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January 2018

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ffmpeg-devel.log.20180105
by burek 06 Jan '18

06 Jan '18

[01:57:34 CET] <cone-709> ffmpeg 03James Almer 07master:d36335bda5e5: avcodec/parser: use a mutex instead of atomics in av_register_codec_parser() [03:32:01 CET] <SortaCore> huh [03:32:13 CET] <SortaCore> I accidentally suspended a bunch of chrome processes and I stayed connected [03:36:27 CET] <cone-709> ffmpeg 03Marc-Antoine Arnaud 07master:206b25f9f45a: avfilter: reorder variable definition in geq [03:36:28 CET] <cone-709> ffmpeg 03Marc-Antoine Arnaud 07master:ac6b0bba79fe: avfilter: slice processing for geq [03:36:29 CET] <cone-709> ffmpeg 03Marc-Antoine Arnaud 07master:e425047a47b6: avfilter: rename variables in geq [03:36:30 CET] <cone-709> ffmpeg 03Michael Niedermayer 07master:b2be76c0a472: avcodec/dnxhddec: Check dc vlc [11:49:27 CET] <JEEB> mistym: yea, but you still need to parse the packets to get the header positions :D [11:49:54 CET] <JEEB> so in theory the decoder post-parser would be the thing to do it [11:50:24 CET] <JEEB> but as I noted I got no replies regarding how to do multiple decoder modes [11:50:26 CET] <JEEB> .17 [12:18:53 CET] <wm4> I didn't need to know that avi supports annexb h264 too [13:39:30 CET] <JEEB> wm4: people stick everything into avi [13:39:37 CET] <JEEB> ubfortunately [13:39:57 CET] <wm4> yeah [13:40:10 CET] <wm4> how does h264 with bframes work at all? [13:45:35 CET] <nevcairiel> it doesnt, you get wront timestamps [13:45:39 CET] <nevcairiel> wrong* [13:47:54 CET] <wm4> aw [13:48:48 CET] <nevcairiel> you basically only get dts timestamps [13:50:24 CET] <wm4> not really, DTS have this weird delay, so it doesn't quite work [16:01:08 CET] <BtbN> kierank, uhm, why did you send a PullRequest on Github? It can't be merged. [16:01:30 CET] <kierank> wanted to see what happens if I press the button saying "Create a code of conduct" [16:01:36 CET] <kierank> and that's apparently what it does [16:01:52 CET] <BtbN> So that's a default CoC Github gives you? [16:02:05 CET] <kierank> it gives you options between large project and small project [16:04:46 CET] <wm4> lol [16:05:22 CET] <wm4> so it's a template [16:05:38 CET] <wm4> and github has a button to make it easy to copy it into every project? [16:05:40 CET] <wm4> that seems dumb [16:26:02 CET] <kierank> wm4: it has a button to apply a code of conduct to a project yes [16:26:25 CET] <durandal_1707> nack , missing contact info [16:26:30 CET] <kierank> a million times better than the CoC we have now that's so bad i wouldn't use it for toilet paper [16:27:06 CET] <wm4> the problem is less with the CoC and more with enforcing it [16:27:20 CET] <durandal_1707> free speech against mplayer devs [16:27:40 CET] <kierank> wm4: well a good CoC has this in it [16:28:30 CET] <atomnuker> its all toilet paper [16:29:05 CET] <atomnuker> empty words that are useless at solving anything, used by weak people losing an argument [16:29:13 CET] <durandal_1707> paper is useful [16:29:38 CET] <wm4> yeah atomnuker would rather start an edit war [16:31:03 CET] <atomnuker> as long as its left as-is without any muting or banning it'll solve itself out by the people waging it [16:31:36 CET] <atomnuker> self-regulation may not work for markets but I'm confident it'll work elsewhere [16:32:16 CET] <atomnuker> particularly in systems that were created by self-regulation, like open source [16:33:52 CET] <wm4> you mean the hotter you flame the more likely you're to win [16:34:00 CET] <wm4> that seems indeed how it works currently [16:34:33 CET] <wm4> solving conflict by creating more conflict [16:34:34 CET] <wm4> genious [16:39:21 CET] <atomnuker> well its better than a coc which just says "mob the first who throws a bad word" [16:41:39 CET] <atomnuker> sure it solves the problem but there's no guarantee the solution it made was the best solution [16:42:13 CET] <kierank> atomnuker: you weren't there in like 2010 and before when it was conflict every day [16:42:21 CET] <kierank> gotta learn from the past [16:42:31 CET] Action: kierank regrettably mentions the past [16:59:41 CET] <jamrial> anyone want to look at the mutex patches so we may finally nuke the lavu atomic stuff? [17:04:39 CET] <wm4> I just "reviewed" the 3 v2 patches [17:07:26 CET] <atomnuker> durandal_1707: are the only issues with yuvj in ffplay and mjpegdec? [17:07:30 CET] <cone-560> ffmpeg 03Paul B Mahol 07master:7bb1be9af0ea: avfilter: add arbitrary audio IIR filter [17:08:37 CET] <wm4> I think it sort of broke down on libavfilter? [17:09:18 CET] <durandal_1707> atomnuker: anywhere where michaelni points out something [17:09:46 CET] <atomnuker> well that leaves 2 issues then [17:09:53 CET] <atomnuker> "warning: passing argument 1 of ff_make_color_range_list from incompatible pointer type [enabled by default]" in the lavfi part of the patchset [17:10:00 CET] <atomnuker> (which is easily fixable) [17:10:37 CET] <durandal_1707> ffplay issue can be fixed by rming ffplay [17:10:39 CET] <atomnuker> and the "[mjpeg @ 0x334dea0] colorspace not supported in jpeg" error in mjeg encoding [17:10:59 CET] <durandal_1707> iirc i fixed that one [17:11:12 CET] <durandal_1707> locally [17:11:36 CET] <atomnuker> oh, ok, that leaves just the warning [17:12:02 CET] <atomnuker> resubmit them and I'll lgtm [17:13:00 CET] <atomnuker> ubitux: did you manage to finish the subtitles in avframe stuff? [17:13:02 CET] <durandal_1707> need docs changes [17:13:43 CET] <ubitux> atomnuker: no& :( [17:13:51 CET] <ubitux> fuck me [17:13:51 CET] <atomnuker> got blocked somewhere? [17:14:04 CET] <ubitux> i have no decent excuse [17:14:12 CET] <ubitux> i'm sorry [17:14:30 CET] <jamrial> ubitux: christmas/new years is a good reason [17:14:51 CET] <ubitux> would be relevant if i was a social person [17:15:03 CET] <jamrial> not even family? :o [17:15:22 CET] <ubitux> nope, haven't seen them for a long time [17:15:31 CET] <jamrial> ah [17:18:52 CET] <cone-560> ffmpeg 03James Almer 07master:9ed4ebc530dd: avcodec/util: use a mutex instead of atomics in avcodec_register() [17:18:53 CET] <cone-560> ffmpeg 03James Almer 07master:57960b1f2800: avformat: use mutexes instead of atomics in av_register_{input,output}_format() [17:18:54 CET] <cone-560> ffmpeg 03James Almer 07master:167e659b289f: avfilter: use a mutex instead of atomics in avfilter_register() [17:19:39 CET] <jamrial> now, lets get rid of lavu atomics [17:25:04 CET] <atomnuker> ubitux: well if you think we're close we could probably postpone the bump some time for you to finish it [17:25:33 CET] <ubitux> no, bump for now [17:25:34 CET] <ubitux> it's fine [17:27:20 CET] <jamrial> you mean postpone closing the unstable abi season? [17:29:55 CET] <wm4> Chloe: time to learn how to resolve conflicts with git rebase [17:31:10 CET] <Chloe> shouldnt be too bad with `git pull -r` and 3 way merges [17:35:23 CET] <ubitux> jamrial: no, don't postpone [17:35:26 CET] <atomnuker> jamrial: yeah, I was hoping to have subs in avframe in but its fine, as long as yuvj and the avpriv atomic stuff is in [17:36:21 CET] <jamrial> does that need to be done while the abi is unstable? sizeof(avframe) is not part of it [17:36:49 CET] <atomnuker> doesn't it deprecate avsubtitles? [17:37:46 CET] <rcombs> mmmmm, subs in avframe [17:46:21 CET] <atomnuker> ubitux: if you want can you push some wip somewhere if someone wants to finish it? [18:07:15 CET] <cone-560> ffmpeg 03James Almer 07master:dcbf034a0fd2: avcodec/error_resilience: remove unused header [18:07:16 CET] <cone-560> ffmpeg 03Anton Khirnov 07master:89b84cb18b54: It has been replaced by C11 stdatomic.h and is now unused. [18:07:33 CET] <jamrial> way to fuck up the commit subject... [18:07:34 CET] <BBB> is there a title missing in that commit? [18:08:12 CET] <jamrial> yes [18:08:17 CET] <jamrial> the actual commit subject [18:08:40 CET] <jamrial> i fucked up and removed it, leaving only the second line [18:08:49 CET] <BBB> it happens, its ok [18:09:09 CET] <jamrial> at least the reference to the original commit is there in the cherry-pick line [18:21:44 CET] <durandal_1707> i want float128 type [18:22:32 CET] <atomnuker> is 80 bit precision not enough? [18:22:48 CET] <rcombs> float256 or go home [18:22:56 CET] <SortaCore> float4096 when [18:23:02 CET] <SortaCore> https://pastebin.com/raw/Ytcivhm0 r8 my patch bois [18:23:05 CET] <kierank> durandal_1707: long double [18:23:21 CET] <rcombs> loooooong double [18:23:30 CET] <SortaCore> long long double [18:24:09 CET] <rcombs> #define loooooong long long [18:24:28 CET] Action: SortaCore thumbs up [18:25:02 CET] <BBB> kierank: long double is 80 bits no? [18:25:17 CET] <wm4> which compiler dev thought "this is a good idea" [18:25:41 CET] <BBB> wm4: the one thats running for president in the next election cycle [18:25:58 CET] <atomnuker> only on gcc with default settings [18:26:05 CET] <atomnuker> long double is 64 bits on msvc [18:26:22 CET] <atomnuker> I think you needed some special flag to enable it there [18:28:41 CET] <rcombs> x87's not even present on e.g. quarks [18:29:45 CET] <wm4> BBB: did I miss anything? [18:30:11 CET] <SortaCore> no r8ing? [18:30:13 CET] <BBB> ignore it, it was a poor attempt at a joke [18:31:02 CET] <BBB> SortaCore: for broader review, Id send patches to the list; with just a link, we dont know what its about or whether we should look at it [18:31:24 CET] <BBB> SortaCore: it could be a patch to any part of the codebase, likely one I know nothing about, so I dont click [18:31:24 CET] <atomnuker> yeah, most of the people who care about qsv are on the list [18:32:23 CET] <jkqxz> There are people who care about qsv? [18:32:56 CET] <BBB> someone just wrote a patch for it :-p [18:33:00 CET] <jamrial> people with intel notebooks maybe [18:33:10 CET] <jkqxz> Wrt that patch, Am I missing something about why you don't just provide a device with only a hardware implementation? [18:34:09 CET] <jkqxz> The internal initialisation is only there so that some simple cases are easier. [18:36:34 CET] <SortaCore> atm there's one setting: select any implementation, software or hardware [18:36:57 CET] <SortaCore> qsv software is probably worse than not using qsv at all [18:37:13 CET] <SortaCore> so I added an option for requiring a hardware impl [18:37:59 CET] <jkqxz> Why would you ever use the internally-made session, though? The implementation is attached to the device context, you pass the type you want when you create that. [18:40:20 CET] <SortaCore> *whoosh* [18:42:25 CET] <jkqxz> ... -c:v h264_qsv -hwaccel qsv -hwaccel_device hw_any -i blahblah ... [18:42:47 CET] <SortaCore> how do you do it in C++? [18:43:10 CET] <jkqxz> Set hw_frames_ctx or hw_device_ctx. [18:43:23 CET] <jkqxz> s/hw/AVCodecContext.&/g [18:44:09 CET] <jkqxz> (The former for hardware frame output, the latter for software. An Intel guy recently wanted to add the latter for hardware too, dunno if that will actually happen.) [18:44:16 CET] <jkqxz> Are you using AV_PIX_FMT_QSV or not? [18:44:34 CET] <SortaCore> I'm not currently, I was getting errors from sws_scale saying it wasn't compatible [18:45:19 CET] <SortaCore> atm I get an RTSP stream in, YUV420P output, I convert to NV12 (and other formats), pass to QSV, read back into a MP4 [18:45:30 CET] <SortaCore> other formats being RGB for display and so forth [18:46:34 CET] <jkqxz> Make a device reference with av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV, "hw_any", NULL, 0), then put it as AVCodecContext.hw_device_ctx on all of you encoders and decoders. [18:47:11 CET] <jkqxz> I assume you are on Windows there, because on Linux the multiple devices without explicit handling is really broken. [18:47:23 CET] <SortaCore> yea, Windoze [18:47:50 CET] <SortaCore> atm I was creating a device, but for dxva2 [18:48:31 CET] <jkqxz> If you have a dxva2 device you can derive the matching qsv device from it using av_hwdevice_ctx_create_derived(). [18:49:22 CET] <SortaCore> atm I have neither for source, because it's RTSP [18:49:42 CET] <SortaCore> so should I go dxva2 + derived qsv or just qsv [18:50:47 CET] <durandal_1707> atomnuker: with very long filtergraph you can do multiband compression with native filtering [18:51:10 CET] <durandal_1707> ignoring mcompand [18:51:28 CET] <jdarnley> > [Fri 05 18:24] <rcombs> #define loooooong long long [18:51:34 CET] <jdarnley> I need to use that comewhere [18:51:38 CET] <jdarnley> *somewhere [18:51:43 CET] <durandal_1707> because it have issues.. [19:00:16 CET] <jkqxz> SortaCore: Use dxva2 for decode and derive the matching qsv device for encode. [19:14:56 CET] <rcombs> long long double [19:17:53 CET] <wm4> and for int128_t, long long long [19:17:57 CET] <wm4> ...int [19:18:07 CET] <wm4> unsigned long long long int [19:18:36 CET] <wm4> oh I thought "long int" is legal but it's not [19:18:57 CET] <wm4> I'm very confused [19:19:20 CET] <jamrial> now that we have to trash every CPU made in the past two decades, some of these might actually become useful [19:29:30 CET] <SortaCore> so probably not the best time to sell my old i5 [19:30:15 CET] <SortaCore> @jkqxz: alright, so create a dxva2, pass it with av_buffer_ref as hw_device_ctx to decoder, then create derived qsv and pass that to the qsv encoder? [19:35:58 CET] <jkqxz> Yes. [19:58:47 CET] <cone-560> ffmpeg 03Paul B Mahol 07master:52c959a23766: avfilter/af_aiir: do not crash with invalid options [20:13:08 CET] <LongChair> hmmm could anyone confirme something for me regarding ML ? [20:13:34 CET] <LongChair> i just submitted some fixes for rockchip rkmpp, and attached two patches [20:13:59 CET] <LongChair> the ML mail seems to contain both, but i can only see one in here https://patchwork.ffmpeg.org/patch/7146/ [20:14:13 CET] <LongChair> did i screw up something ? [20:15:06 CET] <jamrial> maybe it's stuck in moderation queue [20:15:13 CET] <jamrial> Compn: ^ [20:15:44 CET] <wm4> seems to be more about patchwork reading only the first attached patch [20:16:59 CET] <LongChair> Should I resubmit it in two mails ? or is the ML one good enough ? :) [20:20:04 CET] <wm4> good enough for me [20:20:26 CET] <wm4> though separate mails or even using git send-email would make reviewing easier [20:28:20 CET] <durandal_1707> i figured how to do linear phase crossover with iir filters and 4th order linkwitz-riley [20:47:40 CET] <cone-560> ffmpeg 03James Almer 07master:b9ad04b19c0c: fate: add PERSIST_RPARAM_A_RExt_Sony_3 hevc conformance test [21:10:18 CET] <SortaCore> do I have to use av_buffer_ref for both jkqxz? [21:48:57 CET] <Compn> wat\ [21:53:01 CET] <Compn> jamrial : nothing stuck in mod queue atm [21:53:15 CET] <Compn> that was 2 hours ago, possibly lou or someone else cleared it [21:53:19 CET] <Compn> or it never made it to us:\ [21:53:39 CET] <jamrial> Compn: yeah, my bad. i misunderstood what he said and assumed he sent two mails, where only one made it through [21:53:47 CET] <jamrial> when he had sent one with two attachments [21:54:28 CET] <Compn> no prob [21:54:51 CET] <Compn> just letting you know the results of me checking. i can check anytime, and feel free to ask me to check it :) [22:11:36 CET] <jamrial> wm4: regarding the encryption stuff, i don't really like having those two functions using avpacket in lavu [22:12:36 CET] <jamrial> both are something that should be done by the muxer/demuxer itself, like with every other side data type [22:35:16 CET] <wm4> jamrial: the data is relatively complex, so the idea was to make the exact representation internal, and to have the user access it through the new functions [22:35:24 CET] <wm4> I guess they could be in libavcodec as well [22:35:57 CET] <wm4> what we really want is probably not having side data be a flat byte array [22:36:22 CET] <wm4> but currently the general side data API pretty much guarantee that you can [22:38:26 CET] <jamrial> what makes it complex is that subsamples field, yeah [22:41:19 CET] <jamrial> how about making the functions one take a byte array passed by the caller (who got it with av_packet_get_side_data) to fill the AVEncryptionInfo struct, and the other allocate a byte array that the caller then needs to add as side data with av_packet_add_side_data? [22:50:36 CET] <wm4> jamrial: I guess that wouldn't be too bad if you see an advantage in this [22:52:11 CET] <jamrial> not having lavu use lavc structs is worth it [22:56:24 CET] <wm4> oh right, if it's going to be in lavi it actually can't or shouldn't use AVPacket at all [00:00:00 CET] --- Sat Jan 6 2018

1 0

ffmpeg.log.20180105
by burek 06 Jan '18

06 Jan '18

[05:00:11 CET] <illegal> can ffprobe show the size of the streams for .mkv's? [05:04:04 CET] <DHE> if the information isn't part of the metadata (and it doesn't always have to be) then the only way to know is to scan the whole file. which isn't something ffprobe does by default. and I don't think there's such a summary option available without calculating it yourself [05:05:42 CET] <illegal> oh well, thanks homie [05:58:22 CET] <TAFB> I'm trying to stream my 4k ip camera with ffmpeg but it's super unreliable :( After somewhere between 15 seconds and 1 hour it'll just "stop" and begin using 100% of one of my CPU cores (uses less than 2% normally). If I open another copy of ffmpeg the ip camera stream is working fine, and it can stream to youtube fine. Need serious help :( https://pastebin.com/SrA9Lasc [06:43:48 CET] <Fyr> guys, players sometimes freeze for a few seconds on certains moments when playing the video file I muxed with FFMPEG. it looks like those places a new segments. [06:44:16 CET] <Fyr> is it possible to avoid those lags using some options, like +genpts? [07:01:10 CET] <kazuma_> are you cutting/merging on i-frames? [07:04:51 CET] <Fyr> I'm copying video and audio without re-encoding. [07:05:25 CET] <Fyr> freezing appears only when using certain audio channel. [07:05:44 CET] <Fyr> when playing with other channels, it doesn't take place. [07:06:04 CET] <Fyr> when muxing I get "Starting new cluster due to timestamp" [07:06:19 CET] <Fyr> too often, than usual. [08:02:08 CET] <ItWasntMe2013> does anyone have an idea how I could detect audio in a frame, similar to how video filter blackdetect works, but I want to detect say high volume or even a certain frequency? [09:26:45 CET] <vincent-vlk> Hello dear Community, first of all, Happy new Year and thank you for this wonderful software package *ffmpeg*. Have a quick qustion: Is there any possibility do measure CC Errors (Continuity err.) of IP-Multicast/UDP/MPEG streams? Thank you very much! [09:27:47 CET] <vincent-vlk> Forgot to write, I'm expecting to use *ffprobe*, or am i wrong? Thanks in advance. [09:55:01 CET] <phw> I am trying to write chapters into mp3 files; That needs a TLEN frame in the file; ffmpeg does not seem to produce those. What can I do about it? [12:08:24 CET] <bencc> I'm trying to convert mp3 to hls playlist with this command: [12:08:27 CET] <bencc> ffmpeg -i in.mp3 -c:a aac -b:a 128k -hls_time 2 -hls_playlist_type vode -hls_flags single_file playlist.m3u8 [12:09:13 CET] <bencc> the duration is wrong in the playlist and I think also in the .ts file [12:09:14 CET] <bencc> https://pastebin.com/PW39Fbs7 [12:09:28 CET] <bencc> do I need to add fake video stream? [12:43:29 CET] <arcamargo> hello people. someone have some clue about why this 'non-monotonous dts' happen? https://pastebin.com/haW7ERLH [13:33:09 CET] <thm> I have a trp file (basically a ts, iiuc), and my ffmpeg wrongly detects the videotext stream as mp3 with 0 channels. is there a way to manually overwrite that from the commandline? [14:01:56 CET] <TAFB> I'm trying to stream my 4k ip camera with ffmpeg but it's super unreliable :( After somewhere between 15 seconds and 1 hour it'll just "stop" and begin using 100% of one of my CPU cores (uses less than 2% normally). If I open another copy of ffmpeg the ip camera stream is working fine, and it can stream to youtube fine. Need serious help :( https://pastebin.com/SrA9Lasc [15:20:35 CET] <esc-reality> Can someone maybe help me with this question? :) https://superuser.com/questions/1282809/is-ffmpeg-able-to-read-multiple-val… [15:45:54 CET] <InTheWings> wth is pkt.dts/pts rewinding in middle stream of a solid mp4 with ffmpeg. that's nonsense [15:47:05 CET] <InTheWings> dts=17117709 pts=17284709 [15:47:13 CET] <InTheWings> dts=4104701 pts=4271701 [15:47:18 CET] <InTheWings> H264 track... [15:47:43 CET] <pos> I'm doing -vcodec copy of an rtsp stream (which I have no control over) and ffmpeg is printing: [15:47:46 CET] <pos> [mp4 @ 0x22bcea0] pts has no value [15:47:49 CET] <pos> [mp4 @ 0x22bcea0] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file. [15:47:52 CET] <pos> [mp4 @ 0x22bcea0] Non-monotonous DTS in output stream 0:0; previous: 1, current: 0; changing to 2. This may result in incorrect timestamps in the output file. [15:48:21 CET] <InTheWings> pos: because something is borked since rtsp only sends PTS [15:48:23 CET] <pos> the resultant file is playable in mpv/mplayer but ffplay just displays the first frame and thinks the video is 40 minutes long [15:48:54 CET] <pos> is there anything i can pass to ffmpeg to make it fix this on the fly? i've tried -fflags +genpts [15:49:01 CET] <pos> also, -vsync drop [15:49:30 CET] <titou> hey [15:49:43 CET] <pos> InTheWings, please explain? [15:49:49 CET] <kepstin> pos: does it work if you re-encode the video rather than copy? What about with a different container (mkv, nut?) [15:49:57 CET] <pos> yes, re-encode works [15:50:22 CET] <pos> but i'd rather avoid doing that, several multi-megapixel streams plus x264 is not lossless [15:51:10 CET] <titou> I'm trying to stream video from three different V4L2 cameras.. The only solution I found is to launch three instances of ffserver and three instances of ffmpeg.. It works when I have 2 instances of ffmpeg+ffserver but when I add the third one, I lost the two first streams.. [15:53:29 CET] <pos> vlc2 borks when trying to read the resultant file, vlc3rc behaves much the same way as mpv/mplayer [15:53:49 CET] <pos> no way/flag/option/whatever to make ffmpeg fix this on the fly? [15:57:45 CET] <pos> pulling the same rtsp stream through vlc2 also works/displays ok, but it is clear that there are timestamp issues [16:01:47 CET] <pos> i was disconnected right after <titou> I'm trying to stream vide.... [16:01:51 CET] <pos> anyone come up with anything? [16:05:46 CET] <pos> i was reading through a lot of doc trying to solve this on my own, isn't it possible for ffmpeg to just generate its own timestamps based on when the stream was initiated instead of using the borked ones from the source? [16:08:40 CET] <SortaCore> genpts flag I think [16:09:25 CET] <pos> i tried -fflags +gentps [16:10:00 CET] <SortaCore> tps? [16:10:07 CET] <pos> pts ofc [16:11:29 CET] <pos> and igndts, ignidx [16:15:11 CET] <pos> ffplaying the source directly also works [16:15:19 CET] <pos> no complaints [16:19:28 CET] <pos> pts ofc [16:19:30 CET] <pos> ffplaying the source directly also works [16:19:32 CET] <pos> no complaints [16:34:36 CET] <pos> playing the source directly in mpv: Invalid video timestamp: -0.000000 -> -0.000000 [16:49:23 CET] <thm> in mpegts, the streams have hex identifiers, like 0x136 in "Stream #0:1[0x136]: Video: mpeg2video (Main)" - can I use those somehow as arguments to the -map option? [16:51:02 CET] <furq> thm: -map i:0x136 [16:51:08 CET] <furq> bear in mind that the pids might change though [16:51:57 CET] <thm> problem is, that those trp files lack the pat (?), and streams appear in random order, and also streams are misidentified [16:56:10 CET] <thm> (subtitles as mp3) [17:31:41 CET] <ze0n> hi, someone know why i get this error ? https://pastebin.com/01fT7p6X if i already have installed the version 388.71 of the controller ? [17:32:40 CET] <BtbN> have you rebooted? [17:33:17 CET] <BtbN> The error is just printed as a generic hint, it does not mean you neccesarily have a too old driver. It just means it failed to load the named dll [17:35:44 CET] <ze0n> mmm i am looking the that dll on my sistem but can't found [17:37:35 CET] <BtbN> Is this some super old GPU or an Optimus Laptop? [17:39:33 CET] <ze0n> gtx 550 ti [17:40:09 CET] <BtbN> that plain does not have nvenc. So the driver won't install the dll [17:40:20 CET] <ze0n> mm i found it on /system32 [17:41:11 CET] <BtbN> First gen to have nvenc is Kepler. 500 series is Fermi. No nvenc [17:42:41 CET] <ze0n> oh The gtx 550 ti won't support NVENC(it only on 600 series that use kepler\maxwell or newer [17:42:54 CET] <ze0n> fuck my life then [17:43:19 CET] <ze0n> thanks anyway BtbN [18:03:05 CET] <pos> i'm getting [mp4 @ 0xa60880] pts has no valu [18:03:12 CET] <pos> [mp4 @ 0xa60880] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file. [18:03:21 CET] <pos> [mp4 @ 0xa60880] Non-monotonous DTS in output stream 0:0; previous: 1, current: 0; changing to 2. This may result in incorrect timestamps in the output file. [18:03:53 CET] <pos> when doing ffmpeg -vcodec copy manually of an rtsp stream, the resultant file plays ok, however. thing is, the application usually doing this does not seem to adjust this on the fly. is there any way to specify it, setting it manually to 2? [19:56:47 CET] <Mrbriareos> hi [19:57:39 CET] <Mrbriareos> I want to download a video from a site I have to login to (I have an account there so I have my own username and password) is there a command I can use with ffmpeg to allow me to download video from this site? [20:21:10 CET] <kepstin> Mrbriareos: not really in scope for ffmpeg directly. Look into the 'youtube-dl' project, it has plugins to download from a bunch of sites [21:16:06 CET] <SortaCore> when I pass metadata to AVFormatContext, do I do it before avio_open2, before av_write_header, after streams are added, etc? [21:17:15 CET] <SortaCore> A lot of the documentation states if a person should set the field during encoding/decoding, but if there's a function before or after it should be [21:35:18 CET] <SortaCore> right now all my metadata seems to be getting eaten [22:29:44 CET] <lomancer1> hello I am trying to initialize a hw_accel_context and am unable to. I detailed my question here https://stackoverflow.com/questions/48070827/initializing-hwaccel-context-i… Does anyone know how to correctly initialize a hw_accel_context? Thanks for the help! [23:23:37 CET] <aokfire> Hi, I'm trying to crop and resize a video from 1080 to 720, and remove some bars on the top/bottom of the video. Not sure what to do properly to get this result. thanks [23:24:11 CET] <aokfire> -vf "scale=1280:720, crop=1280:718" and other similar attempts have not worked. Either outputs 720, or 1080 [23:30:01 CET] <relaxed> it would be more efficient to crop first [23:31:41 CET] <relaxed> did you read https://ffmpeg.org/ffmpeg-filters.html#crop ? [23:32:23 CET] <aokfire> so crop from 1080 source first, then resize to '720'? [23:32:30 CET] <relaxed> yes [23:33:01 CET] <aokfire> crop=1920:1078, scale=1280:718 ? [23:33:18 CET] <aokfire> as an example, to remove 2px [23:34:07 CET] <aokfire> and this would remove it from the top down yes? [23:34:16 CET] <aokfire> since i only need it gone on the top [23:36:32 CET] <relaxed> correct, add "-t 5" to get a small sample [23:37:46 CET] <furq> aokfire: scale to 1280:-2 and it'll work out the correct height for you [23:54:39 CET] <relaxed> aokfire: sorry, for the top two lines you need crop=1920:1078:0:0 . It defaults to /2 [23:54:51 CET] <aokfire> sorry, Was away for a second. I will try this in a second [23:55:03 CET] <aokfire> what does -t 5 do, trim 5s? [23:55:15 CET] <relaxed> stop after 5 seconds [23:55:20 CET] <aokfire> ah [00:00:00 CET] --- Sat Jan 6 2018

