? [11:11:32 CET] <garyserj> if I do ffmpeg -i a.mp4 -vf "transpose=1" b.mp4 Should I do it with -c copy? And how would I rotate it 90 degrees the other way instead? [11:11:59 CET] <ariyasu> no [11:12:11 CET] <ariyasu> you can't copy the video stream if you are using transpose [11:12:16 CET] <ariyasu> it needs to be re-encoded [11:12:33 CET] <ariyasu> you can do -acodec copy though, no need to re-encode the audio [11:13:37 CET] <ariyasu> ffmpeg -i a.mp4 -vf "transpose=1" -vcodec libx264 -acodec copy b.mp4 [11:13:58 CET] <ariyasu> somthing like that, you may want to use some x264 options though but thats the basic line [11:31:03 CET] <BtbN> If the mp4 rotate metadata is enough, you can add that without transcoding [11:31:10 CET] <BtbN> If it works depends entirely on the player [12:05:18 CET] <th3_v0ice> I have a problem that is making me question my knowledge and sanity. When the program starts it uses 40mb of RAM it then decodes 100 frames and store them in vector<AVFrame> using now 400mb of RAM. So far so good. Stored frames are processed trought some filter (scale) and then sent off to the encoder. While encoding, memory doubles, which I think is fine, so 800mb. Closing the encoder and [12:05:18 CET] <th3_v0ice> filter free's the 400mb of memory. When I start av_frame_free(frame) for each frame from the vector, memory stays at 400mb+. The problem is that this memory stays constant, meaning my program is not increasing this 400mb memory usage even when I decode next 100 frames. What could I be doing wrong? [12:34:30 CET] <rolandaswb> hello hoe to run ffmpeg comand line in python? [12:36:00 CET] <rolandaswb> what is command line for adding audio to video file? [13:15:05 CET] <M6HZ> Hello, it looks like there is an issue with the way ffmpeg handles the options "-referer" and "-user_agent". These options seem to stick only for the first http request, if there is a redirection they seem to be dismissed. [13:18:06 CET] <JEEB> quite possible. check the trac issue tracker for similar issue(s) and either comment on one of them or create a new issue with specifics. [13:20:10 CET] <M6HZ> JEEB, ok, thanks. [15:49:09 CET] <jngk> why can I find files for an encoder under libavcodec, when that encoder isn't listed by `configure --list-encoders` [15:49:44 CET] <jngk> eg. libavcodec/truemotion{1,2}.c [17:40:47 CET] <Dudemanguy> What's the fastest way to demux a large number of audio files? [17:40:57 CET] <Dudemanguy> Using this: https://pastebin.com/raw/QF8yueC7 I can do ~12000 files in about 10 seconds [17:41:11 CET] <Dudemanguy> It's not bad, but is it possible to be more efficient? [17:45:03 CET] <Dudemanguy> oh and I'm not really sure if I'm guessing the input format name right. They're opus files and I think the decoder name is ogg, but I'm not 100% sure on that [17:52:16 CET] <durandal_1707> ogg is container, so it can be either demuxer or muxer [17:53:27 CET] <Dudemanguy> does ffmpeg not have a demuxer for opus containers? [17:53:38 CET] <Dudemanguy> I don't see it in allformats.c at least [18:00:05 CET] <relaxed> Dudemanguy: it's covered by the ogg demuxer [18:00:26 CET] <iive> is there such thing as opus container? I thought they use ogg, because it is done by the same people. [18:01:27 CET] <Dudemanguy> ah you're right it's an og container [18:01:31 CET] <Dudemanguy> ogg* rather [21:37:35 CET] <retal> Hi guys, I am tryng live transcoding and restream video to RTMP H265 but receiving error: Video codec hevc not compatible with flv. But same commands works if I tryng: -f rtsp rtsp:// [21:38:46 CET] <JEEB> rtsp is not rtmp [21:38:49 CET] <JEEB> rtmp is adobe's thing [21:39:02 CET] <JEEB> rtmp is basically FLV over a protocol [21:39:13 CET] <JEEB> and you will have to ask adobe to standardize HEVC in it [21:39:42 CET] <JEEB> there were requests of using some made-up ID (there's just a number for each codec in FLV), but since it's not official adobe could (in theory) just use that number for something else [21:39:59 CET] <JEEB> but most likely at this point even