November 2018 - Ffmpeg-devel-irc

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

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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

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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

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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

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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

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