1 0

ffmpeg-devel.log.20180104
by burek 05 Jan '18

05 Jan '18

[00:01:28 CET] <atomnuker> klaxa: how does your mkvserver_mk2 work? I give it matroska via stdin and there's nothing when I try tcp or http protocols on the server [00:05:03 CET] <BtbN> God damn those new CPU vulnerabilities are quite something [00:05:35 CET] <wm4> hardware bugs are fun [00:05:40 CET] <BtbN> You can in theory execute almost arbitrary code in Kernel-Space from JavaScript in some website. [00:06:02 CET] <wm4> I thought it was only about reading kernel data [00:06:09 CET] <BtbN> It's _way_ more [00:06:28 CET] <BtbN> Just because the CPU speculatively executes your illegal function call. You first use a timing attack on cache loads to read arbitrary data, bit by bit [00:06:52 CET] <BtbN> And then you find a specific function of the Linux Kernel, from the eBPF framework, which executes a pre-compiled and trusted bytecode [00:07:09 CET] <BtbN> And then call it with arbitrary malicious bytecode [00:07:19 CET] <wm4> wouldn't that be just a bug [00:07:20 CET] <BtbN> You will get an error because you're not allowed to call that [00:07:33 CET] <BtbN> But the CPU speculatively still executed your code, and wrote its results to cache [00:07:40 CET] <BtbN> Which you can then extract with the same method, bit by bit [00:08:09 CET] <nevcairiel> you cant really execute more then just one or two asm instructions [00:08:14 CET] <nevcairiel> until the fault is processed [00:08:39 CET] <BtbN> Well, they successfully made a Haswell CPU execute their byte code to the eBPF function [00:08:42 CET] <BtbN> so it seems like you can [00:10:16 CET] <BtbN> "This section describes the theory behind our PoC for variant 2 that, when running with root privileges inside a KVM guest created using virt-manager on the Intel Haswell Xeon CPU, with a specific version of Debian's distro kernel running on the host, can read host kernel memory at a rate of around 1500 bytes/second." [00:10:43 CET] <wm4> that's still just read access [00:10:44 CET] <BtbN> And later they also describe a theoretical attack where it works on any custom Kernel [00:11:00 CET] <BtbN> Yes, of course it's only read access [00:11:12 CET] <BtbN> Any writes you do will get eventually discarded [00:11:31 CET] <BtbN> But reading arbitrary data from a VM host is quite a big deal [00:12:18 CET] <wm4> because it sounded like you claimed something else, but maybe I misunderstood [00:12:37 CET] <BtbN> You read by making the CPU speculatively execute arbitrary BPF bytecode [00:12:56 CET] <BtbN> And then pulling the results of that bytecode from the cache [00:13:36 CET] <BtbN> https://googleprojectzero.blogspot.de/2018/01/reading-privileged-memory-wit… has the full lenghty description [00:14:54 CET] <wm4> yep [01:07:58 CET] <klaxa> atomnuker: it should listen on http://0:8080 [01:20:20 CET] Action: kierank has stopped receiving ffmpeg-devel emails [01:41:50 CET] <Chloe> kierank: unintentionally? [01:42:05 CET] <kierank> yes [01:42:54 CET] <kierank> I want to reply to michaelni's h264 patch [01:45:39 CET] <atomnuker> klaxa: oh yeah it works, didn't notice it crashed when I tried to connect [01:46:04 CET] <klaxa> it's not the most stable piece of software. i'm not even sure why it crashes sometimes [01:46:25 CET] <atomnuker> I'm running it on the rpi so its probably that [01:48:52 CET] <klaxa> in the future" i might check what causes hangs, something i found was weird requests with empty http methods, not sure what to do about that yet, but i haven't looked at that code in a long time [01:50:03 CET] <atomnuker> also it fails to build if ffmpeg was build with static linking in mind [01:54:20 CET] <klaxa> huh... [01:54:35 CET] <klaxa> something something, pthreads? [01:54:59 CET] <atomnuker> no, all av_* functions [01:55:15 CET] <atomnuker> segment.o: In function `av_make_error_string': [01:55:18 CET] <atomnuker> /usr/local/include/libavutil/error.h:111: undefined reference to `av_strerror' [01:57:11 CET] <c3-Win> So the AMD PRO CPU is one of their FX/Bulldozer-based CPUs. No surprise there. [02:00:24 CET] <iive> meltdown and spectre ... [02:25:09 CET] <atomnuker> klaxa: crashes after a second in ffio_ensure_seekback [02:25:26 CET] <atomnuker> do I really really need to apply the first patch in patches? [02:34:25 CET] <c_14> the first one should be in master already [02:35:01 CET] <c_14> I think I found the compilation issue, but it seems like there's something deeper [02:35:07 CET] <klaxa> the second one is a memory leak, that was never fixed :( maybe i should resubmit, it's been like 9 months? [02:36:44 CET] <atomnuker> yeah, or ping the patches if they still apply, I'll take a look if I have to [03:54:41 CET] <cone-503> ffmpeg 03Rostislav Pehlivanov 07master:c29038f3041a: lavr: deprecate the entire library [03:54:41 CET] <cone-503> ffmpeg 03Rostislav Pehlivanov 07master:f141b353e60f: opusenc_psy: disable stereo searches for mono streams [05:40:18 CET] <cone-503> ffmpeg 03Carl Eugen Hoyos 07master:695b1d81117d: lavu/mem: Allow allocations close to max_alloc_size with av_fast_realloc(). [07:24:19 CET] <JEEB> mistym: you're the one working on the ATRAC3+ MPEG-PS encapsulation? [07:24:29 CET] <mistym> JEEB: Yep! [07:24:53 CET] <JEEB> I posted a comment thing about it, and I'll then also link you a PS3 sample which uses PAMF00 instead of PSMF00 [07:25:13 CET] <JEEB> found it out during poking into this stuff [07:25:15 CET] <mistym> Thank you! I'd love to take a look at that sample [07:27:12 CET] <JEEB> mistym: http://kuroko.fushizen.eu/samples/UT2OP.movie [07:27:57 CET] <mistym> If you've been looking at it, I'm also very curious if you've found a way to identify ATRAC-3 within private section 1 headers w/o relying on the PSMF00 header wrapping the MPEG file [07:28:01 CET] <mistym> Thank you! [07:33:02 CET] <JEEB> any work I did was heavily grounded on the header so unfortunately not [07:33:34 CET] <JEEB> and yea, as I noted in the e-mail before people NG'd the parser because it touched the AVCodecContext, so I'll see if people reply regarding it [07:33:49 CET] <JEEB> (it was not specifically to you, but in general for the folks reading the thread) [07:37:15 CET] <JEEB> I was going to check if Sony used something similar for PSVita with ATRAC9 [07:37:36 CET] <JEEB> but the sample videos I've got myself are all MP4s :P [07:43:27 CET] <mistym> Yeah, my hope was to avoid peeking at the header and storing stuff in AVCodecContext, and instead sniff the contents of the private section to figure out if there's going to be ATRAC3P [07:44:15 CET] <JEEB> that's the AVFormatContext :) [07:44:23 CET] <JEEB> which is more OK since you're setting that in the demuxer [07:45:02 CET] <JEEB> the real problem is that currently the decoder's design is one for the WAV-like OMA, which has the packets without a header and the values in the header of the file [07:45:34 CET] <JEEB> while with MPEG-PS encapsulation you've got the header in each packet IIRC [07:46:26 CET] <mistym> Er yeah, AVFormatContext. The real reason I don't want to sniff the header is because I want to include support for .MPS movies from UMD video discs, which don't have the PSMF header [07:46:35 CET] <JEEB> ah yes [07:46:39 CET] <JEEB> I've got a couple of those too [07:48:24 CET] <JEEB> UMD video was funny in the way that it was SD, but could easily have like 6+mbps with H.264 [07:48:53 CET] <JEEB> so some things that never got any HD thing and only a DVD and UMD, the UMD could quite possibly be much better :P [07:50:24 CET] <mistym> I wouldn't call the UMD exclusive video *good* per se but it's at least interesting enough to have support for :b [07:50:50 CET] <JEEB> well compare it with MPEG-2 Video at 8mbps with a 1.5megabit buffer size [07:51:10 CET] <mistym> Oh rather I mean the movies that are exclusive to the format [07:51:13 CET] <JEEB> and yes, it's not necessarily good but still possibly the least retarded version [07:52:07 CET] <JEEB> yea not sure what *exclusives* it had [07:52:36 CET] <mistym> I've been testing with a ridiculous, not-very-good AFL compilation that was like $0.50 [07:52:44 CET] <mistym> I don't even watch AFL [07:53:20 CET] <JEEB> :D [07:54:36 CET] <JEEB> I've got a copy of spider-man that sony used to give out to people who registered their PSPs [07:55:00 CET] <JEEB> and then I have Comic Party Revolution, which has some rather fabulous analog conversion artifacts (dot crawl etc) [07:55:20 CET] <JEEB> but I wouldn't be surprised if the latter is the best version released, lol [07:56:26 CET] <mistym> Oh wow [07:58:08 CET] <mistym> I've taken a break from the ATRAC3 branch for a day or two to come back with a clear head, have been looking at XA audio in AIFF and whatever the hell Grandia uses in the meanwhile [08:09:38 CET] <JEEB> mistym: hmm, VGMToolbox seemed to have some support for PSM [08:09:50 CET] <JEEB> unless it does something really simple :P [08:09:52 CET] <mistym> JEEB: Hmm, I'll take a look [08:10:04 CET] <mistym> I did look at PPSSPP, but it goes off of the PSMF header :/ [08:10:26 CET] <JEEB> yes, the MPEG-PS patch + parser thing is from there [08:21:30 CET] <mistym> Oh interesting, VGMToolbox supports subtitles https://sourceforge.net/p/vgmtoolbox/code/HEAD/tree/format/VGMToolbox/forma… [08:21:56 CET] <JEEB> I decided I am too lazy to look through it in SF so I started a git svn clone :P [08:23:17 CET] <mistym> lol [08:24:28 CET] <JEEB> https://github.com/jeeb/vgmtoolbox [08:26:40 CET] <JEEB> so it has GetAudioFileExtension and IsThisAnAudioBlock ... [08:27:30 CET] <mistym> But IsThisAnAudioBlock just checks for 0xBD, which is private stream 1 and can also mean implementation-defined stuff [08:27:35 CET] <JEEB> yea [08:27:45 CET] <mistym> GetAudioFileExtension seems to assume that any stream ID sub-20 is ATRAC [08:27:48 CET] <JEEB> so it's highly possible he just hard-codes file extensions :P [08:27:55 CET] <JEEB> as in, the input ones [08:28:07 CET] <JEEB> PMF/MPS => sony MPEG-2 [08:28:21 CET] <mistym> I mean, this is somewhat generalized, but I'm not sure what other stuff this would choke on [08:28:53 CET] <JEEB> funny how I see Atrac3Bytes [08:28:56 CET] <mistym> It's true that the mapping here lines up with other private stream 1 usage I've seen, e.g. DVD subs starting at 0x20 [08:28:57 CET] <JEEB> but that's not used anywhere? [08:31:10 CET] <mistym> Huh, weird [08:31:21 CET] <mistym> That might be a useful hint though [08:31:44 CET] <JEEB> last modified @ https://github.com/jeeb/vgmtoolbox/commit/b0362bb2819b8c1e955a74bc753a1e87e… [08:32:09 CET] <mistym> Useful to have a mirror :b [08:36:55 CET] <mistym> I don't see 0x1E, 0x60, 0x14 in the sample files I have :/ [08:39:02 CET] <mistym> Oh, but I *do* see 1E 60 04 in the PMF file, and 1E 60 04 in the MPS [08:39:15 CET] <mistym> I walked back thruogh blame a bit and found the 0x04 version here https://github.com/jeeb/vgmtoolbox/commit/6754b1c81d89bda8d2c109073bf9b4207… [08:41:37 CET] <JEEB> yeh [08:42:02 CET] <JEEB> of course it'd be nice to know what values those exactly are [08:44:35 CET] <mistym> It comes after the PES header I think: http://dvd.sourceforge.net/dvdinfo/pes-hdr.html [08:46:02 CET] <JEEB> ah [09:11:50 CET] <KGB> [13FFV1] 15JeromeMartinez reopened pull request #77: White space for forcing carriage return in a paragraph (06master...06Markdown) 02https://git.io/vQ0Qm [09:57:38 CET] <SortaCore> has YUVJXXX formats been cleansed yet [11:39:31 CET] <wm4> SortaCore: no, durandal_1707 gave up [11:40:13 CET] <mistym> JEEB: Updated version of this branch should work: https://github.com/mistydemeo/FFmpeg/tree/mps_support_atrac3 [11:41:49 CET] <mistym> I used the ATRAC bytes that vgmtoolbox notes to flag the presence of ATRAC and then assumed that any startcode under 0x20 is ATRAC in that case [11:43:45 CET] <mistym> Works with your Utawarerumono sample, plus the 3rd Birthday .PMF file and AFL .MPS file I was testing with [11:44:14 CET] <SortaCore> ah [11:44:22 CET] <SortaCore> then can we drop the deprecated pixel format warnings >.> [11:51:29 CET] <wm4> IMO we should motivate durandal_1707 to continue [11:54:02 CET] <SortaCore> do it durandal_1707 [11:54:08 CET] <SortaCore> don't let your memes be dreams [11:55:18 CET] <thardin> don't let your dreams be memes [11:59:07 CET] <JEEB> mistym: najs [12:01:45 CET] <SortaCore> just use that sweet regex, replace all YUVJ420P so next line has color range set to jpeg [12:01:52 CET] <SortaCore> I see no problems [12:02:39 CET] <SortaCore> top ffmpeg contributor award when [12:03:02 CET] <mistym> JEEB: Going to try resubmitting the patchset like this, unless you can think of any issues with that approach :b [12:05:50 CET] <JEEB> mistym: if it otherwise is the same thing then the question I noted earlier still stands, hopefully someone will answer :P [12:06:15 CET] <mistym> Yeah that one still applies :b So we'll see if anyone thinks it's an issue [12:06:27 CET] <JEEB> if not then I would kind of be laughing :P [12:06:46 CET] <JEEB> because that's what blocked the patch set back then [12:06:57 CET] <mistym> A lot can change in four years [12:07:02 CET] <JEEB> true [12:08:21 CET] <JEEB> anyways, as a separate discussion I wonder which should be the "canonical format" for ATRAC3+ in FFmpeg. the MPEG-PS one with the per-packet headers sounds better, and in either case you would have to make sure the canonical one gets converted to whatever the container requires [12:25:01 CET] <mistym> wrt modifying AVCodecContext from the parser, I suppose I could parse that data out the packet in the demuxer? [12:36:24 CET] <cone-666> ffmpeg 03Paul B Mahol 07master:89bbf5c7ec18: avfilter: add hilbert source FIR filter [12:47:49 CET] <KGB> [13FFV1] 15JeromeMartinez opened pull request #87: Change remaining_bits_in_bitstream to remaining_bits_in_block (06master...06blocks) 02https://git.io/vNvVU [12:48:29 CET] <KGB> [13FFV1] 15JeromeMartinez opened pull request #88: Make the topmost value of the border column explicit (06master...06topmost) 02https://git.io/vNvVI [12:49:19 CET] <KGB> [13FFV1] 15JeromeMartinez opened pull request #89: Replace color space wording by more adapted transformation wording (06master...06transformations) 02https://git.io/vNvVm [12:50:13 CET] <KGB> [13FFV1] 15JeromeMartinez opened pull request #90: Use Golomb Rice wording when applicable (06master...06VLC) 02https://git.io/vNvVZ [12:50:40 CET] <Guest87975> [13FFV1] 15JeromeMartinez opened pull request #91: Add get_bits() definition (06master...06get_bits) 02https://git.io/vNvVc [13:16:05 CET] <durandal_1707> which crossover to use for bandsplit filter? [13:20:17 CET] <durandal_1707> some audiophiles claim LT4 is worse than LT2 [14:05:28 CET] <wm4> BBB: technically you maintain the http code, could you review my reconnect patch? [14:37:20 CET] <BBB> technically :-p [14:37:28 CET] <BBB> dont I technically maintain some rtp code also? [14:39:51 CET] <BBB> is this enabled by default, or is it some AVOption you have to set on http? [14:40:21 CET] <BBB> my question is: could this lead to situations where the current behaviour is that it exits and the new behaviour will be that the user thinks it hangs? [14:40:36 CET] <BBB> (even if it didnt hang but the user just blindly updated) [14:40:43 CET] <BBB> (and no interrupt handler was set) [14:56:28 CET] <durandal_1707> atomnuker: you can already have all sorts of multiband compressors with ladspa and lv2 filters [14:59:09 CET] <wm4> BBB: it's an avoption, off by default [14:59:51 CET] <wm4> BBB: I wanted to use it, but the fact hat it returns an error if the first reconnect fails breaks streaming [15:00:03 CET] <wm4> so I'd say this is just sort of a bug fix [15:02:44 CET] <BBB> yes, ok, then its fine [15:03:00 CET] <BBB> if it was on-by-default, I would have suggested adding a log message while its reconnecting [15:03:18 CET] <BBB> as a warning or so, just so the user has some kind of faint idea whats going on [15:03:47 CET] <BBB> (is info on by default?) [15:04:27 CET] <wm4> there's a second patch which raises the log message level [15:04:35 CET] <BBB> oh nice [15:04:36 CET] <BBB> cool [15:04:41 CET] <BBB> ok no further comments then [15:04:51 CET] <BBB> all good [15:04:56 CET] <wm4> well thanks, then I'll push [15:07:47 CET] <wm4> once I can even access the git repo [15:16:35 CET] <cone-666> ffmpeg 03wm4 07master:8a108bdea06f: http: block while waiting for reconnecting [15:16:36 CET] <cone-666> ffmpeg 03wm4 07master:1b283c4a0dde: http: bump message level for reconnect message and log timeout [15:39:21 CET] <wm4> tls_opentls.c uses pthreads if HAVE_THREADS is defined, isn't this wrong? [15:39:54 CET] <wm4> doesn't HAVE_THREADS only imply the ff_ defines are available? [15:42:07 CET] <RiCON> tls_openssl*? [15:42:45 CET] <wm4> err yes [15:43:24 CET] <wm4> tls_gnutls does apparently the same thing [15:43:26 CET] <nevcairiel> all our emulation things map to pthreads functions [15:43:28 CET] <wm4> GCRY_THREAD_OPTION_PTHREAD_IMPL [15:43:32 CET] <nevcairiel> so what else should it be [15:44:13 CET] <wm4> do I want to know what that macro does [15:44:53 CET] <wm4> nevcairiel: right, I confused that [15:44:57 CET] <wbs> the GCRY one is for gcrypt, which gnutls has stopped using since quite a few years; if you drop gnutls < 3.x, you should be able to just ditch that part [15:45:06 CET] <wbs> and hopefully the same for openssl thread safety callbacks [15:45:22 CET] <wm4> yeah I see a GNUTLS_VERSION_NUMBER check here [15:45:40 CET] <RiCON> and same for openssl [15:45:51 CET] <RiCON> wait, no. [15:46:05 CET] <wm4> I have no idea how important those old versions are for users [15:46:11 CET] <wm4> or if anyone would not like dropping them [15:46:53 CET] <wbs> the gnutls ones probably are safe to drop, they were old (but still required) back when the libavformat tls code was written 6 years ago [15:47:05 CET] <wm4> looking around in the gnutls header files I see a gnutls_global_set_mutex() [15:47:53 CET] <wm4> oh nice, it documents that it defaults to OS API, and you should not use it [15:55:48 CET] <wm4> nice, newer gnutls make gnutls_global_init thread safe, apparently [15:55:53 CET] <wbs> yes [15:56:00 CET] <wm4> GNUTLS_STATIC_MUTEX(global_init_mutex); [15:56:21 CET] <wbs> I think this became really nice since 3.0 or so [15:56:43 CET] <wm4> their win32 code for this is bogus [15:57:02 CET] <wm4> or, lets say, questionable [15:57:36 CET] <wm4> http://sprunge.us/PfIe [15:58:04 CET] <wm4> good enough as far as I'm concerned [15:58:27 CET] <nevcairiel> its fine, but has the uglyness of leaking the mutex [15:58:54 CET] <wm4> yep, and it's not our problem... and maybe they'll eventually drop XP support and fix it [16:01:46 CET] <kierank> michaelni: ping [16:01:46 CET] <wm4> I find it harder to tell what openssl does [16:06:10 CET] <wm4> in any case newer openssl seems to be safe(r) [16:06:41 CET] <wm4> so yeah, I don't see any reasons to care about this stuff anymore [16:07:09 CET] <wm4> we could deprecate avformat_network_init [16:10:33 CET] <wm4> SSL_library_init is kind of unsafe even in openssl-1.1.0g [16:14:14 CET] <wm4> ok it's good enough if you ask me [16:14:31 CET] <wm4> there's some silly stuff about the openssl lib registering an atexit handler for uninit... [16:49:31 CET] <cone-666> ffmpeg 03Humberto Ribeiro 07master:59b126f92225: libavutil/hwcontext_dxva2: Add check for possible errors from GetAdapterDisplayModeEx [16:49:31 CET] <cone-666> ffmpeg 03wm4 07master:18fbfd7bf86e: hwcontext_dxva2: initialize D3DDISPLAYMODEEX correctly [17:03:14 CET] <wm4> so there were almost no code changes required to make those network init functions optional [17:28:00 CET] <michaelni> kierank, pong [17:35:26 CET] <atomnuker> jamrial: can I apply the ffserver removal patches now even though the bump period isn't over? [17:35:49 CET] <atomnuker> I can wait until Chloe's patches are ready [17:47:15 CET] <jamrial> atomnuker: yeah, once we can remove the lavu atomic stuff, we remove everything at once [17:47:26 CET] <Chloe> atomnuker: do that first I think. I've got tons of school work I need to do before the 8th [17:48:27 CET] <nevcairiel> just replace the atomic stuff with static mutexes now that we have those for windows, and dont rush the other things [17:58:02 CET] <wm4> I think those Chloe patches actually don't require ABI bumps, or do they? [17:58:22 CET] <Chloe> they shouldnt no [17:58:28 CET] <Chloe> they just deprecate certain things [17:58:29 CET] <Chloe> oh [17:58:50 CET] <Chloe> They 'clarify' some functionality in registering internal components [17:59:09 CET] <Chloe> you can no longer register them through the external API, so maybe [17:59:10 CET] <atomnuker> they indirectly do since then we can remove the avpriv atomic stuff [17:59:18 CET] <atomnuker> wm4: so is the AV_CODEC_CAP_SUBFRAMES deprecation patch okay? we have 2 ways to output multiple frames from decoders and that'll just be messy [17:59:42 CET] <nevcairiel> like I keep saying repeatedly, there are far easier and simpler options to get rid of the avpriv atomics to finally finish the unstable period [17:59:43 CET] <wm4> atomnuker: I'm just saying it's barely related to the new API [17:59:56 CET] <Chloe> av_register_codec() no longer actually registers an external codec [18:00:13 CET] <wm4> atomnuker: the old API could return multiple frames per packet, the new API can [18:00:30 CET] <wm4> just that the new API makes this simpler (probably) [18:00:45 CET] <atomnuker> nevcairiel: welp, a month ago you volunteered to replace one to show how its done [18:00:48 CET] <wm4> Chloe: IMO that's fine because the only API users doing this would have violated the API anyway [18:01:17 CET] <jamrial> i'm with nevcairiel, lets just remove the lavu atomic stuff in the _register() functions using static mutexes for now [18:01:32 CET] <jamrial> we can then see about Chloe patches after a few review rounds [18:01:37 CET] <Chloe> wm4: this is what I assumed. Then my patches can wait [18:01:44 CET] <atomnuker> wm4: the subframes thing is a hack and the doxygen comment says only to use it as a last resort [18:01:55 CET] <Chloe> jamrial: they still need a bit of work, I also need to do lavfi so i think it'd be better to not rush them [18:01:59 CET] <atomnuker> so yeah, it'll simplify things [18:02:59 CET] <KGB> [13FFV1] 15retokromer opened pull request #95: add v4_notes (06master...06patch-2) 02https://git.io/vNfY6 [18:06:55 CET] <atomnuker> wm4: could you check/okay the 2 patches on the ml then? [18:07:23 CET] <atomnuker> I can make a new ifdef for the second patch if you think its better [18:07:45 CET] <wm4> why bother, you've recently pushed a much larger change against my disagreement [18:07:58 CET] <wm4> personally I don't care of the CAP disappears though [18:08:28 CET] <nevcairiel> the cap does nothing but control a warning anyway [18:08:50 CET] <wm4> yep [18:08:55 CET] <wm4> and it's sometimes wrong [18:10:29 CET] <durandal_1707> so? should i do crossover filter until atomnuker does atrac9? [18:10:57 CET] <cone-666> ffmpeg 03wm4 07master:2477bfe22121: http: avoid logging reconnect warning if stream was aborted [18:11:16 CET] <SortaCore> is there a code style checker for ffmpeg patching? [18:11:38 CET] <SortaCore> I want to fix Intel QSV so I don't have to change the implementation from any to hardware-any every time [18:11:39 CET] <jamrial> SortaCore: tools/patcheck [18:12:05 CET] <jamrial> it can give some false positives, though [18:12:11 CET] <SortaCore> how use? [18:12:14 CET] <SortaCore> I'm on Windows [18:12:29 CET] <SortaCore> ah, it's a bash thing? [18:12:34 CET] <SortaCore> I'll run it under mingw then [18:13:30 CET] <jamrial> yes [18:16:46 CET] <atomnuker> wm4: because if you don't review it then who else will? [18:18:03 CET] <atomnuker> durandal_1707: could you do some kind of an image fingerprinting filter? [18:18:33 CET] <atomnuker> e.g. feed it an image, it spits out some kind of a fingerprint and you can then compare it to other fingerprints of images [18:19:01 CET] <atomnuker> so you could check an entire directory for identical/low resolution copies of images [18:19:40 CET] <durandal_1707> atomnuker: ask youtube to dump its code for checking copyright of uploads [18:20:47 CET] <atomnuker> that's how you get doublecrossed, you get it, they say you've gotten it illegaly and you're off to court [18:21:50 CET] <atomnuker> (it has so many false positives these days no one with over 10000 subscribers is okayd afaik) [18:22:46 CET] <durandal_1707> isnt there already gpl version of filter that does same thing? [18:29:45 CET] <jamrial> nevcairiel: something like this? https://pastebin.com/za3CcxwH [18:30:57 CET] <nevcairiel> jamrial: the logic is really confusing for me, does it really put stuff at the top of the parser list? [18:31:35 CET] <jamrial> nevcairiel: i basically reverted the functions to the point they were before the atomic rewrite was done, then added the mutexes [18:31:48 CET] <nevcairiel> its probably fine then [18:32:11 CET] <nevcairiel> its still practically impossible to test their thread safety [18:32:41 CET] <wm4> atomnuker: what was the title? [18:34:03 CET] <cone-666> ffmpeg 03James Almer 07master:b4eeffffc893: configure: update minimum required version of libvmaf [18:36:22 CET] <atomnuker> wm4: "avcodec: move the old decoding/encoding API under and ifdef", second patch for the subframes cap is a reply to it [18:37:53 CET] <wm4> atomnuker: jamrial apparently had comments [18:47:22 CET] <durandal_1707> atomnuker: there are bunch of ways how to do fingerprinting, haar wavelet seems nice for this [18:53:35 CET] <atomnuker> makes sense that wavelets would be suitable, they essentially store progressively downsized images + diffs [18:55:11 CET] <kierank> michaelni: what is the relevance of 03b82b3ab9883cef017e513c7d0b3b986b3b3e7b to "Do not attempt to render into frames already output" [18:56:22 CET] <wm4> fuzzing [18:56:25 CET] <michaelni> kierank, the testcase doesnt work before that commit, it seems decoding goes a slighty different path [18:56:52 CET] <kierank> apart from OOM I don't see how that can be the case [18:56:58 CET] <kierank> well, threading [18:57:05 CET] <kierank> as well [18:58:22 CET] <michaelni> i dont know / didnt debug this to the point to know exactly but there where warnings aboutz overread, so maybe something after the initialized arrays threw it in a different branch [19:22:53 CET] <cone-666> ffmpeg 03James Almer 07master:414a49d6710d: ffmpeg: use thread wrappers for the thread message functionality [19:22:54 CET] <cone-666> ffmpeg 03James Almer 07master:8d9c9775b248: avutil/log: use thread wrappers for the locking functionality [21:25:10 CET] <SortaCore> durandal_1707: why'd you drop the yuvj removal? [21:28:06 CET] <durandal_1707> SortaCore: not enough motivation [21:28:37 CET] <SortaCore> not enough of Shia LaBeouf? [21:34:29 CET] <atomnuker> durandal_1707: plz deprecate yuvj or it won't be gone for 4 years [21:35:19 CET] <atomnuker> jkqxz: sent a vaapi patch to the ml [21:35:31 CET] <CoreX> so this guy last night was having a problem and i get the same thing [21:35:32 CET] <CoreX> [00:42:35] <Guest39271> Hi, I'm trying to install ffmpeg with H.264 from source. I installed H.264 libraries using VideoLan. However, when compiling ffmpeg, its libavcodec/libx264.c is looking for a defined name "x264_bit_depth", which does not exist in either x264_config.h or x264.h generated by VideoLan H.264. Does anyone know how I could fix this? [21:35:37 CET] <CoreX> whats causing it ? [21:36:15 CET] <philipl> Apple joins AOM. Seems huge. [21:40:30 CET] <atomnuker> their plan to bet on hevc failed [21:40:33 CET] <durandal_1707> SortaCore: who? [21:40:51 CET] <jdarnley> CoreX: your ffmpeg is too old for the new changes to x264 [21:40:57 CET] <microchip_> CoreX: recent changes to x264. get ffmpeg git master [21:41:00 CET] <philipl> the patent licensing monster ate its children. [21:42:52 CET] <jkqxz> atomnuker: Why? IMO the X11 thing isn't really wanted there any more because it causes more confusion than it helps. (Though I haven't actually tried to do that because compatibility something something.) [21:43:53 CET] <wm4> philipl: holy shit [21:44:18 CET] <wm4> maybe they joined only to sabotage it trollol [21:45:27 CET] <rcombs> philipl: source? [21:47:14 CET] <rcombs> oh http://aomedia.org/about-us/ [21:47:51 CET] <atomnuker> jkqxz: what do you mean? [21:48:17 CET] <KGB> [13FFV1] 15michaelni closed pull request #88: Make the topmost value of the border column explicit (06master...06topmost) 02https://git.io/vNvVI [21:48:48 CET] <atomnuker> the point is to be able to just do -hwaccel vaapi and not have to find your drm handle and do -vaapi_device /dev/dri/renderxxxx [21:48:57 CET] <jkqxz> In Wayland, why would you ever use not-DRM? [21:49:14 CET] <jkqxz> To do anything with Wayland you need to manage your own connection anyway. [21:49:35 CET] <jkqxz> (The stupid window system integration stuff which shouldn't be in libva at all, I mean.) [21:49:49 CET] <atomnuker> well fine, should I submit a patch to remove the x11 stuff? [21:49:57 CET] <wm4> some drivers certainly needed it [21:50:09 CET] <wm4> but with DRM everything interops nicely I guess [21:50:29 CET] <wm4> is there any reasonable logic which DRM device should be selected? [21:52:20 CET] <philipl> rcombs: yeah. [21:52:41 CET] <philipl> It seems so weird that Apple sat on the fence for so long, and then jumped on hevc when it was already clear which way the wind was blowing. Then this. [21:54:40 CET] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vNfiZ [21:54:40 CET] <KGB> 13FFV1/06master 1411c75cb 15Jérôme Martinez: Clarify the difference between Pixel and Sample... [21:56:24 CET] <cone-666> ffmpeg 03Misty De Meo 07master:fde057dfb25e: aiff: add support for XA ADPCM [21:56:50 CET] <jkqxz> atomnuker: I don't strongly mind it, I just thing it's kindof confusing to have the window system connections when you can't actually do anything with them (IMO having X11 originally was a mistake). [21:57:24 CET] <jkqxz> The putting it before X11 is presumably because xwayland doesn't have DRI2 and that barfs somehow in libva IIRC? [21:58:01 CET] <KGB> [13FFV1] 15JeromeMartinez opened pull request #96: Makes bits_per_raw_sample more generic (06master...06bits_per_raw_sample) 02https://git.io/vNfiV [21:58:41 CET] <KGB> [13FFV1] 15ablwr opened pull request #97: change definitions from use of luminance to use of luma (06master...06ab/luma-in-definitions) 02https://git.io/vNfiM [21:58:54 CET] <jkqxz> (I know the libva X11 init aborts when run over ssh forwarding, which does annoy me. I should fix that at some point...) [21:59:12 CET] <wm4> "aborts"? [21:59:26 CET] <wm4> as in tries to use it and fails instead of using DRM? [22:00:23 CET] <jkqxz> Yeah, I mean aborts. [22:00:49 CET] <wm4> abort()? [22:01:51 CET] <jkqxz> <https://0x0.st/sZKm.txt> [22:02:30 CET] <wm4> oh that shit [22:02:40 CET] <wm4> it should at least have been XCB [22:02:51 CET] <wm4> because X11's "error handler" is ridiculous [22:03:35 CET] <wm4> it aborts by default and it's process global [22:03:57 CET] <jkqxz> Yeah. Can't turn an X11 connection into an XCB connection though, so hysterical raisins fuck you over. [22:04:18 CET] <jkqxz> DRI3 only works over XCB, too, which is a pain. [22:05:10 CET] <wm4> would we need that? vaapi has a xcb API right? [22:05:33 CET] <wm4> also there's some sort of x11/xcb interop, but forgot how it works [22:05:50 CET] <wm4> modern xlib actually uses xcb [22:05:57 CET] <atomnuker> I don't think so, there's no xcb mention anywhere in va/va_x11.h [22:06:11 CET] <jkqxz> Yeah. You can make an X11 connection from XCB, but not the other way around. [22:06:16 CET] <jkqxz> No XCB in VAAPI, yeah. [22:06:27 CET] <wm4> oh, weird [22:06:33 CET] <wm4> pretty fucked up [22:08:07 CET] <wm4> (ok, you can get a xcb_connection_t from a X11 Display, useless here) [22:11:49 CET] <atomnuker> jkqxz: if you don't mind it can you give me an lgtm? its not big and can be removed alongside the x11 thing whenever you get around to it [22:12:31 CET] <jkqxz> Sure. [22:12:42 CET] <jkqxz> I assume there is no better way to identify a wayland display name? [22:16:56 CET] <atomnuker> there is, through the WAYLAND_DISPLAY variable [22:17:44 CET] <atomnuker> wl_display_connect reads it when the string its given is NULL [22:18:28 CET] <wm4> shouldn't you pass the device string there [22:18:30 CET] <jkqxz> You shouldn't always pass NULL if there is a device string. You'll connect to WAYLAND_DISPLAY by default even with device = ":1" something. [22:18:33 CET] <jkqxz> I replied. [22:18:35 CET] <wm4> the av API one [22:19:00 CET] <durandal_1707> nothing can motivate me to work on yuvj shit removal, except $$$, maybe as next gsoc [22:20:18 CET] <atomnuker> I guess I can, I just copied what x11 did [22:20:21 CET] <atomnuker> durandal_1707: :( [22:21:28 CET] <jkqxz> X11 doesn't pass NULL. [22:21:42 CET] <cone-666> ffmpeg 03Humberto Ribeiro 07release/3.4:7c3d519df90f: libavutil/hwcontext_dxva2: Add check for possible errors from GetAdapterDisplayModeEx [22:21:43 CET] <cone-666> ffmpeg 03wm4 07release/3.4:4a53ecb12ead: hwcontext_dxva2: initialize D3DDISPLAYMODEEX correctly [22:36:03 CET] <atomnuker> sent a v2 [22:37:23 CET] <jkqxz> Er, what? Just pass the device string directly. If it's null then the library reads the environment variable (or at least I assume it does, that's what X does). [22:38:39 CET] <atomnuker> oh, I get it [22:40:35 CET] <wm4> I also sent a comment [22:42:54 CET] <cone-666> ffmpeg 03Marton Balint 07master:f528c49c7cd1: avfilter/vf_framerate: calculate interpolation as integer [22:45:04 CET] <KGB> [13FFV1] 15michaelni pushed 2 new commits to 06master: 02https://git.io/vNfDd [22:45:04 CET] <KGB> 13FFV1/06master 1449a47b1 15Jérôme Martinez: Add get_bits() definition [22:45:04 CET] <KGB> 13FFV1/06master 14cfb65d4 15Jérôme Martinez: Change "current position" to "pointer"... [22:45:26 CET] <atomnuker> wm4: so you want something like "if (!display && device && !strcmp(device, "wayland"))"? [22:45:56 CET] <atomnuker> but what if device is NULL? [22:46:14 CET] <atomnuker> wayland's just going to get skipped which would make the patch useless [22:49:11 CET] <wm4> if device is NULL that can probably be considered using "autodetection" [22:49:18 CET] <wm4> so using wayland in that case would be fine [22:49:36 CET] <wm4> I don't know how X11 device names are formatted and whether the string "wayland" could be valid [22:49:53 CET] <wm4> sounds far fetched though, so why not [22:52:57 CET] <atomnuker> what's the issue with just leaving it as it is? [22:53:24 CET] <jkqxz> I think it's fine just checking for not '/' (assuming '/' isn't valid). [22:53:37 CET] <atomnuker> it isn't [22:54:14 CET] <jkqxz> Though if all wayland display names contain "wayland" that would be a nice check. [22:54:25 CET] <jkqxz> (I've no idea if they do. Google isn't helping much.) [22:54:58 CET] <jkqxz> And then if you have an X11 display called "wayland" and it goes wrong then you will have deserved it. [22:56:43 CET] <atomnuker> jkqxz / wm4: does this look okay then https://0x0.st/sZKx.patch [22:56:56 CET] <atomnuker> it'll handle the case where you have an x11 display called wayland [22:58:34 CET] <jkqxz> Wrong version? That still has the incorrect NULL case. [23:02:23 CET] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vNfSr [23:02:23 CET] <KGB> 13FFV1/06master 14fdc65b7 15Jérôme Martinez: Makes bits_per_raw_sample more generic... [23:05:54 CET] <atomnuker> jkqxz / wm4: bloody hell git, https://0x0.st/sZZb.patch [23:06:34 CET] <atomnuker> (couldn't commit just amend too if there's any cached changes) [23:08:12 CET] <jkqxz> Fine with me. [23:08:24 CET] <jkqxz> (Commit message needs a fixup too.) [23:11:42 CET] <wm4> that completely ignores anything I said on the post on the ML [23:15:53 CET] <atomnuker> what you said makes no sense [23:16:04 CET] <atomnuker> the display name may be NULL [23:16:26 CET] <atomnuker> in which case checking for whether its a valid prefix is pointless and it'll get skipped [23:17:26 CET] <wm4> ... [23:17:45 CET] <wm4> if you add wayland, any user who tried to pass an X11 display name will break [23:23:47 CET] <durandal_1707> how? [23:24:43 CET] <atomnuker> no it doesn't, just "Cannot connect to a Wayland display" gets printed (verbose) and it the code carries on trying x11 [23:24:45 CET] <jkqxz> It shouldn't, should it? The wayland connect thing will just go "wtf is this shit, return error". [23:25:01 CET] <jkqxz> Unless the names really do collide, in which case you lose anyway. [00:00:00 CET] --- Fri Jan 5 2018