adobe is generally just wanting to ignore it had RTMP :P [21:40:15 CET] <JEEB> but yes, pester Adobe about HEVC if you want it in FLV or RTMP [21:44:06 CET] <JEEB> so when HEVC will be in Annex E of https://wwwimages2.adobe.com/content/dam/acom/en/devnet/flv/video_file_form… [21:44:11 CET] <JEEB> then it will work :P [21:44:42 CET] <Mavrik> What's the usecase for RTMP these days? [21:45:33 CET] <JEEB> mostly feeding to servers, although I'm seeing things being able to take MPEG-TS (over UDP etc) or fragmented MP4 over HTTP as well [21:45:40 CET] <retal> JEEB, i know rtsp and rtmp difference :) I just made litle research looks like Adobe doesn't support 265 officially. Thank you [21:45:54 CET] <JEEB> retal: that's literally what I just told you :P [21:46:00 CET] <JEEB> but great that you figured it out as well [21:46:01 CET] <retal> :) [21:46:56 CET] <durandal_1707> lol [21:48:24 CET] <retal> JEEB, I surprised H265 5-6 years old codec [21:49:05 CET] <retal> maybe we need wait 10 years more [21:49:06 CET] <JEEB> that's because adobe gave up on RTMP for end user streaming :P [21:49:27 CET] <JEEB> as I said, > but most likely at this point even adobe is generally just wanting to ignore it had RTMP [21:49:54 CET] <JEEB> (and FLV while at it) [21:50:32 CET] <JEEB> even that specification I linked first and foremost (and most of the document) documents F4V, which is flash's variant of fragmented ISOBMFF (aka "mp4") [21:50:45 CET] <JEEB> s/flash/adobe/ [21:50:55 CET] <JEEB> and then an annex is given to FLV [21:51:29 CET] <retal> thank you [21:55:24 CET] <friki> Hi. I'm trying to write build and test a DASH example with TS segments. Testing with ffplay I got "stream.mpd: Invalid data found when processing input". My mpd example looks like: https://pastebin.com/aB9YxXuk [21:59:01 CET] <friki> The problem seems to be in the mpd, because ffplay doesn't request the segments. I'll appreciate if someone can take a look at my example or share a working one. I've only found examples using mp4 segments, btw [22:09:14 CET] <p1nky> hmm when trying to extract frames like this from an MPEG-TS [22:09:20 CET] <p1nky> $ ffmpeg -i rtp://@224.0.1.2:5004 -vf select=1 -frames:v 1 -y x.png [22:09:29 CET] <p1nky> is there a way to be sure they are 'clean' ? [22:10:04 CET] <furq> define clean [22:10:43 CET] <furq> i'm pretty sure the decoder will just discard everything before the first keyframe [22:10:52 CET] <furq> so select is doing nothing there [22:11:10 CET] <p1nky> hm yeah i'm also having a ton of problems being able to select by program id [22:11:32 CET] <JEEB> ffmpeg.c has a -map selector for that [22:11:43 CET] <JEEB> or well, it's by PID I think [22:11:48 CET] <furq> can you use pids in filter input labels [22:11:50 CET] <JEEB> still helpful I would say [22:12:20 CET] <furq> i assume it's just the same stream specifier as -map [22:12:24 CET] <p1nky> yeah things i am reading say p:1:v for program 1 [22:12:39 CET] <p1nky> but program isn't pid right? ffmpeg says program # 1 and such [22:12:42 CET] <p1nky> i am sure thats not the actual pid [22:12:46 CET] <p1nky> although maybe it is.. [22:13:17 CET] <furq> yeah that's the actual pid [22:13:25 CET] <furq> https://trac.ffmpeg.org/wiki/Map#Example8 [22:13:25 CET] <JEEB> programs and stream ids are separate [22:14:20 CET] <JEEB> http://up-cat.net/p/7106af10 [22:14:51 CET] <JEEB> in this ffprobe output Program XXX is the program, the 0xYYY after the stream index is the stream ID [22:14:58 CET] <JEEB> which in case of mpeg-ts is the PID :P [22:15:49 CET] <furq> yeah in that example you could use p:1048:0 or i:0x111 [22:16:51 CET] <p1nky> ok nice thanks [22:17:07 CET] <p1nky> so select=1 is not actually selecting anything [22:17:11 CET] <furq> or p:1048:v or even p:1048:v:0 [22:17:19 CET] <furq> select=1 is selecting everything [22:17:40 CET] <p1nky> so -vf select p:1048:v in your example? [22:17:45 CET] <p1nky> i was trying that as an argument to select [22:17:47 CET] <furq> oh [22:17:50 CET] <furq> no that's not what select