1 0

ffmpeg.log.20180104
by burek 05 Jan '18

05 Jan '18

[00:03:23 CET] <kingsley> durandal_1707: I use vlc, melt and ffplay. [00:31:46 CET] <kingsley> furq: FYI: I reported the bug to kdenlive at [00:31:54 CET] <kingsley> https://bugs.kde.org/show_bug.cgi?id=388503 [00:32:01 CET] <kingsley> and updated debian's bug report at [00:32:25 CET] <kingsley> https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=884268 [00:33:20 CET] <kingsley> Changing line 6 of my local copy of /usr/share/kdenlive/export/profiles.xml seems to have worked around the bug. [00:33:25 CET] <kingsley> Thanks again. [08:20:29 CET] <fletchii> Hi, I'm having a bit of an issue doing hardware acceleration with NVIDIA cards - actually, the full transcoding process is working fine, but it's all starting to fall apart when using the scale_npp filter and when trying to add an overlay to the video, the latter being a topic for another day. When I try to scale the video using scale_npp my output file just becomes green with a really low file size and therefore bitrate. [08:20:49 CET] <fletchii> I've followed everything on the HWAccelIntro page, here's the command and output: https://pastebin.com/TFHgADEN [08:24:28 CET] <fletchii> Scaling using NVDEC seems to work just fine, that said I'm planning to transcode the same file into multiple variants in one go. So scaling with the decoder isn't really an option, unfortunately. [12:53:46 CET] <broman> how to stream a pipe? I always get an error "illegal dc vlc" or "mcbpc damaged at xx xx" [13:03:24 CET] <DHE> you might need to specify information about the container/codec then. that means the stream is corrupted [13:03:31 CET] <DHE> unless, of course, the stream actually is corrupted [13:21:18 CET] <iAmSlow> hi [13:21:29 CET] <iAmSlow> https://trac.ffmpeg.org/wiki/StreamingGuide [13:21:38 CET] <iAmSlow> am reading this and have some confusion [13:21:57 CET] <iAmSlow> i want to stream from my linux pc to rpi [13:22:18 CET] <iAmSlow> the point that moste confuses me is [13:22:19 CET] <iAmSlow> ffmpeg -f dshow -framerate 20 -i video=screen-capture-recorder -vf scale=1280:720 -vcodec libx264 -pix_fmt yuv420p -tune zerolatency -preset ultrafast -f mpegts udp://236.0.0.1:2000 [13:23:14 CET] <iAmSlow> udp://236.0.0.1:2000 ? shoudent i just specify runing poort on streaming mashine and just do 192.168.1.5:port and connect to stream? [13:24:57 CET] <iAmSlow> and whenwer i use -dshow i get Unknown input format: 'dshow' [13:29:39 CET] <iAmSlow> how do i stram p2p [16:45:48 CET] <pos> so, i've got these rtsp sources which, when using ffmpeg with -vcodec copy, end up with funky files. let's say i've captured 15 seconds of video, i get a video file sized about 5MB but all players think it's 50 minutes long [16:45:53 CET] <pos> <pos> mplayer/mpv complain about invalid timestamps, vlc crashes [16:48:36 CET] Action: kepstin hopes you opened a ticket against vlc about that, players crashing on certain input files is sometimes a security problem [16:52:55 CET] <pos> oh, vlc 2.x really doesn't play well with stuff made with lavf57 [16:53:27 CET] <pos> in a lot of cases. vlc 3rc appears to have a fix [17:10:08 CET] <AlexMG> i've been using ffmpeg to test streams for errors using `-v error` switch, then i was looking deeper into it and from what i have read it seems to ffmpeg tries to encode the stream to null and find errors allthough i could not find any concreate proof on that. I have couple cmd examples here: https://pastebin.com/GvfBa2v7 [17:11:53 CET] <AlexMG> if i am looking to find errors only while demuxing/decoding the stream without re-encoding will the first example on https://pastebin.com/GvfBa2v7 suffice? if not what is then the best approach to detect stream errors only while decoding? [17:22:01 CET] <c_14> yeah [17:22:12 CET] <c_14> ffmpeg only detects fatal errors though [17:22:19 CET] <c_14> not sure what your use case is [17:22:34 CET] <c_14> there is -xerror to make it stricter in some cases [18:33:09 CET] <esc-reality> FFMPEG doesn't correctly follow the ID3v2.4 specs (http://id3.org/id3v2.4.0-frames) The following is stated there; All text information frames supports multiple strings, stored as a null separated list, where null is reperesented by the termination code for the charater encoding. [18:33:34 CET] <esc-reality> i.e. ffprobe only shows the first value in the tag [18:37:06 CET] <esc-reality> Look at the differences between mid3v2 (https://pastebin.com/Q6QEXVsL) and ffprobe (https://pastebin.com/E2027vMF) on the TPE1 and TCON fields. [18:42:43 CET] <esc-reality> Is there something I'm missing, or should I write a bug report? [19:17:10 CET] <hexhaxtron> I got 5 m4v videos. How can I join them together without encoding? [19:23:50 CET] <EvalCopy> Alright, so (I hope i'm in the right channel,) when converting from libx264(high) at 15150 kbps to zmbv pal8 I try to set the average bitrate significantly lower due to the decreased resolution, but no matter what i set with bv or min/maxrate it starts churning out things x4 as large. [19:26:39 CET] Last message repeated 1 time(s). [19:26:39 CET] <EvalCopy> whoops [19:28:06 CET] <kepstin> EvalCopy: zmbv doesn't support targetting a bitrate, it just compresses as much as it can and no more. [19:28:37 CET] <EvalCopy> ohhh. [19:28:45 CET] <kepstin> as is typical for lossless codecs [19:28:54 CET] <EvalCopy> I wasn't aware it was lossless. my bad, guys. [19:29:01 CET] <EvalCopy> that explains a lot [19:29:41 CET] <kepstin> I mean, it only supports pal8, so converting from something else to pal8 will incur some loss. but the codec itself is lossless :) [19:29:52 CET] <EvalCopy> I appreciate it! [19:30:31 CET] <hexhaxtron> kepstin: can you tell me? [19:43:02 CET] <AlexMG> i have ts file and streams, and i am looking for a way to detect if the stream or the ts file has any issues during decoding like missing frames, corrupt frame, etc... for which i am trying to figure out if ` ffmpeg -v error -i file.ts -f null -` is good enough or i should try to use `ffmpeg -v error -c:v copy -i file.ts -f null -` ? [19:43:09 CET] <furq> hexhaxtron: https://trac.ffmpeg.org/wiki/Concatenate#demuxer [19:43:16 CET] <AlexMG> https://pastebin.com/GvfBa2v7 [19:43:32 CET] <furq> AlexMG: copying is actually worse, it won't decode the stream [19:43:47 CET] <AlexMG> ah [19:44:06 CET] <AlexMG> so this is the best: `ffmpeg -v error -i file.ts -f null -` [19:44:12 CET] <furq> there's also -xerror which will bail out with an exit status on error [19:44:20 CET] <furq> but sadly not on warnings [19:44:49 CET] <AlexMG> i am trying to capture all issues that occur, to improve the stream [19:44:57 CET] <furq> yeah just stick with what you had then [19:45:51 CET] <AlexMG> than you [furq], one more question, the cli, does it demux/decode only or, does it try to encode too? [19:46:41 CET] <furq> that command line just decodes [19:48:01 CET] <AlexMG> ah perfect, i wanted to be sure what it does, as i have read from one of the forums that command encodes to null also, which is something i did not want to do [19:50:17 CET] <ascheel> can someone tell me why this is only grabbing 4 images from a directory totally over 19,000? https://gist.github.com/ascheel/90cdf0871fad66fe5fc282132eb65f7a [19:50:38 CET] <furq> ascheel: are all the images sequentially numbered [19:50:39 CET] <ascheel> Joining 19,000+ images into a video. [19:50:47 CET] <furq> if they're not then you'll need to use a glob [19:50:48 CET] <ascheel> furq: They are. [19:51:31 CET] <ascheel> furq: https://gist.github.com/ascheel/f415247622f88f82ec30b4388599a440 [19:51:44 CET] <ascheel> That's just the first 10, but it does show the sequential order [19:51:56 CET] <ascheel> mediainfo shows only 4 frames. [19:52:13 CET] <ascheel> There are no errors, it just doesn't grab the full gamut of images [19:52:46 CET] <ChocolateArmpits> ascheel, have you tried a more recent version of ffmpeg ? [19:52:59 CET] <furq> yeah that looks as though it should work [19:53:01 CET] <ascheel> ChocolateArmpits: This was just pulled from nux-dextop's repo. [19:53:08 CET] <furq> 2.6.8 is pretty old [19:53:23 CET] <ascheel> I'll compile from source. CentOS doesn't like to have modern repos. [19:53:45 CET] <furq> but yeah you could try -pattern_type glob -i G00*.JPG [19:53:59 CET] <ascheel> I'll try that, first. Thank you. [19:54:01 CET] <furq> oh hang on a minute [19:54:04 CET] <furq> -rw-r--r--. 1 arts arts 20998 Jan 4 06:51 G0010088.JPG [19:54:05 CET] <furq> -rw-r--r--. 1 arts arts 24301 Jan 4 06:51 G0020089.JPG [19:54:08 CET] <furq> G001, G002 [19:54:35 CET] <ascheel> Huh.... good catch. WTF happened there? [19:54:41 CET] <furq> something fun no doubt [19:54:45 CET] <furq> glob should still work with that anyway [19:55:33 CET] <ascheel> yep. Globbing is OK with it. Need to investigate that. [19:56:39 CET] <ascheel> Going to trash those first x images, anyway. [19:56:42 CET] <ascheel> Much appreciated, folks. [20:33:33 CET] <TheWild> hello [20:39:56 CET] <TheWild> soooo... I wanted to ask something but it pointed out to be me not reading manual with understanding. [20:49:42 CET] <ayum> there is a filter has a flags option, and I want to specify multiple flags for this option. but how to do? "flags=draw_bbox,enable_sdl" seems wrong [20:52:38 CET] <DHE> what filter is this? I don't see anything like that in the docs [20:52:55 CET] <ayum> I am writing a filter [20:53:09 CET] <ayum> can I show you some code? [20:55:55 CET] <ayum> https://paste.ofcode.org/sJWngH8etMyLkxv72xR33M [20:56:55 CET] <ayum> it's my AVOption, this filter named "compare", but I want to specify some flags when I need debug the filter. [20:57:16 CET] <ayum> like: "compare=flags=draw_bbox:sync_frames=64" [20:58:09 CET] <ayum> but is it possible to specify multiple flags? like this one: "comapre=flags=draw_bbox,open_sdl:sync_frames=64". [20:59:48 CET] <ayum> I think I found how to do it. just add "+" sign. [21:00:40 CET] <ayum> "comapre=flags=+draw_bbox+open_sdl:sync_frames=64". if you need specify multiple flags, just need add "+" sign [21:05:50 CET] <kepstin> yeah, or a - sign to remove a flag. I think it uses the same parser as other "flag" type options in ffmpeg, e.g. on codecs and formats. [21:07:19 CET] <ayum> yes, I didn't found any information how to set multiple flags in avfilter options, but I remember some codec settings supports "-option +flags1,+flags2". [00:00:00 CET] --- Fri Jan 5 2018

1 0

ffmpeg-devel.log.20180103
by burek 04 Jan '18

04 Jan '18

[00:06:20 CET] <Chloe> How does lavd depend on lavf? (like where in the build system is this specified?) [00:11:47 CET] <jkqxz> Isn't it just the lavfi input which depends on lavfi, not lavd as a whole? (Which is <http://git.videolan.org/?p=ffmpeg.git;a=blob;f=configure;h=606cdd0004ea8b2b…>.) [00:13:18 CET] <Chloe> jkqxz: https://0x0.st/sP2w.txt this is the source of my confusion [00:13:51 CET] <Chloe> (and pretty sure the whole of lavd depends on lavf considering it uses av_oformat_next() etc atm) [00:14:44 CET] <Chloe> Those functions are new api functions I've written and are just in alldevices.c [00:15:05 CET] <Chloe> It links fine into ffmpeg_g and all the tools but not that test [00:15:18 CET] <jkqxz> Bleh, I confused lavf and lavfi. [00:15:42 CET] <jkqxz> Then it's <http://git.videolan.org/?p=ffmpeg.git;a=blob;f=configure;h=606cdd0004ea8b2b…>. [00:17:55 CET] <Chloe> jkqxz: do you know if it would be possible to make lavf depend on lavd (if lavd was enabled) [00:18:19 CET] <nevcairiel> that sounds rather backwards, lavd depends on lavf [00:18:23 CET] <jkqxz> The dependencies need to be acyclic for linking to work. [00:20:12 CET] <jkqxz> atomnuker: There, that can do silly transitions as well. [00:21:53 CET] <Chloe> ehh. ok [00:22:11 CET] <Chloe> I guess I can initialise lavf from lavd if lavd is enabled [00:22:53 CET] <Chloe> I lied, I cant. [00:26:48 CET] <Chloe> https://github.com/eintw1ck/FFmpeg/blob/master/libavformat/allformats.c#L504 could someone take a look at this. `make ffmpeg` links and works, but `make fate-api-seek` doesnt link despite using the exact same commandline (apart from some extra objects in ffmpeg one, but I checks the symbols in those objects and they dont have either of the missing functions in them). [00:31:17 CET] <wm4> I'm not sure if you really can get away with a circular dependency, but dunno [00:34:44 CET] <Chloe> https://0x0.st/sP2d.txt make ffmpeg and make-fate-api-seek commandlines alone with the extra symbols [00:35:15 CET] <Chloe> wm4: I was thinking I could move that init function into lavd if lavd was enabled, but then lavf cant call it with the pthread_once stuff anyway [00:36:25 CET] <Chloe> the other alternative is to move all the functions which call av_format_next_init into lavd if lavd is enabled [00:36:43 CET] <nevcairiel> i can already tell you that such a patch will never fly [00:37:49 CET] <nevcairiel> we should just add mutexes to the register function, delete the atomics, and leave unstable time already [00:37:50 CET] <wm4> I still think lavd should register itself with an avpriv function that sets an array ptr in lavf [00:38:19 CET] <nevcairiel> we have statix mutexes now, afterall [00:38:22 CET] <nevcairiel> static* [00:38:38 CET] <wm4> and have the iteration functions switch over to that array when it's set and you're at the last lavf item [00:39:56 CET] <Chloe> nevcairiel: what really needs to happen is for devices to stop being in lavf [00:40:13 CET] <Chloe> that actually fixes everything [00:40:32 CET] <nevcairiel> feel free to work on that for the next 2 years, what we need is something now to drop the atomics [00:41:46 CET] <Chloe> if we're bumping major anyway cant we just remove devices from lavf [00:43:10 CET] <nevcairiel> even with a bump breaking things over night and expecting people to use something brand new is rather bad form [00:44:06 CET] <wm4> we could merge lavf and lavd into 1 lib, but even that seems somewhat controversial [00:44:13 CET] <Chloe> people will cry for a short while, but there's just no way to fix this slowly [00:44:28 CET] <Chloe> merging lavf and lavd is the only way to fix this slowly [00:44:44 CET] <Chloe> but I figured that's just as bad as a sudden change [00:45:08 CET] <wm4> would what I suggested not work [00:45:33 CET] <nevcairiel> there is no pressing reason to really break it right now, the only pressing thing right now is cleaning up the old atomics, which this all degraded from [00:46:57 CET] <Chloe> wm4: maybe [00:48:37 CET] <jkqxz> The libav suggestion was to just dump lavd into avconv/ffmpeg. Pretty much all of the use-cases are there; library use of it is not particularly sane. [00:49:47 CET] <Chloe> wm4: yeah actually, it would work. [00:50:58 CET] <jkqxz> Does it still work if lavd is dynamically loaded later on? [00:51:00 CET] <wm4> jkqxz: heh, that's somewhat true, but the problem is still that ffmpeg.c requires the demuxer API for all reading [00:51:29 CET] <wm4> Chloe: and "what I suggested" meaning that stuff about using an avpriv function? [00:51:55 CET] <Chloe> jkqxz: yes it would [01:10:12 CET] <Chloe> wm4: yeah that fixes it [01:10:16 CET] <Chloe> passes [01:10:49 CET] <wm4> so how exactly does it work now? [01:11:23 CET] <Chloe> wm4: https://github.com/eintw1ck/FFmpeg/commit/faca84969654968e0390c5a8889083950… [01:11:42 CET] <Chloe> lavf stores a pointer to lavd's arrays [01:12:04 CET] <Chloe> which is passed in through an avpriv function when lavd is initialised [01:12:20 CET] <Chloe> lavf is then reinitialised with the lavd components [01:15:14 CET] <wm4> why reinit? [01:20:34 CET] <Chloe> I mean, it just adds them to the linked list [01:22:24 CET] <jamrial> if we're doing this, might as well take the oportunity to rewrite lavd from scratch so it may stop being a lavf hack :p [01:22:54 CET] <iive> +1 [01:23:44 CET] <Chloe> these changes allow the beginning of that [02:37:20 CET] <atomnuker> jkqxz: oh cool [02:37:25 CET] <atomnuker> can you add -opencl_device too? [02:37:58 CET] <atomnuker> actually what would happen if you used different devices for vaapi and opencl? e.g. an intel and an amd? [02:38:39 CET] <atomnuker> would you need hwmap or would mesa be smart enough to transfer it somehow? [02:45:07 CET] <jamrial> rcombs: does that audiotoolboxenc patch look good? [02:47:59 CET] <rcombs> jamrial: looks fine [02:49:07 CET] <jamrial> rcombs: ok, will apply it then [03:02:56 CET] <jamrial> rcombs: nevermind, sent a review about a small issue [04:00:04 CET] <stevenliu> RiCON: cannot right now, dashdec not support multi bitrate now [05:09:28 CET] <Darkie__> hey, im using github.com/filoe/cscore which binds to ffmpeg. Im using it on a linux machine and it outputs some weird logs (https://i.imgur.com/rix3Pfd.png) my tool and csc dont do such things, any ideas where those could come from? [09:19:50 CET] <mistym> Argh, only just noticed I forgot to include a changelog with a patchset I submitted until after git send-email finished. [10:35:20 CET] <nevcairiel> ana [10:35:22 CET] <nevcairiel> woops [13:38:36 CET] <KGB> [13FFV1] 15dericed opened pull request #84: use version variable in makefile (06master...06use-version-variable-in-makefile) 02https://git.io/vbjVo [13:47:42 CET] <durandal_1707> you can use afir filter for binauralization too [14:37:10 CET] <KGB> [13FFV1] 15dericed opened pull request #85: line breaks and formula formatting (06master...06line-breaks) 02https://git.io/vbj63 [14:41:03 CET] <jkqxz> atomnuker: Different devices can work together transparently (after all, that was the original intent of DRM PRIME), but there may be incompatibilities because of different layouts. [14:41:43 CET] <wm4> format negotiation horror? [14:41:45 CET] <jkqxz> For the one case I can test, exporting Mesa (AMD) -> Beignet (Intel) does work, but that is probably helped by the Mesa stuff all exporting linear because it doesn't have the metadata to do anything else. [14:42:19 CET] <wm4> isn't vaapi/drm also linear? (or is that due to magical untiling) [14:42:25 CET] <jkqxz> I imagine it would probably fail in the opposite direction, but Clover (Mesa OpenCL) doesn't support enough for it to work. [14:44:15 CET] <jkqxz> VAAPI on Intel is usually Y-tiled, but you can map it as linear via an untiler in the MMU. (It also supports X-tiled (what Beignet normally uses) for some things, but can't do codec operations on it and therefore generates extra copies where it can to paper over the holes in the abstraction.) [14:44:34 CET] <wm4> interesting [14:45:15 CET] <jkqxz> Wrt negotiating this stuff, it's what the semi-mythical Unix Device Memory Allocator project is meant to solve. [14:45:21 CET] <jkqxz> So maybe in a few more years... [14:46:12 CET] <wm4> that thing that solves all problems with nvidia too? [14:46:20 CET] <wm4> (alleged) [14:46:29 CET] <jkqxz> Yeah. [15:12:57 CET] <KGB> [13FFV1] 15retokromer opened pull request #86: fix alignment in pseudo-code (06master...06patch-1) 02https://git.io/vbjXb [17:10:18 CET] <kierank> michaelni: are you aware of any crashes in h264 pred_spatial_direct_motion [17:11:40 CET] <kierank> 4:11:16 PM <"funman> (gdb) print l1mv0[0] [17:11:40 CET] <kierank> 4:11:16 PM <"funman> Cannot access memory at address 0x7ffff8353de0 [17:41:19 CET] <Chloe> wm4: do you know if AVInputFormat->next is guaranteed to be null? this is the only reason I'm setting it. I wasn't sure if it was [17:42:42 CET] <wm4> why wouldn't it? anything declared globally is 0 inited by default [17:43:10 CET] <wm4> including the case when you init structs with an initializer, but don't explicitly set a field [18:46:42 CET] <TheAlexBarney> atomnuker, to answer your wonderings about the atrac9 encoder: [18:46:59 CET] <TheAlexBarney> The encode time can take an order of magnitude longer when enabling SBR [18:46:59 CET] <atomnuker> hey you're back [18:47:30 CET] <atomnuker> I still don't understand what band extension is [18:47:36 CET] <atomnuker> what is the band extension data for? [18:48:04 CET] <TheAlexBarney> Yeah, I ended up posting what work I've done on a C port of the decoder yesterday [18:48:59 CET] <TheAlexBarney> In a nutshell, indexes into a table of coefficients for adjusting the high spectral coefficients [18:49:34 CET] <atomnuker> so bit allocation then [18:49:52 CET] <atomnuker> shouldn't it touch the scalefactors then? [18:49:55 CET] <TheAlexBarney> Not bit allocation [18:50:00 CET] <TheAlexBarney> It doesn't [18:50:26 CET] <TheAlexBarney> It's a form of SBR [18:50:30 CET] <atomnuker> oh, direct adjustments thne [18:50:40 CET] <atomnuker> ......why isn't it called SBR then? [18:50:55 CET] <atomnuker> band extension is incredibly vague [18:52:52 CET] <TheAlexBarney> Who knows. CRI calls their SBR implementation in their HCA codec HFR or High-frequency Regeneration [18:59:51 CET] <TheAlexBarney> The way the encoder works is by generating a default gradient for bit allocation. Then for each frame it encodes the audio, and adjusts the gradient based on how many bits are leftover, encoding, and repeating until the number of unused bits is minimized [19:01:37 CET] <atomnuker> welp, that's what you got to do when you're forcing a vbr codec to behave in a cbr way [19:02:44 CET] <TheAlexBarney> I don't know too much about audio codecs, but based on what I've learned from RE'ing it, ATRAC9 seems like a rather simple codec to me when compared to others, SBR aside [19:04:25 CET] <atomnuker> yep, its just aac but without the complicated and limited vector quantization [19:04:57 CET] <cone-524> ffmpeg 03Derek Buitenhuis 07master:500a9bb5ba94: lavc/options: Remove unneeded header [19:04:57 CET] <cone-524> ffmpeg 03Derek Buitenhuis 07master:6470abc74036: vf_paletteuse: Add error checking to apply_palette [19:04:57 CET] <cone-524> ffmpeg 03Derek Buitenhuis 07master:631fa0432be8: vf_paletteuse: Don't free the second frame from ff_framesync_dualinput_get_writable on error [19:07:33 CET] <wm4> so I'm thinking about how side data handling could be unified across all the things which need side data (AVPacket, AVFrame, AVStream, global de/muxer side data) [19:07:47 CET] <wm4> any ideas how this should be handled for compatibility? [19:08:04 CET] <wm4> since there isn't really a way to avoid duplicated side data (one for the old API, one for the new) [19:11:18 CET] <atomnuker> so one function to add/remove/get side data from avpacket/avframe? [19:12:32 CET] <wm4> yes, I'd probably put the side data list into an AVSideData thing, and add a pointer member to that to AVPacket/AVFrame etc. [19:13:19 CET] <wm4> or is there a better way? [19:19:45 CET] <atomnuker> how do you plan to identify between an avpacket/avframe argument? some kind of an id or maybe putting the side data fields in the same pos? [19:20:11 CET] <wm4> there'd just be a single enum for all side data [19:20:35 CET] <wm4> plenty of packet side data is just "renamed" and copied to frame side data in libavcodec anyway [19:21:12 CET] <wm4> as long as it happens in the ABI unstable period, we could even define the legacy side data constants to new ones [19:21:27 CET] <atomnuker> I thought the api would be something like av_side_data_add(void *dst, avbufferref *src) [19:21:47 CET] <wm4> so you'd make the dst argument "dynamically" typed? [19:23:07 CET] <wm4> anyway, the main problem is that the user can manipulate the existing side data directly (without even calling any API functions) [19:23:56 CET] <atomnuker> that's something that shouldn't be allowed [19:25:34 CET] <wm4> well, it is allowed [19:26:01 CET] <wm4> I'd make the new thing opaque, but the current API is like that [19:50:17 CET] <cone-524> ffmpeg 03Paul B Mahol 07master:09b24a807ad2: avfilter: add entropy filter [20:02:43 CET] <cone-524> ffmpeg 03Nicolas George 07master:9ace76697ac1: lavfi/framesync: document frame ownership for dualinput. [20:02:44 CET] <cone-524> ffmpeg 03Nicolas George 07master:29b5f3115d9f: lavfi/framesync: remove an invalid free. [21:06:52 CET] <KGB> [13FFV1] 15michaelni pushed 1 new commit to 06master: 02https://git.io/vNely [21:06:52 CET] <KGB> 13FFV1/06master 142f2133e 15Dave Rice: use version variable in makefile [21:33:41 CET] <cone-524> ffmpeg 03Jiejun Zhang 07master:677701c6b319: lavc/audiotoolboxenc: fix noise in encoded audio [22:37:38 CET] <atomnuker> do the apple audio decoding apis go through hardware decoding? do they put some apple-specific hardware audio decoders on their boards? [22:45:06 CET] <wm4> interesting question [22:45:17 CET] <wm4> on phones doing hw decoders is probably worth the trouble? [22:45:25 CET] <wm4> for audio, I mean [22:46:37 CET] <atomnuker> I thought the audiotoolbox apis were pc-specific only and weren't on iphones [22:47:20 CET] <tmm1> the encoders/decoders are available on ios [22:47:22 CET] <atomnuker> huh, apparently they're both on osx and ios [22:47:25 CET] <tmm1> some of the apis are not, but most of audiotoolbox is [22:52:29 CET] <BtbN> I'd be surprised if their PC hardware had audio decoders [22:52:37 CET] <BtbN> but on phones it seems reasonable [22:53:34 CET] <cone-524> ffmpeg 03Paul B Mahol 07master:88cbd25b193d: avfilter: pass outlink to ff_get_audio_buffer() [23:46:22 CET] <rcombs> atomnuker: wm4: audiotoolbox is hardware on iOS [23:46:48 CET] <atomnuker> oh wow, how fast is it compared to our decoder? [23:46:59 CET] <rcombs> no idea; never tested on iOS [23:47:36 CET] <rcombs> there's documented awkwardness around getting interrupted mid-decode when using hardware [23:47:36 CET] <atomnuker> nvm I can't read today, I read that as osx [23:49:33 CET] <wm4> "interrupted"? what [23:59:49 CET] <tmm1> probably if your app goes into the background [00:00:00 CET] --- Thu Jan 4 2018