does [22:17:51 CET] <p1nky> confused as to -map [22:18:28 CET] <furq> if you want to filter one stream then you'd use something like -filter_complex [i:0x111]yadif [22:18:49 CET] <furq> select is for conditionally selecting frames [22:20:27 CET] <p1nky> hm i thought there was a paste thing in the topic.. [22:20:30 CET] <p1nky> but i don't see it [22:20:41 CET] <furq> any pastebin is fine [22:21:12 CET] <p1nky> i only know of pastebin and its down :) [22:21:23 CET] <p1nky> Program 1 [22:21:24 CET] <p1nky> Stream #0:7: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(top first), 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc [22:22:17 CET] <p1nky> should this work? -filter_complex '[p:1]' [22:23:15 CET] <furq> probably p:1:v or 0:p:1:v [22:23:40 CET] <p1nky> with [] around it? -filter_complex '[p:1:v]' [22:23:44 CET] <p1nky> [AVFilterGraph @ 0x5556b6dab380] No such filter: '' [22:23:51 CET] <furq> well yeah you need to put an actual filter in there [22:23:57 CET] <furq> otherwise just use -map [22:27:09 CET] <p1nky> ah ok trying -map 0:0 and i still seem to be getting random programs every time i run it [22:27:29 CET] <furq> well yeah use -map 0:p:1:v [22:28:41 CET] <p1nky> ah! thanks! [22:28:50 CET] <p1nky> that is reliably giving me just one program [22:29:13 CET] <p1nky> but its a crap frame a lot of times, and sometimes it matches no stream [22:31:53 CET] <p1nky> i need to figure out xstack at some point .. [22:32:01 CET] <p1nky> with one box per program [22:35:35 CET] <p1nky> anyway thanks a lot for the help! [22:35:54 CET] <p1nky> i need to do some reading on MPEG-TS i suppose [22:49:28 CET] <friki> p1nky: check for missing frames a the original video with ffprobe (check dst/pst/duration values) [22:50:41 CET] <p1nky> its a satellite stream [22:50:50 CET] <friki> a droped keyframe, may be? [22:51:27 CET] <friki> try recording a sample to disk with "-codec copy" [22:51:36 CET] <p1nky> receiving it with a DVB-s receiver and then using dvbstream to multicast the whole TS [22:55:22 CET] <friki> Can ffmpeg read stream from your receiver? ffmpeg -i yourReceiver -codec copy -t 10 record.ts [22:55:45 CET] <friki> I've used "tzap" to receive TDT [22:56:23 CET] <p1nky> yeah reads it fine, prints all 8 PIDs on this TS [22:56:37 CET] <p1nky> vlc always looks fine [22:56:48 CET] <friki> TDT => DTT [22:56:50 CET] <p1nky> i am sure theres noise that mpeg deals with in some way [22:57:22 CET] <p1nky> when actually watching video, but seems like this method is "get the very next frame no matter its quality and output a png of it" [22:58:55 CET] <p1nky> various messages .. PES packet size mismatch, RTP missed 19 packets, jitter buffer full, error while decoding MB 0 0, bytestream 28483 .. corrupt input packet in stream 0 [22:59:48 CET] <friki> i bet for signal reception problems [23:00:03 CET] <p1nky> this is a big stream, 30Msps QPSK .. 8 1920x1080 video streams with multiple audio each [23:00:37 CET] <p1nky> well yeah its never going to be perfect even if i had a bigger dish .. but between FEC at the DVB level and whatever is in mpeg to handle it .. [23:00:42 CET] <p1nky> actually watching streams off it looks perfect [23:00:50 CET] <p1nky> but apparently not so easy to just capture an arbitrary frame [23:01:25 CET] <p1nky> really i don't care a ton about capturing frames, what i want to do is use xstack to make a grid of every program in the stream at a reduced frame rate [23:01:40 CET] <p1nky> so that i can easily monitor each [23:02:28 CET] <friki> is an interaced signal. do you deinterlace it before export png? [23:02:39 CET] <p1nky> oh, i don't think so :) [23:02:47 CET] <p1nky> although sometimes it looks totally perfect.. [23:03:01 CET] <p1nky> other times its cut off at a various point in the stream i suppose and some portion is gray [23:03:09 CET] <friki> test with: -vf 'yadif' [23:04:11 CET] <p1nky> ffmpeg -i rtp://@224.0.1.2:5004 -map 0:p:1:v -vf select=1 -frames:v 1 -vf 'yadif' -y x.png [23:04:20 CET] <p1nky> nope still nasty most of the time [23:04:43 CET] <p1nky> sometimes fails entirely with: Stream map '0:p:1:v' matches no streams. [23:05:15 CET] <p1nky> i assume because theres only a short window of time where its trying to capture a packet with that PID and if one doesn't come it stops [23:05:36 CET] <p1nky> i think potentially if the image isn't moving theres no packets coming? or very few .. [23:05:39 CET] <friki> if you can share a recorded sample i can check for missing frames using ffprobe [23:06:17 CET] <p1nky> let me try to do that .. but need to get back to something, am at work :( hehe [23:07:30 CET] <p1nky> what args for ffprobe? [23:09:26 CET] <friki> i'll start with: ffprobe -i sample.ts -show_packets -print_format csv > sample.csv [23:10:30 CET] <friki> it will write a line for each packet. I'll check for "jumps" in pts column [23:11:46 CET] <p1nky> hm interesting .. anyway i recorded a 100M sample by just running dumprtp > x.mpg .. if you have a way for me to send it and you're really interested, i can :) [23:11:49 CET] <friki> "jump" -> unexpected big gap between two corelative video frame pts values. Gap should match packet duration [23:11:50 CET] <p1nky> which column is that? [23:12:03 CET] <p1nky> ah pastebin works now [23:12:36 CET] <p1nky> exceeded 512k :) [23:12:51 CET] <p1nky> for pastebin [23:15:11 CET] <friki> column 4 is pts [23:16:33 CET] <friki> column 10 is duration (usually same value) [23:16:55 CET] <p1nky> https://pastebin.com/gwFXxRh2 [23:17:15 CET] <p1nky> col 10 is all N/A [23:19:12 CET] <friki> Only 29 frames for "video,0" stream... None of them are key frame [23:19:47 CET] <p1nky> some of these right now are liable to be colorbars, so not moving at all [23:19:52 CET] <p1nky> i suspect that has something to do with it [23:19:53 CET] <friki> in fact there are no K frames in any video stream [23:20:07 CET] <p1nky> that was a 100mb file but i had to only take some of the lines [23:20:10 CET] <p1nky> let me grep for video,0 [23:20:31 CET] <p1nky> how do you tell a key frame? [23:20:42 CET] <p1nky> 716 lines for video,0 [23:20:45 CET] <friki> 'K' in the last column [23:21:11 CET] <p1nky> oh yeah none for video,0 in the whole file [23:21:40 CET] <p1nky> plenty for others .. K_ anyway but actually all audio [23:23:38 CET] <friki> May be the encoder is using a "key lines wave" (i don't recall the correct naming). It's a technique to avoid transmission bandwith peaks [23:24:07 CET] <friki> i mean, you should check the gap between column 4 values. should be stable [23:24:49 CET] <friki> If FPS is 50, the gap between pts should be 0.02 [23:24:59 CET] <friki> 1/$FPS ;-) [23:25:18 CET] <p1nky> its 29.97 fps according to ffmpeg [23:25:28 CET] <p1nky> Stream #0:8: Video: h264 (High), yuv420p(top first), 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc [23:25:56 CET] <friki> so... 0.033-0.034 :-P [23:25:58 CET] <p1nky> but yeah grep video,0 x.csv | cut -d, -f4 [23:26:02 CET] <p1nky> gap is usually about 1-2 [23:26:08 CET] <p1nky> heres one where its 10 [23:26:25 CET] <p1nky> gah, theres one thats 6174545206 to 6174537698 [23:26:30 CET] <friki> hehe, you got it [23:27:06 CET] <friki> BTW: 5th column is decimal value. Same data as col 4 [23:27:07 CET] <p1nky> thanks for all the help :) [23:27:17 CET] <p1nky> ah nice [23:27:39 CET] <friki> a pleasure [23:28:02 CET] <friki> interesting transmission mode without keyframes. love it [23:29:25 CET] <p1nky> :) [00:00:00 CET] --- Fri Nov 2 2018

1 0

ffmpeg-devel.log.20181031
by burek 01 Nov '18

01 Nov '18