1 0

ffmpeg.log.20180103
by burek 04 Jan '18

04 Jan '18

[00:31:37 CET] <kingsley> How would you test if libavcodec57 supoprts acodec=vorbis? [00:34:59 CET] <DHE> avcodec_find_{en,de}coder_by_name("vorbis"); [00:40:03 CET] <kingsley> DHE: Please elaborate. [00:40:36 CET] <DHE> you are asking about the C API, right? [00:40:55 CET] <kingsley> Oh [00:41:23 CET] <kingsley> Maybe I owe an apology for being terse and vague. [00:42:12 CET] <kingsley> I'd prefer to use the command line, or possibly just read. [00:42:28 CET] <kingsley> ffmpeg's documentation says to use acodec=libvorbis. [00:42:54 CET] <kingsley> But maybe acodec=vorbis works too. [00:43:29 CET] <DHE> $ ffmpeg -codecs | grep vorbis # or something like that [00:43:38 CET] <kingsley> The later is what kdenlive passes to melt which gives /usr/lib/i386-linux-gnu/libavcodec.so.57 a segmentation fault. [00:44:20 CET] <kingsley> DHE: Thanks. [00:46:03 CET] <kingsley> It reports... [00:46:06 CET] <kingsley> DEA.L. vorbis Vorbis (decoders: vorbis libvorbis ) (encoders: vorbis libvorbis ) [00:46:33 CET] <DHE> good. so you have full vorbis support [01:07:18 CET] <kingsley> DHE: Yeah, in theory. [01:08:46 CET] <kingsley> But, at least for me, acodec=vorbis crashes /usr/lib/i386-linux-gnu/libavcodec.so.57 crashes with a seg fault, but acodec=libvorbis does not. [01:09:48 CET] <kingsley> I'm downloading a so called "AppImage" of kdenlive now. Evidently it works. I'll try to test it. [01:09:50 CET] <furq> not that it's a solution, but the builtin vorbis encoder is terrible anyway [01:10:03 CET] <furq> assuming you can actually change what kdenlive wants to use [01:10:41 CET] <kingsley> furq: For what it's worth, I found I could work around the bug by [01:10:54 CET] <kingsley> a.) rendering to the .mkv format, and then [01:11:12 CET] <kingsley> b.) using ffmpeg to convert that to .webm. [01:11:41 CET] <kingsley> But, I'd like to fix the bug. [01:29:09 CET] <kepstin> hmm. if you disable the internal vorbis encoder at buildtime, will it use libvorbis if you request "vorbis"? [01:29:23 CET] <kepstin> or just error out? [01:33:33 CET] <DHE> I'm wondering if you have some kind of ABI error. surprised it crashes otherwise [01:46:04 CET] <kingsley> kepstin and DHE: Sorry guys. [01:46:12 CET] <kingsley> I gotta go for a while. [01:46:26 CET] <kingsley> However, much like California's might ex gubernator... [01:46:38 CET] <kingsley> U'll be bok [01:50:55 CET] <kingsley> kepstin and DHE: Feel free to type any thoughts you may happen to have on how I might test the ABI theory. [01:51:37 CET] <kingsley> I might be OK with trying to disable the internal vorbis encoder too, if it can be done at run time. [01:52:06 CET] <kingsley> I should be able to see your thoughts when I return. [01:52:08 CET] <kingsley> So [03:48:27 CET] <gp5st> hello. I'm sorry if I'm being dense with the docs. My root issue is that i'm trying to encode 192x1080@60fps and I constantly have the video drop a lot of frames. I'm on linux (ubuntu 16.04 on a thinkpad). I was exploring if hardware accel is an option and am greeted by https://dpaste.de/Hj10/raw ("Unrecognized option 'init_hw_device'") So I don't know if the answer is no or if I need a different version of ffmpeg. [03:50:17 CET] <gp5st> the source is an X framebuffer (via OBS) (all ears on dropping obs and recording a display and an audio stream from the command line) [03:57:24 CET] Action: gp5st facepalms https://trac.ffmpeg.org/wiki/Capture/Desktop I didn't see this earlier somehow. same question on if hardware acceleration is an option though :-\ [04:03:20 CET] <Fenrirthviti> gp5st: Are you using OBS to actually encode? [04:04:20 CET] <gp5st> Fenrirthviti, I guess so? It takes encoder options which appear to be passed to ffmpeg, so I'm really not sure. I'm perfectly happy moving to straight command line if I can get it working though [04:04:28 CET] <gp5st> (I don't need any of the other features of OBS) [04:05:04 CET] <Fenrirthviti> (I'm part of the OBS project team) OBS doesn't use ffmpeg to encode if you're using x264 [04:05:08 CET] <Fenrirthviti> It uses x264 directly. [04:05:22 CET] <Fenrirthviti> We only use ffmpeg if you're using the custom ffmpeg output [04:05:27 CET] <Fenrirthviti> or for nvenc [04:06:08 CET] <gp5st> ah, I see [04:06:13 CET] <Fenrirthviti> hope over to #obsproject and I can get some more info and see if I can help determine what's going on [04:06:20 CET] <gp5st> will do, thanks! [04:21:19 CET] <gp5st> circling back to the "Unrecognized option 'init_hw_device'" error, do I need a different version of ffmpeg, or is it only valid with other options [04:50:06 CET] <gp5st> ok back :) switched over to the intel drivers for my machine [07:26:04 CET] <kingsley> How would you debug a seg fault at src/libavcodec/vorbisenc.c:1073? [07:26:33 CET] <kingsley> https://bugs.debian.org/cgi-bin/bugreport.cgi?att=1;bug=884268;filename=sym… [08:26:29 CET] <pmjdebruijn> kingsley: the other questions is whether it may already be fixed in svn? did you check ffmpeg changelogs with regard to that file [08:26:42 CET] <pmjdebruijn> kingsley: (btw) the vorbisenc encoder isn't recommended for general use (libvorbis is) [08:26:52 CET] <pmjdebruijn> (that's a bit beside the point, but I thought I'd mention it) [08:41:04 CET] <hendry> hi, how do I tell if my audio stream in silent? Stream #0:1[0x1e2](und): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 96 kb/s [08:41:21 CET] <hendry> Extract the audio stream and then what tool tells me the levels are too low I wonder? [08:44:21 CET] <Spring> since I can't ask in #handbrake due to not being voice and OPs afk atm does anyone know if Handbrake supports a preset for HDR content? [08:44:27 CET] <Spring> *voiced [08:49:11 CET] <Nacht> hendry: You can use ffmpeg. ffmpeg -i audo.mp3 -af "volumedetect" -f null /dev/null [08:49:35 CET] <dradakovic> Is there any way for ffmpeg to try to reconnect to the input if input was down? My input is http radio and it seems to have a timeout after a while. I would like ffmpeg to reconnect after timeout. I use parameters -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -reconnect_delay_max 4294 [08:54:30 CET] <dradakovic> ffmpeg session is still running but nothing is of course outputted as input was down [08:55:07 CET] <hendry> Nacht: thanks! wonder how I can that into JSON so I can parse it ? https://s.natalian.org/2018-01-03/silent-output.txt [08:55:30 CET] <hendry> I guess anything less than -50.0 dB is pretty much silent? [08:58:42 CET] <Nacht> hendry: There's also silencedetect: https://www.ffmpeg.org/ffmpeg-filters.html#silencedetect [08:59:13 CET] <Nacht> hendry: Only ffprobe can output in JSON. Afaik it doesn't have a volume level detect option. I might be wrong though [09:04:59 CET] <hendry> Nacht: silence detect is interesting! again struggling to parse its output =) [silencedetect @ 0x5648acb24940] silence_start: -0.0030839 [09:05:06 CET] <hendry> I'll figure it out, thank you!! [11:09:44 CET] <NickRubat> hi [11:22:26 CET] <Fyr> guys, I need to remove some parts of the audio, how can I do that? I don't need 00:10:00-00:10:01, 00:20:00-00:20:01, 00:30:00-00:30:01 etc. [11:23:03 CET] <Fyr> I know -ss and -to, but, I don't need 7 parts of the audio. [11:25:31 CET] <Fyr> I need a reverse atrim filter [12:10:50 CET] <Fyr> guys, so nobody knows how to remove a few seconds in the middle? [12:11:31 CET] <durandal_1707> Fyr: use trim with concat? [12:11:51 CET] <Fyr> with -filter_complex? [12:12:19 CET] <durandal_1707> no, you can use simple too [12:12:37 CET] <durandal_1707> its complicated [12:38:00 CET] <DHE> maybe the aselect filter? [12:51:03 CET] <Fyr> wow, it looks cool. [12:53:17 CET] <Fyr> however, my brain is too small to comprehend how to apply it. [12:57:44 CET] <sfan5> aselect='between(mod(t\,10)\, 0\, 1)-1' [12:57:47 CET] <sfan5> something like that probably [13:23:35 CET] <dradakovic> Does anyone have any clue on how to reconnect ffmpeg to the http input in case it drops? Session stays open but if http was dropped there is nothing on the output. [13:46:10 CET] <BtbN> restart it [13:53:39 CET] <sunslide> hi guys, I'm trying to build a command-line to support both RTSP and MJPEG streams and save them to a playlist, RTSP works great but MJPEG streams aren't, https://pastebin.com/K7x4F6AR anyone? [13:59:14 CET] <sunslide> here is the output: https://pastebin.com/vQc9b64W [16:18:01 CET] <nilshi> hello. I am not sure if my x11grab recording skips frame or not. There is a "speed:" indicator during recording that never reaches 1. always at 0.8 to 0.9x. And when I end the recording it says "skipped: xy%". So I guess my recording is not fast enough to keep up? [16:18:32 CET] <nilshi> is there a flag to be super strict and abort if even for one frame recording was too slow? [16:26:17 CET] <c_14> don't think that flag exists, and yeah you're too slow [16:27:13 CET] <nilshi> it is amazing that my powerful computer seems to be too slow to record my desktop 1920x1080 at 60Hz. [16:27:32 CET] <nilshi> well... I'll read some more about ffmpeg in the meantime [16:30:21 CET] <c_14> What's your recording command? [16:31:33 CET] <nilshi> it varies :) [16:31:41 CET] <nilshi> ffmpeg -draw_mouse 0 -y -framerate 30 -video_size 1920x1080 -f x11grab -i :44 output.mp4 [16:31:43 CET] <nilshi> but also [16:31:59 CET] <c_14> try -preset fast/veryfast [16:32:03 CET] <nilshi> with -c:v libx264rgb -crf 0 -preset:v ultrafast [16:32:09 CET] <c_14> oh [16:32:11 CET] <c_14> eeh [16:32:28 CET] <c_14> try utvideo or eh ffvhuff or huffyuv [16:32:37 CET] <furq> don't bother with huffyuv [16:32:43 CET] <nilshi> I think I tried that also. Well, I can give it another try [16:32:43 CET] <furq> ffvhuff is faster and compresses better [16:32:57 CET] <furq> also don't use mp4, you'll get an unplayable file if the recording is aborted [16:32:59 CET] <furq> use mkv [16:33:40 CET] <nilshi> the thing is I have plenty of RAM and 8 cores and a Radeon RX 480. And I don't see them even maxed out in htop. not even close. Recording onto an SSD. So my computer hardware seems not to be the bottleneck [16:34:26 CET] <nilshi> furq: right. I normally do. I have around 20 very similar ffmpeg commands in a text file here and I chose one of the recent ones to show [16:34:26 CET] <furq> you could maybe use the new amd vce stuff if you have an rx 480 [16:34:41 CET] <furq> that's very new though, you'll need ffmpeg master afaik [16:34:50 CET] <nilshi> no problem, I'll compile. [16:34:52 CET] <nilshi> I am on arch [16:35:41 CET] <nilshi> first I want to test a few other things though. It may neither be the hardware nor ffmpegs fault but just some Xorg setup [16:35:45 CET] <nilshi> I am not recording my main X [16:36:00 CET] <nilshi> thanks for the input so far. [16:36:20 CET] <nilshi> furq: where can I read about amd vce stuff? I am not very experienced with ffmpeg at all [16:36:47 CET] <nilshi> like so many I do copy and paste from the internet and then look up what the flags mean. Well, the last point my already put me above the average :) [16:36:50 CET] <furq> i've not used it and i'm not sure if it's made it into the docs yet [16:36:57 CET] <furq> it only got added about a month ago [16:38:00 CET] <jkqxz> On Linux it's VAAPI, as it has been for a while. The recently-added stuff (AMF) is Windows-only so far. [16:38:11 CET] <furq> oh right [16:38:24 CET] <furq> does encoding through vaapi work properly yet [16:40:01 CET] <jkqxz> The screen capture examples on <https://trac.ffmpeg.org/wiki/Hardware/VAAPI> should work, maybe with "-profile:v constrained_baseline" or something on AMD. [16:40:33 CET] <furq> nilshi: try ffvhuff first anyway [16:40:44 CET] <furq> if that's dropping frames then it's doubtful any encoder will solve the problem [16:41:04 CET] <furq> unless you're loading up other cores ofc [16:42:31 CET] <jkqxz> Yeah. If the capture isn't running fast enough in itself then offload isn't going to help. [16:43:11 CET] <furq> x264 ultrafast ought to be good enough for 1080p60 with that cpu, but only if the cpu is relatively unloaded otherwise [16:43:40 CET] <jkqxz> (Well, kmsgrab might, but it is pretty annoying to use.) [16:56:02 CET] <mickie> Hi All, I have a .ts file from DVB-T2 and want to transcode its audio stream (aac_latm 48000 Hz, 5.1, fltp) into an AC3 also 6ch. The video stream is h264 (High). How can I achieve this? [17:04:06 CET] <nilshi> a recording speed of 0.9x is enough? does it ever start at 1 and stays there, stable? [17:04:19 CET] <nilshi> fps seems to be locked at 30 at the moment. did not try 60 with ffvhuff yet [17:04:20 CET] <DHE> ffmpeg -i inputfile -c:v copy -c:a ac3 -b:a 384k outputfile # adjust bitrates etc needed [17:04:45 CET] <DHE> nilshi: you might not get 1.0 exactly, but 0.98x is likely okay if your CPU isn't pegged [17:04:50 CET] <nilshi> some drops happened :( [17:05:43 CET] <nilshi> all 8 cores are at 25% to 30% during recoring. I have 4 GB RAM free (of 8) [17:06:39 CET] <jkqxz> Does it make any difference if you set "-vsync 0"? The timing synchronisation stuff can mess with live capturing. [17:07:10 CET] <nilshi> that is the output so far https://hastebin.com/aqifomekud.pas [17:07:43 CET] <nilshi> is there something obviously wrong? [17:08:10 CET] <nilshi> jkqxz: I'll try without vsync as well [17:08:24 CET] <nilshi> I have a long list to try things out now :) [17:08:47 CET] <mickie> Thanks DHE, will test the output now. [17:09:57 CET] <nilshi> with 60fps ffmpeg puts out Past duration 0.998070 too large [17:10:04 CET] <nilshi> like crazy. several times a second [17:10:09 CET] <nilshi> what kind of error is that? [17:28:59 CET] <nilshi> time for ffmpeg-git for the vaapi stuff [17:29:10 CET] <nilshi> or is that already released? [17:32:16 CET] <alexpigment> nilshi: that happens very frequently (past duration too large). in my experience it's when the input frame rate and output frame rate don't match [17:32:32 CET] <alexpigment> for example, if I have a 60fps video and encode it to 30fps [17:34:22 CET] <nilshi> it doesn't happen when I do 30fps. And I guess in the end 30fps would be enough for all intents and purposes. But my ambitions are higher :) [17:34:43 CET] <alexpigment> don't shortchange yourself, 30fps is not enough :) [17:34:51 CET] <alexpigment> i only work with 30i or 60p primarily [17:35:15 CET] <alexpigment> anyway, your pastebin shows -framerate 30 [17:35:18 CET] <nilshi> I'm a teacher and production of high quality educational video material where you need to see mouse movement and details.. that should have a certain technical quality [17:35:22 CET] <alexpigment> and presumably x11grab is at 60fps [17:35:42 CET] <nilshi> alexpigment: with 30fps I don't get that message that often. When I use the same command line and change to 60fps it begins spamming [17:36:17 CET] <alexpigment> what if you don't specify a -framerate value? what does the final framerate become? [17:36:18 CET] <nilshi> BUT! I think I should have mentioned that this is very experimental here. I am actually recording xvfb. That could be a total bottleneck in itself [17:36:29 CET] <nilshi> probably 0 :) I have to check [17:36:52 CET] <alexpigment> i haven't tried it with x11grab. just wondering if there's an inherent frame rate [17:37:38 CET] <alexpigment> the other thing you could try is putting a redundant -r [30 or 60] after the input is defined [17:37:54 CET] <nilshi> 29.970 frames [17:38:51 CET] <alexpigment> ok, so say that you use 60 instead like you were talking about. when you frame step the resultant video, are there a bunch of duplicate frames? [17:39:29 CET] <alexpigment> actually, i'm guessing the answer is yes; every other frame is probably duplicated [17:39:32 CET] <nilshi> is there a tool that analyses that for me? I think my video material does not allow me to distinguis [17:39:45 CET] <alexpigment> i think VLC has a frame step key [17:39:51 CET] <alexpigment> like the right arrow or something [17:39:52 CET] <nilshi> but if it is 29.97 frames and I say 60... [17:40:12 CET] <alexpigment> right, so past duration too large may be every time it makes up a frame [17:40:23 CET] <alexpigment> try just setting -framerate to 30000/1001 [17:40:30 CET] <nilshi> no -r ? [17:40:33 CET] <alexpigment> do you get the message at all at that point [17:40:34 CET] <alexpigment> not for now [17:41:38 CET] <nilshi> with that it ran for 11 seconds without anything and then the "too large" message began, many. [17:41:59 CET] <nilshi> it is dropping, not duplicating [17:42:03 CET] <nilshi> dup=0 drop=10 [17:42:18 CET] <alexpigment> hmmm, i just don't know then [17:42:37 CET] <alexpigment> sounds like your buffer is just running out then [17:42:56 CET] <alexpigment> it can't capture in realtime [17:43:44 CET] <nilshi> it happens when I record my real X :0 as well. So it has nothing to do with xvfv [17:43:49 CET] <nilshi> xvfb [17:44:08 CET] <furq> does -rtbufsize work with xcbgrab [17:44:08 CET] <nilshi> Î am running an RT-Audio system though. That may interfere [17:44:10 CET] <furq> i forget now [17:44:20 CET] <furq> if it does then you can try bumping that [17:44:30 CET] <nilshi> xcbgrab? [17:51:04 CET] <nilshi> I compiled from git now and activated my ffmpeg vaapi device. The screen capture command from https://trac.ffmpeg.org/wiki/Hardware/VAAPI spits out an error: [17:51:33 CET] <nilshi> https://pastebin.com/Rs9WSqMd [17:53:09 CET] <alexpigment> ah one of those messages :) [17:53:26 CET] <alexpigment> all the hardware encoders i've worked with will do that when you specify something it doesn't understand [17:53:31 CET] <alexpigment> or software encoders too i guess [17:53:49 CET] <alexpigment> what if you get rid of the -vf and the -qp? [17:54:37 CET] <alexpigment> and intel QSV did that on me when i didn't specify -look_ahead 0 , which is weird, but whatever [17:55:36 CET] <nilshi> different error [17:55:38 CET] <nilshi> https://pastebin.com/eRsdizr6 [17:55:40 CET] <nilshi> looks worse [17:55:54 CET] <furq> pretty sure you need the -vf [17:56:03 CET] <furq> try adding -bf 0 [17:56:04 CET] <alexpigment> ok, put the vf back in, then instead of -qp whatever, try -b:v 5M [17:56:10 CET] <alexpigment> or try what furq said [17:56:15 CET] <nilshi> that was the same I got when I tried to run that command without ffmpeg from git [17:56:28 CET] <alexpigment> seems weird that you should have to specify -bf 0 when ffmpeg knows it can't do b-frames but still [17:56:41 CET] <furq> i'm guessing it maybe doesn't know that [17:56:50 CET] <alexpigment> it says it in the pastebin [17:57:01 CET] <alexpigment> right above the error [17:57:08 CET] <furq> i assume that's a strerror from vaapi [17:57:10 CET] <furq> idk though [17:57:25 CET] <nilshi> that was better with -bf 0 [17:57:27 CET] <nilshi> https://pastebin.com/G8SmJZ94 [17:57:28 CET] <alexpigment> i'll trust your knowledge on this. i know *some* part of the process knows b frames aren't supported [17:57:41 CET] <nilshi> some drops and dups at the beginning, then 30sec stable recording before I ended [17:57:57 CET] <furq> i know very little about hwenc, i'm just saying the error message doesn't necessarily mean ffmpeg knows [17:58:08 CET] <alexpigment> that's fair [17:58:46 CET] <alexpigment> does vaapi have a speed mode? [17:58:50 CET] <jkqxz> I suspect you are going to want "-profile:v constrained_baseline" too. It defaults to high, which has some weird problems on AMD. [17:59:13 CET] <alexpigment> well, constrained baseline should at least be faster, albeit very gimped [17:59:25 CET] <jkqxz> alexpigment: Yes, it's mapped from "-compression_level". It only has much effect on recent Intel, though. [17:59:27 CET] <nilshi> I don't know what either of this means [17:59:35 CET] <nilshi> is it like fast/ultrafast? [17:59:45 CET] <alexpigment> constrained baseline is like a subset of features of h.264 [17:59:52 CET] <alexpigment> a very basic subset [18:00:55 CET] <alexpigment> anyway, i guess it's worth putting -compression level in there then. i have no idea what the scale is [18:02:07 CET] <jkqxz> It doesn't do anything at all on AMD, and will probably complain that the option isn't supported. [18:02:36 CET] <alexpigment> are you thinking the lack of b frames message indicates it's AMD? [18:02:41 CET] <alexpigment> (e.g. the 500 series)? [18:02:41 CET] <nilshi> hm... guys. the video is empty [18:03:07 CET] <furq> it's a 480 [18:03:20 CET] <nilshi> RX 480, yes. [18:03:29 CET] <nilshi> open source drivers [18:03:38 CET] <alexpigment> ah, i didn't see that info [18:03:38 CET] <jkqxz> Probably add the "-profile:v constrained_baseline"? High profile is the default and generally doesn't work properly there. [18:03:47 CET] <nilshi> alexspeller: it was before you joined [18:03:57 CET] <alexpigment> gotcha [18:05:08 CET] <nilshi> nope, still nothing [18:05:28 CET] <nilshi> or wait... [18:05:33 CET] <alexpigment> add the -framerate 30000/1001 back in at the beginning maybe? [18:05:51 CET] <nilshi> mplayer says wrong video. vlc now plays a very cropped version [18:06:05 CET] <alexpigment> hmmm [18:06:35 CET] <nilshi> 10 seconds equal 120kb of video. That doesn't sound right [18:07:02 CET] <nilshi> or it is the magical codec which stores in another dimension [18:07:08 CET] <alexpigment> well, it kinda does ;) [18:07:32 CET] <alexpigment> if the video doesn't change much over time, it doesn't use much more bitrate [18:07:51 CET] <nilshi> it is recording a terminal with asciiquarium [18:07:53 CET] <nilshi> a lot of black [18:08:02 CET] <alexpigment> sounds easy to encode [18:08:21 CET] <nilshi> I could start glxgears instead [18:08:32 CET] <nilshi> no, that is opengl. Another can of worms [18:08:48 CET] <nilshi> right now the resolution is the problem. And that mplayer does not want to play the file at all [18:09:27 CET] <nilshi> ... where is video size [18:11:03 CET] <nilshi> ok, if I actually put in the resolution it records correctly. mplayer still doesn't like it though. [18:12:57 CET] <alexpigment> do you have mediainfo or a similar program to look at the video properties? [18:13:11 CET] <alexpigment> it might be obvious what's going on by seeing that [18:13:41 CET] <alexpigment> i guess ffprobe will do the same [18:14:08 CET] <nilshi> yes [18:14:09 CET] <alexpigment> also, did you try the -profile:v constrained baseline like jkqxz? [18:14:15 CET] <nilshi> yes, it is in [18:14:20 CET] <alexpigment> k [18:14:42 CET] <alexpigment> he works a lot with hardware encoder stuff, so he's really the main authority imho [18:14:42 CET] <nilshi> https://pastebin.com/z4V3jJV3 [18:15:33 CET] <alexpigment> weird. i doubt mplayer would have a problem with the h.264 level being 5.1, but it is very high still [18:16:06 CET] <alexpigment> also, it's cabac with baseline profile [18:16:14 CET] <alexpigment> maybe it expects those two to be mutually exclusive [18:16:44 CET] <nilshi> ffmpeg -y -vaapi_device /dev/dri/renderD128 -video_size 1920x1080 -f x11grab -i :44 -vf 'hwupload,scale_vaapi=format=nv12' -c:v h264_vaapi -bf 0 -profile:v constrained_baseline output.mp4 [18:16:53 CET] <nilshi> that is the current command [18:17:27 CET] <alexpigment> try adding -coder cavlc -level 41 [18:17:37 CET] <nilshi> just before output? [18:17:42 CET] <alexpigment> right [18:19:49 CET] <nilshi> same behaviour with mplayer https://pastebin.com/pEDyt2Em [18:20:03 CET] <alexpigment> cabac is still on... [18:20:05 CET] <alexpigment> hmm [18:20:31 CET] <nilshi> maybe it is not related to this video and I got an mplayer problem, unrelated [18:20:53 CET] <alexpigment> it could be [18:21:05 CET] <nilshi> /usr/bin/mplayer: error while loading shared libraries: libswscale.so.4: cannot open shared object file: No such file or directory [18:21:20 CET] <nilshi> where does this come frome. The only thing that changed is ffmpeg from git. no other system update [18:21:27 CET] <nilshi> *shrugs* [18:23:38 CET] <alexpigment> maybe related; https://www.linuxquestions.org/questions/slackware-14/mplayer-error-libwsca… [18:24:27 CET] <nilshi> I see. I just compiled ffmpeg from git. There is a ffmpeg-full-git package as well. Described as "with all features. Maybe I need that [18:25:05 CET] <alexpigment> possibly. i'll claim ignorance [18:26:17 CET] <nilshi> it is part of ffmpeg. https://github.com/FFmpeg/FFmpeg/tree/master/libswscale [18:26:57 CET] <nilshi> I didn't install the full package because it draws in pulseaudio :( [18:40:55 CET] <sfan5> nilshi: you need to recompile mplayer if you install a git version of ffmpeg [18:43:41 CET] <nilshi> better than installing pulseaudio [18:47:02 CET] <nilshi> no luck, but I'll work with vlc for now [18:47:16 CET] <nilshi> I can always revert to the binary ffmpeg and wait for the next release. Right now I just want to find out [18:59:05 CET] <nilshi> it doesn't look good. I can't shake the feeling that something in my system may be misconfigured. [18:59:22 CET] <nilshi> I'll settle with 29.97 fps and the occasional fram dop or dup for now. [19:00:04 CET] <nilshi> last question: can I record RGB colors with vaapi ? I want to try how that looks different [19:17:17 CET] <Fyr> Can anyone explain to me why if output is 45 minutes long? [19:17:17 CET] <Fyr> ffmpeg -ss "00:28:16" -i "input.mkv" -to "00:45:52" -map 0 -c copy "output.mkv" [19:17:45 CET] <Fyr> it's supposed to be 17 minutes. [19:26:20 CET] <alexpigment> Fyr: i don't recall the difference between -t and -to but I think generally people use -to [19:26:24 CET] <alexpigment> gah [19:26:31 CET] <alexpigment> *people generally use -t is what i meant [19:26:46 CET] <alexpigment> secondly, you're defining your starting point before the input and the ending point after the input [19:27:17 CET] <alexpigment> you should put them on one side or the other [19:28:37 CET] <alexpigment> also, looking now, but i guess -to is the ending timestamp but -t is the length, so I guess that's fine. try putting the -ss and -to on the same side of the input in the command line [19:43:09 CET] <Fyr> is there a way to split file efficiently? something like -ss 00:01:00 -t 60 1.mkv -ss 00:02:00 -t 60 2.mkv [19:43:09 CET] <Fyr> etc [19:48:06 CET] <alexpigment> https://stackoverflow.com/questions/5651654/ffmpeg-how-to-split-video-effic… [19:49:48 CET] <Fyr> thanks [20:12:57 CET] <kepstin> Fyr: when you use the -ss input option, ffmpeg resets the timestamps so that '0' is the seek point, therefore the -to option will go until 45:52 after the seek point. [20:17:40 CET] <kepstin> (there are ways to avoid this, e.g. the -copyts option or using -itsoffset) [20:22:25 CET] <Mista_D> ffmpeg with two filters runs at 0.4 real time (usually its 6-7 times real time on 12 core Xeon DELL T7500). Any advice please? [20:22:25 CET] <Mista_D> ffmpeg -i 1080p_2997_tc_yuv422p.mxf -map_metadata -1 -vf fieldmatch,yadif,decimate,fps=1/3 -crf 4 -pix_fmt yuv420p -an -s 960x540 -g 1 -threads 0 3sec_intervals.mov [20:23:23 CET] <kepstin> yadif is a pretty slow filter [20:23:36 CET] <kepstin> but that's a bid difference, hmm. [20:28:00 CET] <alexpigment> is there any advantage to using "fieldmatch,yadif,decimate"? [20:29:17 CET] <kepstin> if you have bad cuts or fades or whatnot in telecined video, the yadif there will deinterlace the fields that couldn't be matched [20:29:31 CET] <alexpigment> ah [20:29:37 CET] <Mista_D> alexpigment: its to convert TC back to 24 fps [20:30:02 CET] <alexpigment> Mista_D: understandable, I just didn't understand the point of using yadif in that context [20:30:06 CET] <alexpigment> i'll trust kepstin on this [20:31:02 CET] <kepstin> normally you'd use it was "yadif=deinterlace=interlaced" so it only runs on frames marked as interlaced tho (iirc, fieldmatch sets that on frames where it had unmatched fields, or you could use idet) [20:31:27 CET] <alexpigment> yeah, that was what I was reading here on the FAQ: fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate [20:31:32 CET] <kepstin> er, "yadif=deint=interlaced" [20:31:53 CET] <alexpigment> in which case, maybe that would speed the operation up, if that is indeed th eproblem [20:32:05 CET] <kepstin> Mista_D: i'd suggest just moving the yadif filter to after the fps filter, and add the deint=interlaced option to it [20:32:18 CET] <kepstin> that'll make it run on way fewer frames, should speed it up a fair bit [20:32:32 CET] <alexpigment> very good point [20:32:33 CET] <Mista_D> OKTried removing fieldmatch and decimate, now it at x4.87 RT. seems good on output (will burn in timecode to make sure the frames are matching). [20:32:54 CET] <Mista_D> will try the yadif after too in a minute... [20:33:04 CET] <kepstin> if your content is already progressive (24p or whatever), there's no point in using fieldmatch/decimate [20:33:17 CET] <kepstin> but that looks like it's 60i telecined? [20:33:26 CET] <alexpigment> the filename seems to indicate it's telecined [20:33:29 CET] <kepstin> you'd really have to look at the video to check [20:33:31 CET] <alexpigment> well [20:33:36 CET] <alexpigment> "1080p_2997"... [20:33:37 CET] <alexpigment> not sure [20:35:33 CET] <alexpigment> honestly, he's doing fps=1/3 and 960x540... seems like you could just drop half of the fields rather than doing anything complicated [20:35:53 CET] <kepstin> ah, true, if you're downscaling that much no point in deinterlacing at all [20:36:03 CET] <kepstin> just use a filter that drops the other field [20:36:10 CET] <kepstin> be way faster than downscaling even [20:36:57 CET] <alexpigment> -vf field,fps=1/3 [20:37:14 CET] <kepstin> you'd still need to scale it horizontally [20:37:28 CET] <alexpigment> he's specifying the resolution explicitly [20:37:39 CET] <alexpigment> which should imply a scale filter [20:37:44 CET] <kepstin> oh, right, using -s [20:39:16 CET] <kepstin> Mista_D: give this a try, then: "ffmpeg -i 1080p_2997_tc_yuv422p.mxf -map_metadata -1 -vf field,fps=1/3 -crf 4 -pix_fmt yuv420p -an -s 960x540 -g 1 -threads 0 3sec_intervals.mov" [20:39:46 CET] <kepstin> that's with alexpigment's filters, should be the fastest way to do what you're trying to do. [20:39:50 CET] <Mista_D> kepstin: cool thanks! [21:00:31 CET] <nilshi> can I count from the end instead from the start in st=? I just want 0.5 seconds fade to black at the end. this is for batch processing: ffmpeg -i input.mts -vf 'fade=t=out:st=SEC:d=0.5' out.mkv [21:01:25 CET] <alexpigment> https://video.stackexchange.com/questions/19867/how-to-fade-in-out-a-video-… [21:01:54 CET] <alexpigment> a lot of that is about fading the audio [21:02:02 CET] <alexpigment> but i would assume the principle still applies [21:03:40 CET] <nilshi> just read that before asking. I was wondering if a workaround was still needed or if that feature was already in ffmpeg now [21:04:50 CET] <alexpigment> i'm not aware of a different way to do it [21:05:25 CET] <alexpigment> i suppose you could use a variable for the time and calculate it with an ffprobe call beforehand [21:06:50 CET] <nilshi> ok, will do that. Not the biggest problem in the world :) [21:06:58 CET] <alexpigment> also not the cleanest solution :) [21:07:12 CET] <alexpigment> seems like it should be an option for the filter [21:07:17 CET] <nilshi> I have two days of ugly hacking behind me. I don't mind about some awk now [21:07:55 CET] <alexpigment> fair enough :) [21:14:36 CET] <nilshi> why u no work with grep, ffprobe [21:15:36 CET] <nilshi> well, not needed here. https://trac.ffmpeg.org/wiki/FFprobeTips#Duration [21:34:00 CET] <kepstin> the only way the filter could do that is by buffering (delaying) frames for the duration of the fadeout. Doable, but kinda messy & might be memory intensive if it's a long fade. [22:17:25 CET] <nilshi> kepstin: it is just the convenience to let ffmpeg run ffprobe itself and get the duration [22:17:44 CET] <nilshi> on the other hand ffmpeg is not for endusers, so.. who cares [22:20:41 CET] <kepstin> and of course that's not possible at all on stuff like piped media or network streams [23:01:15 CET] <nilshi> when I apply the filter my video shrinks from 250MB to 4MB. That sounds like degredation [23:01:49 CET] <nilshi> ffmpeg -y -i "$FILENAME" -vf "fade=t=out:st=$STARTFADE:d=$FADE" "$FILENAME-faded.mkv" [23:02:17 CET] <durandal_1707> nilshi: pastebin full ffmpeg cmd output, increase cbr [23:06:37 CET] <nilshi> durandal_1707: https://pastebin.com/BT4HY1B4 [23:07:00 CET] <nilshi> I have not set cbr at all [23:08:19 CET] <durandal_1707> use similar preset [23:08:40 CET] <furq> nilshi: if the input is lossless then you need to set the same params for the output [23:08:59 CET] <furq> and also you need to specify libx264rgb or it'll convert to yuv [23:09:07 CET] <kingsley> pmjdebruijn: I'm back. [23:09:19 CET] <nilshi> -c:v libx264rgb -crf 0 -preset:v ultrafast -af aresample=async=1:first_pts=0 [23:09:27 CET] <nilshi> these are my recording options [23:09:32 CET] <kingsley> Thanks for your vorbis-y thoughts. [23:09:50 CET] <furq> set the first two [23:09:59 CET] <sfan5> or three [23:10:46 CET] <nilshi> let's see [23:11:03 CET] <kingsley> You reasonably asked if I checked if src/libavcodec/vorbisenc.c seg fault at line 103 is already fixed in svn. [23:11:06 CET] <kingsley> I have not. [23:11:33 CET] <durandal_1707> svn is not used long ago [23:12:01 CET] <kingsley> I was interested to see libvorbis is generally preferred to vorbisenc. [23:12:15 CET] <durandal_1707> use libvorbis [23:12:22 CET] <kingsley> If you happen to have the time, and are so inclined, I'd be interested in your further thoughts on... [23:12:33 CET] <nilshi> yes that worked. [23:12:52 CET] <nilshi> Now I can finally upload the video to youtube so that my nice quality shall be destroyed [23:12:54 CET] <kingsley> 1.) Where can I check if the vorbisenc.c's seg fault is fixed in svn, and [23:13:22 CET] <durandal_1707> kingsley: which project use svn? [23:14:45 CET] <kingsley> 2.) can you cite a link to a web page that I can show kdenlive's maintainer, so he might consider passing "acodec=libvorbis" to melt instead of "acodec=vorbis"? [23:15:35 CET] <furq> kingsley: https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio#AudioencodersFFmpegcan… [23:15:40 CET] <furq> Please note it is not recommended to use the experimental vorbis for Vorbis encoding; use libvorbis instead. [23:17:01 CET] <kingsley> durandal_1707: I don't know, but if I understand correctly, pmjdebruijn seeemd to think a few hours ago that src/libavcodec/vorbisenc.c is. [23:17:05 CET] <furq> with that said it's 2017 so he should really be using libopus [23:17:36 CET] <therage3> Agreed [23:17:39 CET] <therage3> Opus ftw [23:18:30 CET] <kingsley> furq: Thank you very much for the vorbis link. [23:18:44 CET] <kingsley> I hope to draw it to the attention of the proper maintainers. [23:19:12 CET] <nilshi> all this work for just a silly animation! https://youtu.be/CL37Cxb5Eh4 [23:19:23 CET] <kingsley> My testing suggests it explains why kdenlive isn't rendering .webm format video. [23:23:48 CET] <furq> https://github.com/KDE/kdenlive/blob/640d4467558db92b5d2dbcc0baaf74a4a5f664… [23:23:52 CET] <furq> that's a bit of an oversight yeah [23:24:05 CET] <furq> especially considering it was updated to use the new aac encoder for mp4 [23:24:55 CET] <kingsley> furq: I'm happy to report that much like "acodec=libvoris", "acodec=libopus" avoids the seg fault caused by acodec=voris". [23:25:01 CET] <kingsley> That's the good news. [23:25:04 CET] <kingsley> The bad news? [23:25:11 CET] <furq> if the segfault is in vorbisenc.c then that's no surprise [23:25:18 CET] <kingsley> The rendered video was silent! [23:25:25 CET] <furq> it wouldn't surprise me if that was a lavc bug because nobody uses the builtin vorbis encoder [23:25:29 CET] <furq> except kdenlive, apparently [23:26:04 CET] <durandal_1707> kingsley: which player you use? [23:26:24 CET] <kingsley> furq: Yes. The seg fault is in vorbisenc.c. [23:26:45 CET] <furq> you probably want to change quality=good and aq=%audioquality if you're using opus [23:26:54 CET] <kingsley> furq: gdb's full back trace is at [23:26:56 CET] <furq> idk what -quality good even does, i've never heard of that [23:26:57 CET] <kingsley> https://bugs.debian.org/cgi-bin/bugreport.cgi?att=1;bug=884268;filename=sym… [23:27:14 CET] <furq> but -aq does nothing with libopus afaik, you should use e.g. -ab 128k [23:30:22 CET] <kingsley> furq: Thank you very much for pointing your huge ffmpeg brain at my little seg fault. [23:30:41 CET] <kingsley> It seems to me that you've made not one, but two important contributions: [23:31:18 CET] <kingsley> 1.) a link to web page saying to use libvorbis instead of vorbis, which I can show to maintainers, and [23:31:33 CET] <kingsley> 2.) perhaps the exact line of kdenlive's code with the bug. [23:40:13 CET] <alexpigment> nilshi: looks nice! [23:40:19 CET] <alexpigment> oh nm, he left ;) [00:00:00 CET] --- Thu Jan 4 2018