[00:00:18 CET] <atomnuker> I haven't figured out how to do the part afterwards, where for i >= val you go the other way [00:04:52 CET] <BBB> ty! [00:05:15 CET] <j-b> atomnuker: \o/ [00:05:19 CET] <BBB> well I think you do both right? [00:05:32 CET] <BBB> and then you mask them based on val [00:05:34 CET] <BBB> and then you or them [00:05:40 CET] <BBB> thats how I would do it at least [00:06:26 CET] <BBB> but gramner is better at this than me [00:09:13 CET] <atomnuker> yeah, I suppose I could splat the val across a reg, then run a compare, and convert the output of that into a mask [00:10:07 CET] <atomnuker> then using that I could combine the 2 paths [00:10:16 CET] <BBB> you can also do what I did in cdef/lr [00:10:33 CET] <BBB> have a 64 byte constant times 32 db 0xff times 32 db 0x0 [00:10:37 CET] <BBB> and then index into that using val [00:10:44 CET] <BBB> its almost the same [00:11:08 CET] <cone-547> ffmpeg 03Mark Thompson 07master:edcdf3512376: cbs_vp9: Ensure that reserved zero bits are actually zero [00:11:28 CET] <BBB> but what yousaid works also, yes [00:21:28 CET] <atomnuker> https://0x0.st/sIHr.jpg [00:21:36 CET] <atomnuker> yep, gimp still works even though I've never used it [00:43:22 CET] <kierank> durandal_1707: sucks [00:43:25 CET] <kierank> adding bayer to swscale [00:43:27 CET] <kierank> mess on top of mess [01:11:14 CET] <cone-547> ffmpeg 03James Almer 07master:ec440339b2cd: avcodec/cbs_vp9: keep track of reference frames [01:11:15 CET] <cone-547> ffmpeg 03James Almer 07master:7aaf092f8ce8: avcodec/cbs_vp9: store profile in the private context [01:13:04 CET] <jamrial> jkqxz: only thing that remains would be to add frame_type to VP9ReferenceFrameState [01:14:03 CET] <jamrial> it's not contemplated in the spec, but without it, an avparser can't feasibly set pict_type for frames with show_existing_frame = 1 [01:14:54 CET] <BBB> show_existing_frame is by definition an inter frame, right? [01:14:55 CET] <jamrial> the decoder for example keeps avframe references instead of a custom struct like VP9ReferenceFrameState, which has that and more [01:15:00 CET] <BBB> its what vc1 calls a pskip frame [01:15:11 CET] <BBB> it copies the whole reference with no motion or residual [01:15:13 CET] <jamrial> it can point to invisible frames, which can be intra_only [01:15:14 CET] <BBB> thats still inter:) [01:15:23 CET] <BBB> the reference of the frame is intra [01:15:27 CET] <BBB> but the frame itself is inter [01:15:35 CET] <BBB> its just specially coded [01:15:56 CET] <BBB> or you want to point out keyframe search poiints using this? [01:16:09 CET] <BBB> I would say the keyframe search point is the invisible frame, not the show-existing-frame making it visible [01:16:10 CET] <jamrial> show_existing_frame is literally a 1 byte frame that says "present this old frame" [01:16:56 CET] <jamrial> in av1 it certainly can be intra, as it's the case with delayed RAPs [01:26:38 CET] <BBB> right, I know [01:26:46 CET] <BBB> I just wonder why you want to annotate them as such [01:26:51 CET] <BBB> you cant actually seek to the show-existing-frame [01:27:03 CET] <BBB> you can seek to the invisible frame, skip the next few frames that reference frames beforeit [01:27:06 CET] <BBB> and then show it [01:27:12 CET] <BBB> and then go from there [01:27:18 CET] <BBB> but the seek is not to the show-existing-frame then [01:27:21 CET] <BBB> so ... [01:27:26 CET] <BBB> its complicated ;) [01:28:35 CET] <jamrial> BBB: i'm trying to port the parser to cbs, to export more fields than just profile [01:29:03 CET] <BBB> oh so its just metadata, not for seeking or anything input to the muxer? [01:29:04 CET] <jamrial> since reimplementing parsing code to handle super frames, keeping track of references and such is annyoning [01:29:08 CET] <jamrial> yeah [01:29:34 CET] <jamrial> so i want to set AVCodecParserContext.pict_type correctly for all frames [01:51:20 CET] <jkqxz> It seems like it depends on what you are going to use it for. [01:51:28 CET] <jkqxz> If you are looking at packets then the superframe containing an invisible key frame is obviously itself key - making the displayed actually-non-key frame the one that gets marked as key, while the later-displayed actually-key frame doesn't. [01:51:45 CET] <jkqxz> Whereas if you are looking at frames only then it's more logical for it to be the other way around. [01:52:02 CET] <jkqxz> It seems like the first interpretation is more useful, though, since that's what you actually want for seeking purposes? [01:53:46 CET] <jkqxz> (At least we don't have the wrangle with the wtf confusing thing in AV1 where the IDRalike frame only flushes the DPB when it's shown, not when it's decoded.) [02:48:22 CET] <jamrial> BBB, jkqxz: like this https://github.com/jamrial/FFmpeg/commit/c4dc8cffcdcaae3d489ed991fa7b267fba… [02:49:00 CET] <jamrial> pretty much the same as the av1 parser [02:55:38 CET] <BBB> yeah I guess [02:55:46 CET] <BBB> (still a cbs n00b) [02:55:49 CET] <BBB> (me) [03:01:36 CET] <jamrial> BBB: it's really neat and powerful. it parses everything for you [03:01:45 CET] <jamrial> you just then read some values and export metadata, or you change them then output it all as a new frame where the new value is neatly written without fucking up the bitstream [03:02:04 CET] <jamrial> props to jkqxz for writing it :p [03:04:59 CET] <jamrial> it's ideal for bitstream filters and parsers for obvious reasons, but also can be used in decoders to handle the header bitstream parsing. each frame's tile/slice data is also handed to you as an avbufferref [03:08:46 CET] <BBB> Ill play with it at some point [03:08:53 CET] <BBB> might be useful for the stream analyzer going forward [03:09:09 CET] <BBB> the amount of code in there right now related to header parsing is just awful [03:44:53 CET] <cone-547> ffmpeg 03ManojGuptaBonda 07master:4a6d5f3cadaa: avcodec/vdpau: Enable HEVC support for working Nvidia driver versions [11:52:42 CET] <cone-674> ffmpeg 03Paul B Mahol 07master:323c2cfd384a: avfilter: add (a)graphmonitor filter(s) [14:54:25 CET] <pasouza> hi, I'd like to know the right procedure for reimbursement request (GSoC mentor summit) [14:58:21 CET] <durandal_1707> write it to ML [15:00:26 CET] <Compn> include invoice [15:00:38 CET] <Compn> who you mentored, project worked on, etc [15:00:43 CET] <Compn> committed status [16:02:23 CET] <pasouza> ok, thanks! [16:04:00 CET] <pasouza> can I email the invoice privately? as it may contain personal info [16:40:01 CET] <durandal_1707> pasouza: do not include invoice, there are already siimilar mails available in archive [17:34:19 CET] <akravchenko188> jkqxz: hi, could you please answer on my comments in forum?. Probably I can remove amf logging on this stage, and decide later how to integrate it [17:36:16 CET] <akravchenko188> jkqxz: there are some customers, who need hw pipeline in FFMpeg (hwdecode -> scaler -> encode) and we have hwcontext_amf blocker for this [17:38:21 CET] <akravchenko188> jkqxz: it would be very nice if we try to speed up the integration process [19:08:24 CET] <durandal_1707> why is everything is so silent? [19:24:03 CET] <jamrial> it's spooky day [19:25:41 CET] <atomnuker> its cold [19:58:26 CET] <atomnuker> how do I right shift a gpr with a value from another gpr? shr only accepts constants [20:00:09 CET] <nevcairiel> they accept register CL [20:00:18 CET] <nevcairiel> and only that [20:01:01 CET] <atomnuker> register cl? [20:01:22 CET] <nevcairiel> its an 8-bit slice o f RCX [20:03:16 CET] <atomnuker> wow, that's bad design [20:07:04 CET] <jamrial> atomnuker: if you're targeting avx2, use shrx [20:08:19 CET] <nevcairiel> not sure using a super vex instruction is really worht it =p [20:08:24 CET] <nevcairiel> super long* [20:10:04 CET] <jamrial> it's like five bytes instead of three or four from shr. it's not that bad [20:10:31 CET] <jamrial> and you can use any gpr with it [20:12:49 CET] <atomnuker> no, I choose maximum efficiency and least readibility, I'll make it so cl (r3b) ends up as my final shift amount [20:34:04 CET] <jamrial> maximum efficiency and least readibility <-- the ffmpeg way :p [20:36:25 CET] <atomnuker> didn't need to do it in the first place, I remembered I didn't need variable right shifts after all [20:36:41 CET] <atomnuker> I'll just use psrlw [20:37:08 CET] <atomnuker> ...which I can't because it takes either a constant or a memory address [20:37:48 CET] <atomnuker> what were intel smoking, vectors are nice but then gprs are inflexible crap [20:38:08 CET] <atomnuker> shenzhen io's assembler had more flexible