1 0

ffmpeg-devel.log.20180102
by burek 03 Jan '18

03 Jan '18

[00:01:33 CET] <jamrial> how would you give them visibility? writing a blog post detailing their usage, like ubitux did for palletegen/use? [00:02:07 CET] <wm4> that would be a start [00:02:37 CET] <Chloe> do I need anything special to test rdt/rtp, or it just in FATE? [00:07:39 CET] <durandal_1707> wm4: which filters do same things? [00:09:18 CET] <durandal_1707> and afir is easy to use, you just need IR files [00:09:37 CET] <Chloe> durandal_1707: giving some cool examples of usages of all your filters might be a nice idea [00:10:07 CET] <Chloe> people look for things they want to do, they dont look for a filter which solves their problem [00:10:50 CET] <wm4> wtf is an IR file [00:14:20 CET] <durandal_1707> Impulse Response [00:14:58 CET] <durandal_1707> with FIR filtering one can do bunch of stuff [00:17:07 CET] <durandal_1707> equalization, phase change, reverberation, etc [00:32:53 CET] <Chloe> can someone with knowledge on how to test rtp/rdt test this: https://0x0.st/sKFi.patch (or tell me how I can do it myself) [00:32:56 CET] <Chloe> BBB maybe? [00:33:05 CET] <BBB> huh what? [00:33:24 CET] <BBB> I think wbs is a better person to ask [00:33:31 CET] <BBB> I havent touched rtp for ages :( [00:33:51 CET] <Chloe> sorry I thought you were Ronald S. Bultje [00:33:52 CET] <Chloe> my bad [00:36:02 CET] <Chloe> also there's probably a better way to do this than exposing the rdt handlers [00:36:04 CET] <jamrial> he is [00:36:37 CET] <Chloe> ah ok [01:32:12 CET] <BBB> Chloe: sorry, I just used to touch network stuff a lot more in the past than I do now... [01:32:29 CET] <BBB> Chloe: peoples interest shift, and this is one that sort of fell off my radar :-/ [01:32:48 CET] <Chloe> it's alright, I'd rather have not had to look at this either :D [03:00:08 CET] <Chloe> atomnuker: will resubmit lavc patch along with 2 extra lavf patches in 2hr or so (when they finish testing). Sorry it's going so slowly, my computer is struggling a lot [03:20:39 CET] <Chloe> wm4: so you said that we shouldnt allow registering external codecs, but what about formats? [03:22:35 CET] <JEEB> (32 [03:22:41 CET] <Chloe> fifo tests fail because I disallowed registering of external formats. I'm thinking about having another array for externally registered codecs, then iterating that after the internal ones [03:23:10 CET] <Chloe> This would be fairly simple to do with the opaque pointer in the _iterate functions, but not sure if this is the best way to do it [03:47:57 CET] <cone-178> ffmpeg 03Vishwanath Dixit 07master:26e1efb04f38: avformat/hlsenc: revamped master playlist url creation logic [03:47:57 CET] <cone-178> ffmpeg 03Vishwanath Dixit 07master:e872befdb597: avformat/hlsenc: configurable variant stream index position in filenames [03:47:57 CET] <cone-178> ffmpeg 03Vishwanath Dixit 07master:41e51fbcd9ed: avformat/hlsenc: creation of variant streams in subdirectories [04:06:32 CET] <atomnuker> Chloe: no external format reg either, or we'll get something worse than external plugin codecs [04:19:00 CET] <Chloe> atomnuker: guess the fifo test will have to be rewritten/disabled [04:19:10 CET] <Chloe> atomnuker: what do you think? [09:57:16 CET] <jpsollie> hello everyone, happy 2018! can somebody explain to me where the h264_vaapi profile->features mapping is done in the ffmpeg master branch? I encountered a problem whereby the stream uses unsupported features with the profile 100, but profile 77 and 66 are fine. I want to inspect the code where it is done so I can fix it. [09:58:34 CET] <BtbN> Somewhere in the driver. [09:58:51 CET] <jpsollie> but if ffmpeg doesn't do it, please tell me, so I can take a step forward to libva :) [09:58:54 CET] <jpsollie> ah [09:58:58 CET] <jpsollie> ok, sorry [09:59:10 CET] <jpsollie> I thought ffmpeg did it [09:59:26 CET] <jpsollie> so I have to search the amdgpu driver then? [09:59:34 CET] <BtbN> it just tells it to use main/high/baseline, and it does whatever it feels like [09:59:51 CET] <BtbN> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/vaapi_encode_h264… [10:00:59 CET] <jpsollie> all right, so it is the driver and not libva? [10:02:27 CET] <jpsollie> anyway, thank you for your help [10:26:14 CET] <fx159159> Hey, it was suggested to me to overwrite the AVPacket deallocation in a patch I submitted to the mailing list, how does one overwrite the deallocation correctly? Write my own av_packet_unref method? [12:22:13 CET] <wm4> Chloe: no formats either, except libavdevice [12:58:52 CET] <Chloe> wm4: so just remove the fifo test? seems to rely on this behaviour [13:04:12 CET] <wm4> what fifo test? [13:04:45 CET] <Chloe> wm4: libavformat/tests/fifo_muxer.c [13:05:26 CET] <wm4> first I'd ask why it even exists [13:07:28 CET] <wm4> that's pretty annoying [13:08:43 CET] <Chloe> it is annoying, since it seems like a pretty useful test [13:11:37 CET] <wm4> Chloe: can't you use avformat_open_input? [13:11:54 CET] <wm4> wait it's output [13:12:09 CET] <Chloe> wm4: the fifo muxer is opening the muxer [13:12:20 CET] <wm4> ok but avformat_alloc_output_context2 still can take a format directly [13:12:29 CET] <wm4> int avformat_alloc_output_context2(AVFormatContext **ctx, AVOutputFormat *oformat, [13:12:29 CET] <wm4> const char *format_name, const char *filename); [13:12:43 CET] <wm4> so pass &tst_failing_muxer for oformat [13:13:38 CET] <Chloe> The fifo muxer uses av_guess_format or whatever to find the "test" muxer internally [13:14:07 CET] <Chloe> So you'd need to add a way for the fifo muxer to take a AVOutputFormat directly [13:14:36 CET] <wm4> so what's the avformat_alloc_output_context2 call for? [13:15:12 CET] <Chloe> Isn't it just allocating the fifo muxer? [13:15:22 CET] <wm4> ok actually reading the code I start realizing the mess [13:15:35 CET] <Chloe> welcome to ffmpeg [13:16:48 CET] <wm4> and the test muxer seems to do a lot [13:18:48 CET] <wm4> does it ever prints its "summary"? [13:20:36 CET] <Chloe> https://0x0.st/sPz6.txt is the output if you just noop the _register function [13:21:17 CET] <wm4> yeah [13:21:36 CET] <Chloe> (not unexpected) [13:21:49 CET] <wm4> what I'm wondering, isn't there another way to test these failures (like with an existing muxer) [13:22:25 CET] <wm4> otherwise this test muxer could just be added to lavf itself or something [13:25:34 CET] <Chloe> adding it as ff_fifo_test_muxer would be the simplest solution maybe not the nicest though [13:26:30 CET] <wm4> to be honest I wonder why the fifo muxer even exists [13:26:49 CET] <wm4> but I guess it's here to stay [13:46:59 CET] <wm4> Chloe: I think you _could_ also try to make the format settable as pointer avoption in the fifo muxer... [13:47:21 CET] <Chloe> that seems way worse than just adding another muxer [13:48:19 CET] <wm4> hm I don't think it's that bad [14:37:06 CET] <Chloe> I think that's it for lavf, now just lavd and lavfi [14:48:19 CET] <KGB> [13FFV1] 15dericed opened pull request #82: Minor edits (06master...06minor-edits) 02https://git.io/vbpyl [15:03:50 CET] <Chloe> lavd done. Lavfi is gonna be a configure nightmare [15:08:10 CET] <atomnuker> is lavf done too? [15:08:20 CET] <atomnuker> with lavd support in lavf? [15:08:23 CET] <Chloe> yes [15:09:01 CET] <Chloe> well I just added another iter function for lavd, unless you think lavd stuff should be within the lavf iter functions [15:09:12 CET] <wm4> uh, it has to [15:09:36 CET] <nevcairiel> yeah it has to, lavd is basically just lavf [15:43:30 CET] <Chloe> atomnuker: https://github.com/eintw1ck/FFmpeg I've put what I've done so far here [15:43:42 CET] <Chloe> Got an issue with configure not configuring devices for some reason [15:44:54 CET] <atomnuker> what do you mean? its not listing any? [15:47:17 CET] <Chloe> the opposite, it always enables all of them [15:47:20 CET] <Chloe> i.e. http://sprunge.us/YIHQ [15:50:41 CET] <atomnuker> does print_enabled_components correctly skip disabled devs? [15:51:11 CET] <atomnuker> I suggest checking what's different between lavc and lavd [15:53:24 CET] <Chloe> atomnuker: I do the exact same 4 line change for codecs [15:53:56 CET] <Chloe> checking if I can even disable codecs [16:00:28 CET] <Chloe> seems like disabling codecs doesnt work either, but disabling BSFs works [16:02:40 CET] <Chloe> I cant tell you why though, there's nothing different between BSFs, codecs, formats or devices [16:03:34 CET] <atomnuker> put debuggint prints in that large loop which parses enable= and disable= [16:13:47 CET] <atomnuker> maybe you need to move where disable-outdev is on that loop [16:28:43 CET] <Chloe> ah I know why [16:29:03 CET] <Chloe> the new list has devices as _muxer and _demuxer [16:33:51 CET] <atomnuker> you can make a wrapper to map disable-indev in that loop if you can't find a way to fix that [16:34:21 CET] <Chloe> yeah so this is the issue with codecs too [16:34:39 CET] <Chloe> I was thinking to make find_things_extern take another argument as the output suffix [16:35:01 CET] <Chloe> then change the function which outputs the array to be able to change the suffix [16:35:17 CET] <Chloe> but mapping it in disable-*=* might be cleaner [16:35:25 CET] <Chloe> 'cleaner' lol. it's all a hack [16:35:54 CET] <wm4> even just rewriting the configure script in python would be a big win IMO [16:36:05 CET] <wm4> no more emulated stacks and stuff [16:36:19 CET] <Chloe> meson [16:36:23 CET] <atomnuker> crap [16:36:37 CET] <atomnuker> meson is unusable [16:36:41 CET] <Chloe> waf [16:36:57 CET] <Chloe> cmake [16:37:07 CET] <Chloe> surely there's something other than shellscript chaos [16:37:07 CET] <atomnuker> "meson: oh no you can't reconfugre, you've already done so, EVEN THOUGH IT ERRORED OUT, delete build dir" [16:37:48 CET] <wm4> atomnuker: that's the only reason? [16:38:19 CET] <atomnuker> apparently you can't depend on libraries NOT installed in /usr/local or /usr [16:38:30 CET] <atomnuker> even if you supply a pkg_config_path [16:39:16 CET] <atomnuker> waf is impractical if you want to distribute source code only since you'll either need internet connection to download the waf binary or carry the binary [16:39:44 CET] <Chloe> isnt it just a python bundle? [16:39:45 CET] <atomnuker> (python bytecode but still binary) [16:39:53 CET] <Chloe> which can be put in a tarball easily [16:40:06 CET] <Chloe> and if people are dl-ing from git then they have an internet connection anyway [16:40:06 CET] <BBB> /leave #build-tools /join #multimedia-channel [16:40:15 CET] <BBB> :-p [16:40:33 CET] <wm4> waf expects you to provide the binary blob with the source [16:40:57 CET] <wm4> (in mpv we just download it instead, which is not "proper") [16:41:22 CET] <Chloe> BBB: if the build system was sane then it wouldnt break everything with every minor change [16:41:36 CET] <BBB> are there sane build systems [16:41:46 CET] <BBB> some are featureful but bloated as crap [16:41:48 CET] <wm4> I keep looking for them [16:41:56 CET] <Chloe> I haven't found one yet [16:42:03 CET] <BBB> others are more shiny and sleek, but can barely zip up their own pants [16:42:22 CET] <BBB> I think our configure script was an attempt by diego/mans to find a middle road that works for us [16:42:27 CET] <BBB> and that really worked out well, didnt it [16:42:49 CET] <BBB> so my feel is that this is an unsolvable problem and I dont care anyway, so I ignore it and do more itnerestinh things ;) [17:01:01 CET] <atomnuker> well I'm glad no one suggested cmake [17:01:58 CET] <atomnuker> or as I like to call it -DWHOEVER_THOUGHT_OF_THIS_ATE_CRAYONS=ON [17:02:27 CET] <Chloe> 15:36:57 <Chloe> cmake [17:02:55 CET] <j-b> BBB: yes, meson [17:03:32 CET] <BBB> doesnt libaom use cmake? [17:03:32 CET] <Chloe> atomnuker: I just use this https://github.com/nemequ/configure-cmake/blob/master/configure [17:03:55 CET] <Chloe> it's a pretty interface for cmake, works nicely [17:05:15 CET] <atomnuker> neat, the configure system which takes longer to compile than gcc needs a shell script to be functional [17:06:22 CET] <Chloe> eh, not really. you can just use cmake, though I like to use the wrapper [17:06:33 CET] <atomnuker> all this talk about configure systems just makes me think our slow complicated configure script is the most functional [17:06:55 CET] <nevcairiel> it should just be written in an actual script language [17:07:00 CET] <nevcairiel> and it would probably be ok [17:07:01 CET] <sfan5> i'm sure it would be much faster if you rewrote it in Python [17:07:21 CET] <atomnuker> python is crap, have you seen waf? [17:07:34 CET] <sfan5> right [17:07:41 CET] <Chloe> what would you write it in then [17:07:44 CET] <sfan5> ^ [17:07:46 CET] <wm4> SHELL [17:07:50 CET] <nevcairiel> thats like saying C is crap because X is written in C [17:07:57 CET] <nevcairiel> how does the language relate to some project written in it [17:08:10 CET] <atomnuker> no, I'm saying python is woefully inadequate to make a build system [17:08:13 CET] <Chloe> There's always rust. [17:08:15 CET] <atomnuker> and to configure a build system [17:08:17 CET] <atomnuker> LUA! [17:08:18 CET] <nevcairiel> shell is far worse [17:08:27 CET] <atomnuker> we should use lua [17:08:38 CET] <sfan5> does lua even have a popen-equivalent? [17:08:53 CET] <Chloe> If you want to use lua then you may as well write an external and separate build system, then use that in ffmpeg [17:09:01 CET] <wm4> bare lua is... too bare [17:09:03 CET] <sfan5> nevcairiel: configure's task involves lots of interaction with shell-like programs and python is admittedly not good at that [17:09:14 CET] <atomnuker> io.popen [17:09:37 CET] <nevcairiel> what do you really need to make a configure replacement? string processing, the ability to invoke external commands and process their output, and some typical primitive data structures. shell fails at two of those [17:11:30 CET] <Chloe> durandal_1707: is there a filter to tile an image to a specified size? [17:11:59 CET] <durandal_1707> Chloe: tile? [17:12:46 CET] <atomnuker> what's fascinating about our configure script was that it was used in libvpx and libaom [17:13:09 CET] <Chloe> durandal_1707: display the image multiple times side by side to fit the specified size [17:13:16 CET] <atomnuker> libaom got cmake support half a year ago though but still everyone uses the configure script mostly [17:13:53 CET] <iive> i don't like cmake... [17:14:10 CET] <atomnuker> no one does [17:14:13 CET] <nevcairiel> cmake is ok for very simple things, but it can grow out of control [17:14:48 CET] <wm4> atomnuker: some do [17:15:57 CET] <iive> i don't like having to hunt what env var i have to change, for each different project... or not having consistent way to enable or disable things... [17:20:03 CET] <durandal_1707> Chloe: tile filter? or hstack and vstack with crop? [17:21:36 CET] <Chloe> latter looks like, thanks [17:40:44 CET] <chance> trying to understand the purpose of this initial padding [17:40:50 CET] <chance> https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/aacenc.c#L970 [17:41:16 CET] <chance> shouldn't this be configured based on pre_skip in codecpar? [17:41:39 CET] <nevcairiel> no, the encoder controls the initial padding [17:41:55 CET] <nevcairiel> the caller cannot be expected to know how much a certain codec might need [17:48:01 CET] <chance> @nevcairiel how will this work for segmented audio? [17:48:13 CET] <chance> it will inject 1024 samples into each segment? [17:48:33 CET] <Chloe> (you dont need the @ btw) [17:48:43 CET] <chance> uggg [17:48:49 CET] <chance> HipShat. Sorry [17:48:59 CET] <nevcairiel> no, the encoder is not restarted for each segment, it outputs a continous stream thats split apart later [17:51:55 CET] <jugnoo> Does anyone happen to know why inband FEC option is not supported by libopus wrapper in ffmpeg while the packet_loss option is supported? [17:51:58 CET] <chance> nevcairiel... why is this 1024 samples needed? [17:52:49 CET] <jugnoo> Looking into the code of encoder wrapper ~/libavcodec/libopusenc.c the line ret = opus_multistream_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss)); sets up the packet loss percentage in libopus but there is no reference to OPUS_SET_INBAND_FEC(...) libopus control in the wrapper. [17:53:25 CET] <jugnoo> Also looking into libopus the packet loss is used in conjunction with FEC settings i.e from opus/src/opus_encoder.c [17:53:52 CET] <jugnoo> <snip> /* When FEC is enabled and there's enough packet loss, use SILK */ if (st->silk_mode.useInBandFEC && st->silk_mode.packetLossPercentage > (128-voice_est)>>4) st->mode = MODE_SILK_ONLY; <snip> [17:54:44 CET] <jugnoo> setting one and not being able to set the other would seem to be moot or am I reading it wrong!!!!! [17:55:49 CET] <wm4> nevcairiel, wbs: do you think it'd be ok if I sent a patch that always calls WSAStartup on demand, and removes all calls to WSACleanup? [17:56:21 CET] <wm4> my goal is that the user doesn't need to call avformat_network_init() (it handles TLS, and this win32 thing) [17:57:09 CET] <Chloe> atomnuker: after I work out how to send emails from wsl, I'll send my updated set [17:57:18 CET] <Chloe> (after it finishes testing as well) [18:17:51 CET] <durandal_1707> atomnuker: no i can not reuse it [18:23:13 CET] <durandal_1707> its float, i want double [18:23:23 CET] <atomnuker> why do you want double? [18:24:09 CET] <durandal_1707> number of coefficients differ between nominator and denominator [18:24:38 CET] <durandal_1707> also my code is nicer [18:24:53 CET] <atomnuker> yeah, I'll give you that [18:25:03 CET] <durandal_1707> double is for all us audiophiles [18:25:23 CET] <atomnuker> double halves simd throughput [18:25:46 CET] <atomnuker> at least its not long double [18:26:02 CET] <durandal_1707> atomnuker: still very fast, also i will use long double [18:26:58 CET] <durandal_1707> more bits for golden ears [18:28:04 CET] <durandal_1707> atomnuker: move/copy fft from lavc to lavu and make it faster @/ [18:30:17 CET] <atomnuker> doesn't lavfi use av_fft? [18:30:22 CET] <atomnuker> which is in lavc [18:31:09 CET] <durandal_1707> yes, so move to lavu [18:32:30 CET] <atomnuker> yeah alright, will do it after atrac9 [18:38:05 CET] <atomnuker> might as well move both mdcts and ffts, I need the mdct for the rnnoise [19:41:28 CET] <durandal_1707> atomnuker: also high order iirs need more bits, thats why they usually cascade 2nd order to create similar effect [19:44:20 CET] <cone-554> ffmpeg 03Paul B Mahol 07master:92b32664cdc0: avcodec/utvideodec: add support for UMH2, UMY2, UMH4, UMY4, UMRA, UMRG [19:44:20 CET] <cone-554> ffmpeg 03Paul B Mahol 07master:57d0c24132c5: avcodec/utvideoenc: switch to planar RGB formats [20:11:43 CET] <durandal_1707> atomnuker: how much time of a day you spent on atrac9? [20:45:56 CET] <wbs> wm4: that's probably ok. but you can probably still call WSACleanup as well, at least for cases where you have a clear context for it [20:49:32 CET] <wm4> hm I can see if there's an obvious way to do it [20:51:08 CET] <nevcairiel> opening/closing tcp/udp protocols? [20:51:15 CET] <nevcairiel> when else is it even needed? [20:52:18 CET] <wbs> the rtsp (or was it sdp?) code does some amount of getaddrinfo/getnameinfo outside of the tcp/udp protocols, so at least within rtsp/sdp you'd need a separate enclosing init/cleanup for it [20:57:47 CET] <RiCON> stevenliu: can you select which representation you want in dashdec? [21:12:16 CET] <wm4> wbs: seems to call ff_network_init [21:12:52 CET] <wm4> most things do it indirectly via URL_PROTOCOL_FLAG_NETWORK [21:12:59 CET] <wm4> (then avio.c calls ff_network_init) [21:16:24 CET] <atomnuker> durandal_1707: 3, 4 days now [21:16:32 CET] <atomnuker> I can read all scalefactors! [21:17:02 CET] <atomnuker> still need to do that sbr thing, balance them according to the bit allocation and then read the spectral coeffs [21:17:40 CET] <atomnuker> I can't imagine how slow the encoder should be [21:18:00 CET] <atomnuker> if it does sbr correctly [21:18:13 CET] <durandal_1707> encoder is faster than realtime [21:18:38 CET] <atomnuker> since the packets need to be aligned to 8 bits to the packet size per second specified in the header [21:19:00 CET] <atomnuker> they probably do multiple encodes to figure out the correct amount of quantization [21:24:54 CET] <atomnuker> the whole codec's design was as if someone explained AAC-HE to someone over the phone but because they didn't need to keep backwards compat they were smart about it [21:26:13 CET] <durandal_1707> :) [21:26:47 CET] <Chloe> ah. I see the issue with separating lavd from lavf now [21:27:06 CET] <Chloe> all the code depends on devices being formats [21:27:48 CET] <durandal_1707> awful code [21:28:05 CET] <Chloe> yeah there's actually no way around this [21:32:12 CET] <wm4> Chloe: one suggestion was to have an avpriv function that registers the lavd list by setting a pointer to an array [21:32:35 CET] <wm4> this would complicate iteration somewhat, but can still be handled [22:02:39 CET] <durandal_170> im best dev you will ever have [22:03:00 CET] <wm4> ok [22:12:10 CET] <rcombs> I dunno, my dad's got a friend named dev and he's pretty cool [22:12:33 CET] <durandal_170> :) [00:00:00 CET] --- Wed Jan 3 2018

1 0

ffmpeg.log.20180102
by burek 03 Jan '18

03 Jan '18

[01:20:34 CET] <Johnjay> not that this matters [01:20:51 CET] <Johnjay> but i found youtube-dl seems better at downloading mp2t streams and encoding them than ffmpeg alone does [01:21:25 CET] <Johnjay> in this case I gave the .m3u8 file as the input and ffmpeg stopped after a minute or so, but youtube-dl seems to have finished [01:55:29 CET] <Johnjay> yeah spoke too soon, nevermind [01:55:39 CET] <Johnjay> youtube-dl just went longer is all [03:00:31 CET] <voldial> Can I specify the mkv Segment UID? ffmpeg -y -i 'file:test.mp4' -i 'file:test.webm' -c copy -map 0:v:0 -map 1:a:0 'file:test.mkv' [03:07:23 CET] <voldial> hm. MODE_WEBM https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/matroskaenc.c#L1923 [03:11:48 CET] <DHE> $ ffmpeg -h muxer=matroska # shows what options the MKV muxer accepts. yeah, nothing there. [03:31:13 CET] <voldial> ah. thanks [08:01:54 CET] <jpsollie> hello everyone, can anybody help me finding the part of ffmpeg where the h264_vaapi profile -> features mapping is done? [08:03:12 CET] <jpsollie> i'm looking for this part because my AMD Fiji and Tonga cards both output a stream which uses separate color planes, but it is unsupported [08:03:38 CET] <jpsollie> so I want to force it to disable this part [08:04:00 CET] <jpsollie> sorry, feature [09:03:13 CET] <ESphynx> please help guys... how do I just select another audio stream? [09:10:10 CET] <illegal> for what? anyway I think the option is -map # [09:18:09 CET] <ESphynx> right I was just struggling to understand that I nede to -map 0:0 for the video, then -map 0:4 for my audio, both options together... [09:18:23 CET] <ESphynx> though the man page explained it :P [10:15:49 CET] <kingsley> Could melt crash with a segfault if it passes acodec=vorbis instead of acodec=libvorbis to ffmpeg? [10:46:42 CET] <Spring> can ffmpeg copy streams from m2ts files for trimming? [10:47:14 CET] <Spring> basically wondering what would be best for creating short clips of rips for encoding test purposes [10:49:19 CET] <sfan5> it can yes [10:49:31 CET] <pmjdebruijn> Spring look int o-ss -t [10:49:54 CET] <pmjdebruijn> and -c:v copy -c:a copy [10:50:17 CET] <pmjdebruijn> although I do wonder whether you can use both [10:50:24 CET] <pmjdebruijn> both regard to iframes and such [10:50:27 CET] <pmjdebruijn> with* [11:32:02 CET] <kiranbsravi> Hi. I am trying to mux audio and video to a webm file using libavformat. For video stream, when i call av_interleaved_write_frame() with AVPacket, write_packet() -> mkv_write_flush_packet() in matroskaenc.c is called. [11:33:36 CET] <kiranbsravi> there MatroskaMuxContext.cur_audio_pkt is unrefed, and it is throwing "malloc: *** error for object 0x7fda5f816200: pointer being freed was not allocated" error. implying the AVBuffer's data was never allocated inside AVPacket. [11:34:37 CET] <kiranbsravi> i haven't freed any modules. Is there something i am missing ? [11:49:35 CET] <BtbN> Sounds like you never zero-initialized the struct, or did not allocate it properly [11:57:02 CET] <kiranbsravi> Which struct u mean ? [12:05:07 CET] <strfyr> i'm wondering, can i do something to improve the avcodec h264 decoder against packet loss? [12:27:59 CET] <DHE> while h264 does support a form of forward error correction for that sort of thing, I don't recall it being supported by ffmpeg [13:09:03 CET] <pmjdebruijn> DHE: the other issue is whether your client supports it [13:09:18 CET] <pmjdebruijn> DHE: also look into whether libx264 supports it [13:09:37 CET] <pmjdebruijn> https://www.ietf.org/rfc/rfc5109.txt it seems to be part of RTP as opposed to h264 [17:22:45 CET] <dongs> am i not supposed to be allowed to use -vf/crop filter when doing cuvid->nvenc encode? [17:22:58 CET] <dongs> does the decoded stuff never leave the GPU or something [17:26:16 CET] <dongs> ok looks like it [17:26:26 CET] <dongs> ill jus use dxva instead [17:28:58 CET] <Chloe> I'm trying to 'tile https://0x0.st/sPKc.png [17:29:00 CET] <Chloe> fk [17:29:15 CET] <dongs> gg [17:29:19 CET] <Chloe> hate this keyboard, enter key trash [17:29:28 CET] <dongs> you must be on a japanese keyboard [17:29:30 CET] <dongs> or a mac [17:31:26 CET] <Chloe> I'm trying to 'tile' an image (https://0x0.st/sPKc.png) there seems to be two ways to do it. The first using the tile filter and looping the image with `ffmpeg.exe -loop 1 -t 1 -i in.png -filter_complex tile=50x50 out.png`, but it seems to output a broken image https://0x0.st/sPKA.png [17:31:58 CET] <Chloe> the second is using hstack + vstack, though im not sure how to use it exactly (still need to read the docs for it) [17:33:08 CET] <durandal_1707> Chloe: thats just black for missing frames [17:33:39 CET] <durandal_1707> for single image you need to loop 50x50 frames [17:33:41 CET] <Chloe> I actually have no idea how to use ffmpeg, nice. Guess I need to increase loop time [18:32:36 CET] <Captain_Haddock> I'm looking to add a fade in/out options to text via drawtext. Examples of the net involve doing this by splitting the input file, adding the text to one, overlaying it, and manipulating its alpha channel. These examples are however from a few years ago. Is this still the case with the latest versions? [19:09:16 CET] <pgorley> hi, what's the difference between d3d11va and d3d11va2 in the configure script? [19:12:23 CET] <pgorley> ./configure --list-hwaccels shows both h264_d3d11va and h264_d3d11va2 [20:04:42 CET] <d4re> hey im having an issue with concat. the audio drops after a while and the logs are populated with [hls @ 0x25fe120] Delay between the first packet and last packet in the muxing queue is 10010000 > 10000000: forcing output [20:05:58 CET] <d4re> command: -loglevel debug -hide_banner -re -f concat -safe 0 -fflags +genpts -async 1 -stream_loop -1 -i /trailerparkboys.txt -y -framerate 24 -c:a libfdk_aac -b:a 128k -c:v copy -pix_fmt yuv420p -level 3.2 -maxrate 2M -bufsize 6M -crf 18 -r 24 -g 72 -f hls -segment_list_flags +live -hls_flags +discont_start -hls_time 10 -hls_wrap 20 -s 1280x720 /srv/stream.m3u8 [20:23:41 CET] <}8]> hey fellas. does anyone know if support for the mkv crop flag is on the horizon? [20:24:10 CET] <durandal_1707> whats mkv crop? [20:25:11 CET] <sfan5> probably PixelCrop{Top,Bottom,Left,Right} (https://www.matroska.org/technical/specs/index.html) [20:26:54 CET] <furq> }8]: if this is h264 then you can set the h264 crop values now, which should be better supported by players [20:27:04 CET] <durandal_1707> isnt that exported as side data? [20:27:21 CET] <furq> !bsf h264_metadata [20:27:21 CET] <nfobot> furq: http://ffmpeg.org/ffmpeg-bitstream-filters.html#h264_005fmetadata [20:29:29 CET] <}8]> furq - mkv crop settings dont require re-encoding the video. its very convenient & user-friendly to just set the crop values in the container [20:29:43 CET] <furq> you don't need to reencode to apply a bsf [20:30:08 CET] <furq> i'm not saying ffmpeg shouldn't support it but if it is h264 then it's better to change it in the sps [20:30:26 CET] <furq> all h264 decoders should support that whereas mkv crop values aren't always obeyed by players [20:31:07 CET] <}8]> theres always a player somewhere that doesnt work right but ive found most do [20:31:38 CET] <}8]> crippled players shouldnt set the bar imo [20:33:04 CET] <}8]> theres still tons of mpeg2 out there, and a lot of people remuxing from avi/mp4 into mkv now that mkv is widely supported and superior to those containers [20:35:44 CET] <alexp> }8] is this just so that kodi will show 2.39:1 rather than 16:9? [20:36:53 CET] <alexp> rather, will show it in the metadata properties? [20:41:47 CET] <}8]> i dont know about that but so any player using ffmpeg has proper video when mkv crop is used [20:42:34 CET] <sfan5> i'm fairly sure cropping would be responsibility of the API user, so support in just ffmpeg would mean nothing [20:42:50 CET] <alexpigment> }8] i figured i'd ask. I saw some thread about people wanting kodi to say a video was [whatever] aspect ratio in their library rather than 16:9. it seemed like a very superficial and weird request [20:43:20 CET] <}8]> easy way to get rid of unwanted borders, which in some cases arent even equal on all sides [20:43:24 CET] <alexpigment> anyway, unless i had a display wider than 16:9, the crop feature wouldn't really do much for me [20:44:27 CET] <alexpigment> }8] I've definitely seen that. I guess I'd be more concerned with non-integer scaling when cropped. my completely irrelevant 2 cents on the matter ;) [20:54:36 CET] <furq> }8]: oh right, you mean in the demuxer [20:54:49 CET] <furq> that makes more sense then i guess [21:00:48 CET] <}8]> yeah furq :) [00:00:00 CET] --- Wed Jan 3 2018

1 0

ffmpeg-devel.log.20180101
by burek 02 Jan '18

02 Jan '18

[01:21:55 CET] <kierank> Happy new year FFmpeg psychopaths [01:24:13 CET] <DHE> high functioning psychopaths, thanks [01:47:07 CET] <iive> i find that really offensive. [01:57:17 CET] <atomnuker> you have to be mental and a psychopath to a degree to do what we do [01:57:42 CET] <atomnuker> else we'd be writing decoders in python or whatever else is in style [01:58:38 CET] <iive> psyhopathy is very specific desices and is in no way releated to coding anything. [02:00:27 CET] <atomnuker> guess your dictionary's different then, words tend to get watered down the more they're used [02:01:22 CET] <iive> psychpaths are people who have no empaty and act mostly in self interest [02:01:42 CET] <iive> calling this people who work on open source project is insult. [02:07:32 CET] <atomnuker> whatever [02:08:29 CET] <tmm1> 2017 is over, lets agree that words have meanings [02:09:01 CET] <iive> we shuold have learning this back in 2016... [02:47:54 CET] <jamrial> iive: i thought that was sociopaths? [02:50:46 CET] <iive> jamrial, psychopaths are born like this, sociopaths are trained to behave like this. [18:06:18 CET] <cone-636> ffmpeg 03Carl Eugen Hoyos 07master:bddf31ba7570: configure: bump year [20:15:47 CET] <RiCON> atomnuker: getting this linking ffms2 with ffmpeg with disabled opus encoder/decoder since your changes, any idea what's causing it? https://i.fsbn.eu/4zl2.txt [20:20:12 CET] <atomnuker> can you check if adding #include <opus.h> right after ffmath.h helps? [20:20:44 CET] <atomnuker> (opus.h is already included by opus_celt.h which is already included by opus.c) [20:20:59 CET] <atomnuker> and opus.h includes opus_rc.h [20:33:07 CET] <RiCON> atomnuker: does opus_parser.c include the necessary headers? [20:33:24 CET] <RiCON> it's probably because of that (disabled dec/enc but not disabled parser) [20:35:00 CET] <atomnuker> none of the errors look like opus_parser.c issues [20:38:36 CET] <RiCON> opus_parser doesn't list opus_rc.o in the makefile [20:51:11 CET] <RiCON> atomnuker: yeah, this is enough to fix it: https://0x0.st/sKk6.txt [20:56:55 CET] <cone-636> ffmpeg 03Rostislav Pehlivanov 07master:fdbd26196702: lavc/Makefile: fix opus_parser dependencies [20:57:00 CET] <atomnuker> RiCON: fixed, thanks [21:09:40 CET] <kierank> michaelni: thanks [21:57:41 CET] <cone-636> ffmpeg 03Paul B Mahol 07master:9f7dbaad7e36: avfilter/af_crystalizer: use outlink instead of inlink [22:11:25 CET] <wm4> another part in the "fucking broken EOF change" saga: https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/tls_schannel.c#L419 [22:25:45 CET] <kierank> michaelni: so you don't have an issue with s->block2? [22:25:48 CET] <kierank> i dislike it a lot [22:27:48 CET] <cone-636> ffmpeg 03Carl Eugen Hoyos 07master:1112ba012df3: lavf/mov: Use av_fast_realloc() in mov_read_stts(). [22:30:39 CET] <cone-636> ffmpeg 03Carl Eugen Hoyos 07master:21b5990da461: lavu/mem: Do not realloc in av_fast_realloc() if size == min_size. [22:39:48 CET] <rcombs> wm4: what changed in EOF handling anyway [22:39:56 CET] <michaelni> kierank, i dont see a obvious better way. So no i dont see an issue. Maybe it could be renamed to something with 32 in the name to indicate the type [22:39:58 CET] <rcombs> tbh I've never really understood how EOF works in lavf [22:40:27 CET] <BtbN> rcombs, a lot of functions just returned 0 on EOF [22:40:37 CET] <BtbN> but reading 0 bytes is actually a thing, and AVERROR_EOF exists [22:40:56 CET] <BtbN> And the more stuff is fixed, the more other stuff breaks because of it... [22:41:43 CET] <wm4> 0 for EOF matches POSIX semantics too [22:41:44 CET] <rcombs> read()-style [22:42:03 CET] <rcombs> I guess it's kinda unavoidable for e.g. avio_r8 [22:42:04 CET] <wm4> but now returning 0 will strictly make the avio buffer code retry [22:42:28 CET] <wm4> (which means things will get stuck instead of ending) [22:42:54 CET] <wm4> and the assumption that 0==eof is apparently hardcoded in tons of code [22:43:17 CET] <rcombs> was it undocumented [22:43:53 CET] <wm4> I guess pretty much, which lead to nicolas george claim things [22:46:05 CET] <wbs> no it was documented; for external use, a wrapper allows the old behaviour still (with a warning about this changing), but for internal code no such wrapper should be used since all code was intended to be changed at once [22:46:31 CET] <rcombs> but there wasn't enough testing so cases weren't found, I guess? [22:47:05 CET] <wbs> probably not much outside of what fate covers [22:47:09 CET] <wm4> wbs: I thought we were talking about the situation before the change [22:47:24 CET] <rcombs> and fate covers little in the way of avio [22:47:25 CET] <wbs> wm4: before the change, I think it was documented that 0 meant EOF [22:47:55 CET] <rcombs> 0 for EOF seems particularly bad for nonblocking I/O [22:48:11 CET] <wm4> nobblocking IO returns EAGAIN [22:48:16 CET] <wm4> *non [22:48:21 CET] <rcombs> ah [23:52:40 CET] <durandal_1707> nobody uses afir :( [23:54:41 CET] <wm4> why would anyone [23:55:29 CET] <wm4> these filters have no visibility and are probably hard to use [23:55:58 CET] <wm4> and also too many filters that appear to roughly do similar things [00:00:00 CET] --- Tue Jan 2 2018