gprs than this [20:46:21 CET] <atomnuker> wait, psrlw m3, xm3 works? [20:50:27 CET] <BBB> atomnuker: yes! [20:50:30 CET] <BBB> atomnuker: but its slow [20:50:49 CET] <BBB> atomnuker: because its crosslane, it takes the first qword (!?!?!?!?!?) from xm3 to shift all words (in both lanes) in m3 [20:51:14 CET] <BBB> and if the qword >= 16, itll set everything to 0 [20:51:24 CET] <BBB> which, given that theyre words (possibly non-zero), is quite likely [20:52:38 CET] <atomnuker> I think it'll be faster than giving it a memory argument, plus I'd have to either bloat my grps to 7 (since r6m exists) or reserve some stack [20:57:54 CET] <BBB> for me, stack mem wa sfaster [20:57:59 CET] <BBB> probably because no cross-lane [21:00:54 CET] <atomnuker> can't do a dumb 1 to 1 look up table either, it requests fpic [21:01:18 CET] <atomnuker> honestly what the fuck intel, why did they not make it accept gprs [21:12:30 CET] <BBB> lea r6, [table] [21:12:45 CET] <BBB> mov r5d, dword [r6+idx*2] [21:12:46 CET] <BBB> or whatever [21:12:53 CET] <BBB> its annoying yes [22:22:16 CET] <kurosu_> shr design is older than you, respect your elders :p [22:28:58 CET] <atomnuker> I wonder if the 68000 had the same level of flag abuse and opaqueness [22:34:26 CET] <cone-330> ffmpeg 03Michael Niedermayer 07master:793a3e7bc909: doc/fate.texi: Mention that samples should be uploaded before pushing dependent commits [22:39:25 CET] <cone-330> ffmpeg 03Mark Thompson 07master:30fcc1093796: cbs_h264: Include SEI type names in trace output [22:39:26 CET] <cone-330> ffmpeg 03Mark Thompson 07master:2dee0679e9a8: cbs_h265: Add PTL parsing for Main 10 Still Picture profile [23:46:20 CET] <nevcairiel> atomnuker: for the record, which actual register actually backs r0-r14 in x86inc.asm code varies between platforms, because it prefers those it doesnt have to save, so r3b is only CL on unix64, on 32-bit its r1b and on windows64 its r0b, but i see you're not using it anymore anyway [23:50:38 CET] <nevcairiel> some code in avcodec even uses SHR with a CL argument, it has 3 different register assignments in there for the 3 cases [23:56:34 CET] <atomnuker> yeah, I thought so too [00:00:00 CET] --- Thu Nov 1 2018

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ffmpeg.log.20181031
by burek 01 Nov '18

01 Nov '18

[00:05:33 CET] <brimestone> is there an FFplay wrapper for objective c? [00:55:14 CET] <nicolas17> do .ts files have such thing as 'faststart'? [00:55:31 CET] <furq> no [00:56:01 CET] <nicolas17> do they not need it? can always play before getting the end? [00:56:59 CET] <furq> yes [00:57:08 CET] <furq> the whole point of mpegts is that it's streamable [00:57:45 CET] <furq> you can start reading from any arbitrary point in the file and it'll play [00:58:00 CET] <furq> the downside is that sending headers inband adds a lot of overhead and there's no seek index [00:59:47 CET] <nicolas17> wonder why it was chosen for HLS [01:00:43 CET] <nicolas17> I guess so that something can produce a long mpegts and something else can chop it up without having to generate new headers or anything? [01:01:16 CET] <furq> hls is usually used for livestreaming, so it's a pretty natural choice [01:01:37 CET] <furq> you can use fragmented mp4 as well but that whole format is a bolt-on afterthought [01:35:40 CET] <Zexaron> Helllo [01:36:05 CET] <Zexaron> is there a way to detect what kind of build options were used on precompiled .lib files [01:36:17 CET] <Zexaron> I think I did that in the past but forgot [01:36:27 CET] <Zexaron> I have like avcodec, avformat etc [01:36:39 CET] <Zexaron> I have the libs without exe [01:38:11 CET] <Zexaron> im not sure if precompiled lib is proper term, since it's 8 mb size, but when final project exe is built it's only a like less than 2 mb diff [01:39:47 CET] <BtbN> avcodec_configuration() [01:41:34 CET] <Zexaron> oh okay, I was browing through the raw file heh [01:42:10 CET] <nicolas17> search for "--enable" or "--prefix"? :P [01:42:19 CET] <BtbN> it's in there somewhere, strings will probably find it [01:46:57 CET] <Zexaron> yeah not quite getting it const char a = *avcodec_configuration(); debugger a is undefined ... i was in a rush [01:53:10 CET] <nicolas17> doesn't avcodec_configuration return a char array? [01:53:17 CET] <nicolas17> const char* a = avcodec_configuration(); [01:54:21 CET] <Zexaron> oh that [01:55:07 CET] <Zexaron> Yes with VS2017 error reporting I just go trial and error as sometimes the suggestions are so off [01:59:42 CET] <Zexaron> ain't working either [01:59:54 CET] <Zexaron> but I can watch avcodec_configuration() directly then [02:00:14 CET] <Zexaron> using the same code [02:01:57 CET] <Zexaron> funny how lib file leaks the username of the machine it was compiled in [02:03:03 CET] <Zexaron> anyway, it's just showing some disassembly and source stuff, not sure if this is it [02:08:39 CET] <Zexaron> oh okay it's not that big deal, I'll figure it out tomorrow [02:38:55 CET] <Matador> bloody hecks [03:08:38 CET] <nicolas17> any idea on how I could get the first frame of an mpegts while reading/downloading as little as possible? I think I would need to parse headers myself to know how big the first packet is and then download only that much... [03:10:19 CET] <nicolas17> and mpegts headers don't look like the easiest thing to parse ^^ [13:01:07 CET] <Zexaron> Hello [13:01:22 CET] <Zexaron> does ffmpeg, well, codecs, support changing resulution on-fly ? [13:01:40 CET] <Zexaron> Probably not but just checking if there's any oddball chance of any codec that can do that? [13:02:03 CET] <Zexaron> or let's say, input resolution change, but through a filter that scales it to the same output one [13:03:52 CET] <relaxed> it can decode such streams, not sure about encoding [13:30:40 CET] <Chagall> just wondering, why is y4m "c420" interpreted as c420jpeg (while c422/c444 have colocated chroma as far as I remember) while c420paldv is interpreted as having colocated chroma instead of the correct pal-dv format? [13:31:52 CET] <Chagall> it seems like the c420paldv implementation is missing and c420paldv is what c420 should be, though I'm not sure c420 was defined in yuv4mpeg [13:32:27 CET] <Chagall> that is, the current c420paldv implementation is what c420 should be as it tends to be defined [13:34:22 CET] <Chagall> I traced back the addition of 420 to this commit https://code.oxygene.sk/mirrors/ffmpeg/commit/b864b38397a2962d265ad39005265… and I am assuming that LOC_TOPLEFT and LOC_CENTER are exactly what they imply [13:37:30 CET] <Chagall> in master 420paldv is just treated as a clone of 420mpeg2, and 420 as a clone of 420jpeg (https://code.oxygene.sk/mirrors/ffmpeg/blob/master/libavformat/yuv4mpegdec.c) [13:39:08 CET] <Zexaron> relaxed: this gave me an idea, if there is a scaling mode that would scale the canvas size not the actual input so it would encode black bars but keep the original input size in place , since this is about frame dumping from a gaming emulator, it's live, it's not offline video, so it may only work if resolution is less than it originally started with [14:12:19 CET] <blenderwitch> hi! AFAIK, the following command should add image as an attachment stream to mkv container but instead adds it as a video stream: `ffmpeg -i test.mkv -c copy -attach cover.png -metadata:s:t mimetype=image/png test2.mkv` [14:17:49 CET] <blenderwitch> here's the output of `ffmpeg -i test2.mkv` after the attachment: http://pasted.co/b3e12750 [15:05:08 CET] <blenderwitch> ah, nvm. apparently ffmpeg just shows it weird. mkvinfo does it proper [15:05:10 CET] <blenderwitch> cheers! [15:11:36 CET] <lemourin> does avformat have stream preloading built in? that would be useful for http e.g or do i need to implement it myself? [16:18:20 CET] <pomaranc> identify separate [18:54:07 CET] <CoreX> best way to go about getting to know if the first and last frames are B frames? [18:54:53 CET] <furq> ffprobe -show_frames [18:57:32 CET] <CoreX> thanks [00:00:00 CET] --- Thu Nov 1 2018

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