1 0

ffmpeg.log.20180101
by burek 02 Jan '18

02 Jan '18

[00:16:30 CET] <Tachyon> hny [00:16:41 CET] <kerio> HAPPY HALLOWEEN [04:53:42 CET] <Hotpot33> well, how do I uhm.. set the video and audio to different outputs [04:54:26 CET] <Hotpot33> like I want to extract the video into YUV files, and audio to pcm_s16le [04:54:41 CET] <Hotpot33> BOTH in one go, since it's live [05:02:50 CET] <DHE> ffmpeg -i ...... -map 0:v [options] video-out-file -map 0:a [options] audio-out-file [05:03:08 CET] <Hotpot33> ah ok [05:03:14 CET] <DHE> where "file" might be an RTMP target, literal file on disk, or whatever. usual rules [05:16:44 CET] <utack_> relaxed yes i used that now [05:16:47 CET] <utack_> thank you [05:18:14 CET] <johnnny22> does ffmpeg work with libva 2.0.0 ? [05:23:48 CET] <hiihiii> hi [05:24:10 CET] <hiihiii> I'm trying to upload media to twitter [05:24:25 CET] <hiihiii> my video is 45 seconds long [05:24:54 CET] <hiihiii> this is my command [05:26:10 CET] <hiihiii> ffmpeg -i in.flv -vsync 1 -c:v libx264 -preset slow -tune film -crf 15 -vf "fps=30,format=yuv420p" -c:a aac -ac 1 out.mp4 [05:26:52 CET] <hiihiii> but I keep getting "unable to upload" from twitter [05:27:29 CET] <hiihiii> everything seems like it's following their encoding settings [05:31:07 CET] <hiihiii> http://dpaste.com/228ETCP [05:35:56 CET] <utack_> Hi atomnuker do you happen to know about aptx or were you not really part of implementing that? [07:55:52 CET] <justswim> Hello all! Not sure if I've come to the right place, but I was wondering if anyone had any tips for adding colored emoji to a video with ffmpeg? [07:56:27 CET] <justswim> This question on stack overflow has been unanswered for a while and I was hoping that it might be possible to solve! [07:56:28 CET] <justswim> https://stackoverflow.com/questions/34880734/how-to-use-ffmpeg-with-drawtex… [07:59:51 CET] <justswim> This ticket has also been filed a while back without much progress: https://trac.ffmpeg.org/ticket/5777 [10:08:57 CET] <Fyr> guys, subtitles are a few seconds late. how to move them earlier when muxing with FFMPEG? [11:07:49 CET] <atomnuker> utack_: not really part of writing it, I just reviewed it [11:08:11 CET] <atomnuker> I don't think aptx can adjust the bitrate, its a very simple voice codec [15:35:35 CET] <utack_> atomnuker is it really fixed 352kbit/s bitrate? [15:36:28 CET] <utack_> that is ridiculous, using a codec with patents to make 352kbit/s stereo sound good [15:36:49 CET] <utack_> any moron could have cooked up a patent free codec to achieve that in half a day... [15:42:42 CET] <bencoh> :D [15:49:00 CET] <mrsip> Hi. I am messing around with a ring doorbell, and I want to try and take a picture every 60 seconds. I have done this using ffmpeg to generate a still image for other video streams, but does ffmpeg support sip urls? or do I need to rebuild with another lib (I am using linux) or does it not support it? [15:54:25 CET] <c_14> I don't think it does [15:59:44 CET] <SilenceDogood> What do I do if I run ffmpeg on an avisynth script and the result has the audio out of sync? [16:00:00 CET] <SilenceDogood> Any way I can get an exact number to use to delay the audio so it syncs back up? [16:04:51 CET] <SilenceDogood> oh wait, using audacity, I have determined that it is 0.342 seconds. I hope that's precise enough [16:08:22 CET] <SilenceDogood> my new command is ffmpeg -map 0:1 -itoffset 0.342 -i VTS_19_1.VOB -c:v libx264 -preset slow -crf 18 -c:a aac -b:a 192k -pix_fmt yuv420p output.mkv [16:08:35 CET] <SilenceDogood> Anyone please tell me if anything about that looks completely stupid and wrong. I'd appreciate it. [16:09:40 CET] <SilenceDogood> oops, typo. meant itsoffset [16:10:28 CET] <SilenceDogood> ah, apparently that is completely wrong. [16:17:01 CET] <SilenceDogood> new command ffmpeg -i danger.avs -itsoffset 0.342 -i danger.avs -map 0:0 -map 1:1 -c:v libx264 -preset slow -crf 18 -c:a aac -b:a 192k -pix_fmt yuv420p output.mkv [16:35:40 CET] <furq> SilenceDogood: itsoffset only affects video streams so you probably don't need multiple inputs [16:35:54 CET] <furq> assuming those are both supposed to be your avisynth script which seems weird if one is audio [16:38:05 CET] <SilenceDogood> furq o rly. I wasn't aware of this. So all I'd need would be one input and no map switches? [16:38:34 CET] <SilenceDogood> like this? ffmpeg -itsoffset 0.342 -i danger.avs -c:v libx264 -preset slow -crf 18 -c:a aac -b:a 192k -pix_fmt yuv420p output.mkv [16:38:48 CET] <furq> probably [16:38:49 CET] <SilenceDogood> this article here misled me then https://wjwoodrow.wordpress.com/2013/02/04/correcting-for-audiovideo-sync-i… [16:38:52 CET] <furq> i've not used avisynth in a while [16:39:00 CET] <furq> or itsoffset [16:39:25 CET] <atomnuker> utack_: you're forgetting its 20 or more years old and meant to be really fast to decode in hardware (of over 20 years ago) [16:39:34 CET] <furq> i would have thought you'd actually want -i foo.avs -i foo.vob -map ... [16:39:48 CET] <furq> but maybe your script actually processes the audio, idk [16:39:56 CET] <SilenceDogood> shouldn't actually hurt anything the way i did it initially though, should it? [16:40:17 CET] <SilenceDogood> I don't believe this script processes the audio at all [16:40:22 CET] <SilenceDogood> I mean, it just copies it AFAIK [16:40:29 CET] <furq> that works too [16:40:50 CET] <SilenceDogood> probably slower to encode though because of doing the script twice [16:40:58 CET] <furq> encode a sample to check if it works [16:41:28 CET] <SilenceDogood> the entire video's only 8 minutes so it's not such a big deal [16:42:03 CET] <SilenceDogood> but i may want to try a clip if the encode currently running (from that two input version) isn't successful [16:42:33 CET] <SilenceDogood> wow, ffmpeg crashed at the end of that [16:45:51 CET] <SilenceDogood> This encodes with silent audio: ffmpeg -itsoffset 0.342 -ss 30 -i danger.avs -c:v libx264 -preset slow -crf 18 -c:a aac -b:a 192k -pix_fmt yuv420p -t 15 danger.mkv [16:47:59 CET] <SilenceDogood> furq help? why's the audio silent [16:49:46 CET] <quax288> we have got more than 200% of the monthly donations today, thank you all so much!(weechat devs)crnojnqsqo: Cracki RonaldsMazitis D-ion [16:49:51 CET] <quax288> we have got more than 200% of the monthly donations today, thank you all so much!(weechat devs)djiiivpk: DasMoeh zcx dashcloud_ [16:49:56 CET] <quax288> we have got more than 200% of the monthly donations today, thank you all so much!(weechat devs)qevsbok: Yagiza aphirst zcx [16:50:01 CET] <quax288> we have got more than 200% of the monthly donations today, thank you all so much!(weechat devs)oludrvo: D-ion dashcloud_ dreamon__ [17:06:28 CET] <Llamageddon> Hello, I want to make a video file from an mp3 file(copying the stream without reencoding). Do you know if it's possible to use the cover image embedded in the file as the video's image? [17:06:49 CET] <durandal_1707> yes its possible [17:08:12 CET] <Llamageddon> How do I do it? [17:09:17 CET] <SilenceDogood> LOL furq apparently all your advice is wrong. [17:10:46 CET] <mickie> Hi All and a happy new year! [17:12:16 CET] <mickie> Can soneome explain what ffprobe works? I'm asking about the reporting of audio streams, in a .ts recording. [17:13:17 CET] <mickie> One recording shows aac_latm (LC) stereo, while the other which I started recording half way through the TV stream shows aac_latm (LC) 5.1 [17:13:32 CET] <mickie> They both should have 5.1 tracks on them. [17:20:00 CET] <mickie> So, if a stream starts as stereo and then switches to 5.1 audio tracks, how can I check to see if this is so? [17:23:24 CET] <Llamageddon> `ffmpeg -i "file.mp3" -c:a copy file.mp4` works but with issues: Seeking does not work in mplayer, and the image does not show up in VLC [17:25:24 CET] <mickie> OK, I used -ss to start half way through and they both show 5.1 [17:25:39 CET] <mickie> Bye all [17:33:41 CET] <utack_> atomnuker all the more reason to dump it instead of advertise and pay for it [17:36:19 CET] <atomnuker> welp [17:39:32 CET] <TD-Linux> well if it's more than 20 years old you don't need to pay for it anymore. but yes, it was pretty clear it only survived by marketing [17:39:35 CET] <Llamageddon> Okay so everything I try wrt/ converting mp3 to mp4, has a broken/missing index, making seeking impossible [17:39:48 CET] <Llamageddon> Including the examples from http://trac.ffmpeg.org/wiki/Encode/YouTube [17:39:51 CET] <TD-Linux> they made it really hard to abx test it, in fact the ffmpeg encoder is the first real chance to compare it to sbc. [17:50:03 CET] <SilenceDogood> I think instead of trying to use ffmpeg options to fix syncing issues, it might make more sense to just add or subtract silence manually in audacity [17:50:33 CET] <SilenceDogood> haha that seemed to fix it [17:51:06 CET] <SilenceDogood> so in the end, the command was just ffmpeg -i danger.avs -metadata:s:v:0 stereo_mode=1 danger.mkv [17:51:43 CET] <SilenceDogood> but fixed the audio source in the avs to point to an external WAV which was just the right amount out of sync with the video to counter-act ffmpeg's bizarre behavior [17:52:03 CET] <SilenceDogood> so that (hopefully) it comes back into sync on the otherend [17:53:46 CET] <EmleyMoor> Is there an easy way to have ffmpeg overlay a block of pixels at a fixed position onto a video? [17:54:07 CET] <SilenceDogood> EmleyMoor, it's ffmpeg. nothing's easy. but there might be a way [17:54:11 CET] <SilenceDogood> don't know it, sorry [17:57:55 CET] <utack_> EmleyMoor according to google "ffmpeg overlay image" yes [17:58:04 CET] <utack_> https://video.stackexchange.com/a/12111 [17:59:26 CET] <durandal_1707> learn to google [18:00:39 CET] <EmleyMoor> It can often be a case of finding an expression to Google that doesn't lead you down some crazy paths [18:01:46 CET] <utack_> TIL the default video encoder is x264, and i do not have to specify it... [18:19:56 CET] <EmleyMoor> Just doing some test runs for position now [18:25:38 CET] <SilenceDogood> How do I extract only the red channel of a video [18:32:22 CET] <furq> SilenceDogood: -vf extractplanes=r [18:32:34 CET] <furq> or more likely -vf format=rgb24,extractplanes=r [18:36:52 CET] <SilenceDogood> furq actually here's a better thing: could I reduce the video to greyscale in a way that only pays attention to red? [18:37:09 CET] <furq> that's what extractplanes does [18:41:06 CET] <SilenceDogood> furq OK thanks! [18:41:13 CET] <Captain_Haddock> Hi, first time user here. I'd like to programmatically generate videos that involve (looping) animated GIFs, captions, and audio from separate sources. Can I get this done with FFmpeg? [18:42:12 CET] <Captain_Haddock> i.e., the sources are animated GIFs; I've already been able to convert them to mp4s using FFmpeg. [18:43:39 CET] <Captain_Haddock> I'd like to crossfade different clips from the GIFs and perhaps also the audio. I'm trying to figure out what the limits, if any, are with FFmpeg. [18:54:36 CET] <Llamageddon> I'm trying to convert mp3 files(copying the stream) to mp4/mkv, and no matter what I do, seeking does not work for mp4 at all, and seems broken for mkv in vlc [18:56:56 CET] <Buster> doc/Makefile:51: recipe for target 'doc/fate.txt' failed [18:56:56 CET] <Buster> make: *** [doc/fate.txt] Error 1 [18:58:16 CET] <SilenceDogood> furq thanks for the advice on the red filter. That totally cleared up my problems with color on this video. Now the only problem left is working out how to make an exact audio sync [19:01:20 CET] <furq> SilenceDogood: ffmpeg isn't particularly great at processing dvds [19:01:29 CET] <furq> you might want to use another tool to demux the title set and then work on that instead [19:02:15 CET] <furq> ffmpeg will just discard any information about the title set that's part of the ifo, it just reads the vob as an mpeg-ps stream [19:04:17 CET] <therage3> that does explain why i was getting bad encodes with ffmpeg on this cartoon DVD [19:04:31 CET] <therage3> I had to make mkv's first and then parse those to ffmpeg [19:11:15 CET] <illegal> What could cause ffmpeg to be recognize by the cmdlet on windows when you use ffmpeg, but not when you add commands such as ffmpeg -i "file.mka" -a:c libopus "outfile.opus" [19:12:28 CET] <furq> it's -c:a [19:13:08 CET] <illegal> https://ghostbin.com/paste/ndvvn [19:13:27 CET] <illegal> furq, still got the error [19:13:36 CET] <furq> fffmpeg : The term 'fffmpeg' is not recognized as the name of a cmdlet, function, script file, or operable program. [19:13:56 CET] <furq> that might have something to do with it [19:13:59 CET] <illegal> lol [19:14:08 CET] <illegal> yeah [19:14:50 CET] <illegal> Thanks furq [19:15:11 CET] <illegal> I swear I never inspect the most obvious shit [19:25:17 CET] <SilenceDogood> I'm trying to play it by ear on syncing here which I totally hate [19:25:26 CET] <SilenceDogood> There's gotta be some way to fix this syncing issue properly [19:25:46 CET] <SilenceDogood> I can't get any more precise than 0.1 seconds by ear [19:26:09 CET] <SilenceDogood> But it might still feel wrong to people watching :( [19:29:10 CET] <therage3> you could try opening both the original stream and the new one in something like Audacity and then do the comparison [19:29:35 CET] <therage3> with Audacity you can go down to the sampling frequency and make it that precise if you want [19:33:10 CET] <furq> SilenceDogood: have you tried not sending the audio through avisynth [19:33:44 CET] <furq> there might be a start delay hardcoded in the vob that isn't making it through avs [19:33:53 CET] <furq> i forget whether that would be in the vob or ifo [19:34:17 CET] <furq> i'm also not entirely sure how ffmpeg deals with that if you have multiple inputs but there's only one way to find out [20:08:16 CET] <sjuxax> Hello. I am trying to transcode a video that contains an equirectangular side data stream. I can't seem to get this side data to come over. The situation is identical to https://stackoverflow.com/questions/44760588/preserving-side-data-informati…. [20:09:09 CET] <sjuxax> I tried the suggested -fflags +keepside but it hasn't made a difference, as reported on that question. [20:11:54 CET] <SilenceDogood> What the hell. my video came out in widescreen instead of 16:9 [20:13:57 CET] <SilenceDogood> look at this. Everything's right except it's been squished into widescreen https://www.youtube.com/watch?v=qQ4LPFywwOY [20:17:54 CET] <therage3> that's more than likely because of SAR and DAR stuff. i also had the same issue with a DVD cartoon I uploaded to youtube at, presumably, 720 x 480, but YouTube always uploaded at 640 x 480 [20:18:19 CET] <therage3> also, just a fyi, "widescreen" per se doesn't mean a specific aspect ratio, just anything greater than 1.37:1 [20:18:28 CET] <therage3> which 16:9 is [20:18:39 CET] <therage3> i understood what you meant, but remember that [20:18:41 CET] <furq> that video is just green here [20:20:28 CET] <SilenceDogood> furq what [20:20:43 CET] <SilenceDogood> it's a 3D video FYI [20:20:50 CET] <furq> yeah it's just solid green [20:20:57 CET] <SilenceDogood> furq well that's bizarre [20:20:59 CET] <furq> i changed it to 2d and it's still green, but a different shade [20:21:01 CET] <SilenceDogood> try refreshing the page [20:21:04 CET] <furq> i did [20:21:11 CET] <SilenceDogood> what browser [20:21:15 CET] <furq> firefox [20:21:23 CET] <SilenceDogood> Looks OK in Chrome [20:21:31 CET] <SilenceDogood> except for the aspect ratio issue [20:21:34 CET] <SilenceDogood> I think I figured it out [20:21:42 CET] <SilenceDogood> the original video is 720x480 [20:21:50 CET] <SilenceDogood> the 3D SBS is coming out as 1440x480 [20:22:15 CET] <SilenceDogood> To get the right aspect ratio without losing any quality, I need to tell this script to do 1440x960 [20:22:40 CET] <SilenceDogood> It's working on that now. hopefully that'll look better [20:23:46 CET] <therage3> SilenceDogood: is it supposed to be in grayscale? [20:23:49 CET] <therage3> in 2D mode I mean [20:23:57 CET] <SilenceDogood> therage3 yes. [20:24:17 CET] <therage3> SilenceDogood: then I see it as intended, not quite sure why furq is seeing green o.O [20:24:32 CET] <SilenceDogood> Might be your graphics card furq [20:25:24 CET] <kerio> SilenceDogood: are you sure you can't just set the PAR? [20:25:32 CET] <SilenceDogood> kerio what's PAR? [20:25:34 CET] <kerio> it seems awkward to have a vertically stretched video [20:25:36 CET] <kerio> pixel aspect ratio [20:25:50 CET] <SilenceDogood> kerio where's that set [20:27:38 CET] <kerio> or just setdar i guess [20:27:40 CET] <kerio> https://www.ffmpeg.org/ffmpeg-all.html#setdar_002c-setsar [20:29:20 CET] <SilenceDogood> kerio that's a filter isn't it? So all that does is stretch the video the same way doesn't it? [20:29:53 CET] <kerio> nope, that changes the aspect ratio without changing the video [20:30:40 CET] <SilenceDogood> So all I gotta say is "setdar=dar=16/9" ? [20:31:34 CET] <kerio> make that 16:9 i guess [20:32:16 CET] <kerio> if your video is h264 already you can use the h264_metadata bitstream filter to change the SAR [20:32:42 CET] <therage3> without reencoding? :/ [20:32:44 CET] <SilenceDogood> that was copy+pasted from the help page tho. it said use slash [20:33:19 CET] <SilenceDogood> My command to make that video on youtube was ffmpeg -i danger.avs -metadata:s:v:0 stereo_mode=1 danger.mkv [20:33:31 CET] <SilenceDogood> The avs gets the video straight from a VOB as a directshowsource [20:33:53 CET] <SilenceDogood> which came from a DVD [20:34:06 CET] <SilenceDogood> fortunately unencrypted [20:34:32 CET] <SilenceDogood> the audio portion is from a lossless WAV I made in audacity [20:34:35 CET] <kerio> ew mpeg4 [20:34:45 CET] <furq> mkv defaults to x264 [20:34:53 CET] <kerio> furq: dvd source [20:35:03 CET] <furq> where have you seen an mpeg4 dvd [20:35:10 CET] <kerio> mpeg2? D: [20:35:14 CET] <furq> there you go [20:35:20 CET] <kerio> ew mpeg2 [20:35:26 CET] <furq> ya got that right!! [20:35:50 CET] <kerio> SilenceDogood: https://ffmpeg.org/ffmpeg-bitstream-filters.html#mpeg2_005fmetadata [20:35:56 CET] <kerio> display_aspect_ratio [20:36:34 CET] <furq> don't modify the source [20:36:41 CET] <furq> if it's off a dvd then it can only be 4:3 or 16:9 anyway [20:37:01 CET] <kerio> i imagine it's some weird 3d dvd? [20:37:42 CET] <furq> afaik he's converted a red/blue anaglyph into stereo 3d [20:38:17 CET] <kerio> do you really need a dvd source for homestar runner tho [20:38:27 CET] <kerio> it exists in beautiful vector form [20:38:33 CET] <SilenceDogood> kerio, not Dangeresque Three [20:38:56 CET] <kerio> :o [20:39:00 CET] <SilenceDogood> It was created in-engine from the Strong Bad's Cool Game for Attractive People episode by Telltale [20:39:11 CET] <kerio> anyway i object to transcoding from a dvd [20:39:21 CET] <kerio> except at the point where youtube does i [20:39:22 CET] <kerio> t [20:39:30 CET] <SilenceDogood> so no flash is involved here: it's 3D [20:39:33 CET] <kerio> because you've done all you could [20:39:35 CET] <furq> fwiw the ar in the output video is probably the result of doing two 4:3 to 16:9 conversions [20:39:46 CET] <furq> it almost lines up with 1280 / 1.77 / 1.77 [20:40:01 CET] <furq> so presumably your avs script is needlessly doing that conversion [20:40:31 CET] <SilenceDogood> You want to look at the script? [20:40:37 CET] <furq> it couldn't hurt [20:40:39 CET] <kerio> whats avs [20:40:41 CET] <furq> i've not used avs in a long time though [20:40:42 CET] <furq> avisynth [20:40:44 CET] <SilenceDogood> Would this be the right command? ffmpeg -i danger.avs mpeg2_metadata=display_aspect_ratio=16/9 -metadata:s:v:0 stereo_mode=1 danger.mkv [20:41:10 CET] <SilenceDogood> The script is from this forum thread here https://www.youtube.com/redirect?v=qQ4LPFywwOY&event=video_description&redi… [20:41:11 CET] <kerio> needs a -bsf or something [20:41:16 CET] <SilenceDogood> oops sorry [20:41:18 CET] <kerio> also a -c:v copy [20:41:18 CET] <SilenceDogood> stupid redirect [20:41:29 CET] <SilenceDogood> http://www.dvinfo.net/forum/3d-stereoscopic-production-delivery/489934-how-… [20:41:35 CET] <SilenceDogood> kerio why [20:41:57 CET] <kerio> hold on i don't understand the avs part [20:42:07 CET] <furq> SilenceDogood: there's a 3d conversion filter in ffmpeg [20:42:12 CET] <furq> !filter stereo3d [20:42:12 CET] <nfobot> furq: http://ffmpeg.org/ffmpeg-filters.html#stereo3d [20:42:33 CET] <furq> actually nvm it doesn't support analygph as input, only as output [20:42:43 CET] <furq> anaglyph [20:42:53 CET] <SilenceDogood> furq yeah, already investigated that. completely useless for this. [20:43:12 CET] <SilenceDogood> the script is from this post here. you can click on the attachment to the post to DL it. http://www.dvinfo.net/forum/3d-stereoscopic-production-delivery/489934-how-… [20:43:43 CET] <kerio> there's no good way to do this losslessly :( [20:44:08 CET] <SilenceDogood> Yep [20:44:38 CET] <SilenceDogood> There are still some minor artifacts in the final product but it's pretty good overall [20:44:45 CET] <SilenceDogood> except for that aspect ratio problem [20:46:14 CET] <SilenceDogood> I get "mpeg2_metadata=display_aspect_ratio=16/9: Invalid argument" so that's evidently not the right way [20:47:11 CET] <kerio> SilenceDogood: i don't understand that avisynth thing but what does it output? [20:48:00 CET] <SilenceDogood> kerio, video? [20:48:07 CET] <kerio> what kind [20:48:45 CET] <SilenceDogood> kerio, don't know, that's why I linked the script [20:50:12 CET] <SilenceDogood> kerio, the video that comes out the other end inside the mkv that the command outputs is mpeg4 [20:51:17 CET] <SilenceDogood> wait no it comes out as h264. i was reading another thing my bad [20:54:24 CET] <SilenceDogood> OK I am trying new command, "ffmpeg -i danger.avs -vf setdar=dar=16/9 -metadata:s:v:0 stereo_mode=1 danger2.mkv" [20:54:32 CET] <SilenceDogood> let me know if that looks wrong [20:56:11 CET] <kerio> SilenceDogood: honestly just use -c:v copy and drop the filtering [20:56:18 CET] <SilenceDogood> But anyways, the point is that I'd never do this for a normal Homestar cartoon, but Dangeresque Three is the only one ever released in 3D, so this is a one-of-a kind case [20:56:25 CET] <SilenceDogood> kerio what does -c:v copy do? [20:56:29 CET] <kerio> if it's still wrong, fix the avisynth script [20:56:52 CET] <kerio> SilenceDogood: it doesn't do another transcode just to put the video in a mkv [20:56:58 CET] <kerio> or something, idk [20:58:18 CET] <SilenceDogood> kerio but it has to do that transcode because that's the format YouTube accepts [20:58:37 CET] <SilenceDogood> Goal is that it has to work with YouTube VR, which is fussy [20:59:09 CET] <SilenceDogood> quality was fine on that video I uploaded earlier, just wrong aspect ratio [21:00:44 CET] <Llamageddon> Can anyone help me? I'm trying to convert mp3 files with embedded covers into mp4 but no matter what I do seeking is broken and the video is half-broken [21:02:20 CET] <sfan5> Llamageddon: which commands have you tried so far? [21:03:01 CET] <Llamageddon> I tried almost every single switch I could find, every example and a few different combinations [21:03:12 CET] <Llamageddon> This has almost everything I tried: ffmpeg -i "file.mp3" -crf 21 -bf 2 -g 30 -flags +cgop -movflags faststart -framerate 2 -preset medium -c:a copy -r 1 -tune stillimage -c:v libx264 -pix_fmt yuv420p -b:v 1000k -coder 1 -loop 1 "file.mp4" [21:03:41 CET] <Llamageddon> I tried at least 8 different commands posted online and some other switches besides [21:06:51 CET] <furq> Llamageddon: ffmpeg -i file.mp3 -vf loop=-1 -tune stillimage out.mp4 [21:08:30 CET] <Llamageddon> furq: Trying that with added -c:a copy results in the exact same problem [21:08:35 CET] <sfan5> can confirm [21:08:46 CET] <Llamageddon> Is seeking somehow impossible with mp3 audio? [21:08:55 CET] <Llamageddon> Should I just give up and convert it to aac? [21:09:26 CET] <sfan5> that doesn't seem to help either [21:09:36 CET] <sfan5> is there a reason why you can't just use a m4a + cover? [21:09:50 CET] <Llamageddon> sfan5: What is m4a? [21:10:14 CET] <sfan5> the exact same format as mp4 but only supposed to store audio dat [21:10:16 CET] <sfan5> data* [21:10:25 CET] <Llamageddon> I want to convert my file for uploading to youtube while preserving the original audio steam since YouTube is going to convert it anyway afaik [21:10:46 CET] <sfan5> ah, yea so you need an actual video as output [21:10:51 CET] <Llamageddon> Yes [21:11:12 CET] <SilenceDogood> Specifying setdar=dar=16/9 did not change the default aspect ratio in the skybox player [21:11:26 CET] <SilenceDogood> I guess I could try it on YouTube but it probably won't do anything [21:13:19 CET] <sjuxax> I've uploaded a folder of sample output from a RICOH Theta V with ffprobe outs. I have also included my attempt to use ffmpeg to copy the streams whilst retaining side data (equirectangular info). See https://drive.google.com/drive/folders/1mbSFgFpDQrU52eFYQpUZttrk8jdcq9RB [21:13:40 CET] <sjuxax> Ongoing discussion at the SO question linked above, which was https://stackoverflow.com/questions/44760588/preserving-side-data-informati… [21:14:10 CET] <sjuxax> If SO's "LordNeckbeard" is present here, /me waves; we can chat on IRC if it's more convenient/useful, but I'll be in and out [21:14:58 CET] <sfan5> Llamageddon: ffmpeg -i file.mp3 -f lavfi -i color=s=640x640 -shortest -filter_complex 'movie=file.mp3 [x]; [1:v][x] overlay [outv]' -map '[outv]' -map '0:a' -c:a copy -tune stillimage -y file.mp4 [21:15:10 CET] <sfan5> you will have to adjust the resolution at the beginning to match the cover art [21:17:32 CET] <Llamageddon> sfan5: That seems like a pretty complicated line, where did you get it, and how does it work? And why did what I was trying not work? [21:18:07 CET] <SilenceDogood> Nope, YouTube accepted that just fine [21:18:10 CET] <sfan5> I made it up; it uses an empty (black) 640x640 video and then overlays the cover art on it; I have no idea [21:18:13 CET] <SilenceDogood> I think I'm done!! :D [21:18:26 CET] <furq> i just checked and the command i posted is almost right [21:18:42 CET] <furq> for some reason -vf loop=-1,fps=25 is broken but -vf loop=-1 -r 25 works [21:18:46 CET] <furq> even though they should be identical [21:18:49 CET] <furq> anyway [21:19:08 CET] <furq> ffmpeg -i file.mp3 -vf loop=-1 -r 25 -tune stillimage -shortest -pix_fmt yuv420p out.mp4 [21:19:26 CET] <furq> probably change -r 25 to -r 6 if this is for youtube, although it won't make a huge difference [21:19:40 CET] <sfan5> shouldn't -r 1 work aswell? [21:19:43 CET] <furq> sure [21:20:02 CET] <furq> although actually potentially no [21:20:15 CET] <furq> there used to be an issue where doing that with -c:a copy would cause the duration to get fucked p [21:20:22 CET] <furq> the lower the output fps, the worse it was [21:20:36 CET] <furq> that was actually with image2 and -loop, i forget if it happened with the filter [21:20:38 CET] <Llamageddon> furq: The output of the command you posted now does not have any video when played in mplayer2 (although I added -c:a copy) [21:20:57 CET] <Llamageddon> Only the thing sfan5 works, and perfectly so too [21:21:08 CET] <furq> lol [21:21:12 CET] <furq> well that's cool [21:21:13 CET] <sfan5> it's a huge workaround but as long as it works ¯\_(Ä)_/¯ [21:21:28 CET] <furq> it does the same thing here with copy [21:21:29 CET] <Llamageddon> It takes non-negligible time to encode the video, but it works, at least [21:21:32 CET] <furq> that's definitely fucking broken [21:21:45 CET] <furq> the command didn't work at all in 3.3, i had to upgrade [21:23:46 CET] <furq> [mp4 @ 0000000002bea940] track 1: codec frame size is not set [21:23:50 CET] <furq> i get that when copying and not otherwise [21:24:02 CET] <furq> so yeah that's surely broken [21:24:37 CET] <Llamageddon> furq: Well, so at the very least a bug was discovered thanks to my floundering about [21:24:43 CET] <Llamageddon> Feel free to report it, I don't really know where [21:24:57 CET] <furq> actually the output file has the stream order reversed so that must be the audio stream [21:25:02 CET] <furq> so that makes sense [21:25:11 CET] <furq> this is still broken though [21:26:02 CET] <furq> wtf [21:26:09 CET] <furq> i get an x264 encoding log but the stream isn't there in the output [21:26:54 CET] <Llamageddon> furq: What do you use to check the composition of the final file? [21:26:58 CET] <furq> ffprobe [21:27:43 CET] <Llamageddon> Okay thanks ^^ [21:27:52 CET] <furq> i must be doing something stupid here [21:28:19 CET] <Llamageddon> Do you know if there's a way to not have to manually specify the cover image's resolution like in sfan5's command? I'd like to make a generic bash function for doing this [21:29:07 CET] <furq> well what i would have suggested is extracting the cover art [21:29:22 CET] <furq> then you can just do ffmpeg -i foo.mp3 -loop -1 -i foo.jpg out.mp4 [21:29:28 CET] <furq> that all works as expected [21:29:47 CET] <Llamageddon> Ahh [21:29:55 CET] <furq> -loop only works with image inputs [21:30:12 CET] <furq> cover art is treated as a muxed video stream [21:30:21 CET] <furq> which might explain why this is being such a pain in the arse [21:31:25 CET] <Llamageddon> furq: That is also broken with -c:a copy [21:31:33 CET] <Llamageddon> >_> [21:31:49 CET] <furq> what's wrong this time [21:31:54 CET] <furq> oh [21:32:01 CET] <furq> add -shortest? [21:32:08 CET] <furq> otherwise it'll run forever [21:32:19 CET] <furq> seeking should definitely work with that though [21:32:34 CET] <sfan5> inb4 it doesn't [21:32:42 CET] <Llamageddon> Adding -shortest just makes the file 0.01s or so long [21:32:52 CET] <furq> oh ugh [21:32:53 CET] <furq> -loop 1 [21:32:59 CET] <furq> typo [21:33:23 CET] <Llamageddon> I tried that too, same thing [21:33:37 CET] <furq> paste the command [21:33:46 CET] <Llamageddon> Dunno what but something is completely fucked when it comes to -c:a I'd say [21:33:53 CET] <Llamageddon> ffmpeg -i in.mp3 -i out.jpg -c:a copy -loop 1 -shortest out3.mp4 [21:34:01 CET] <furq> -loop 1 goes before -i out.jpg [21:34:15 CET] <Llamageddon> Wait that matters? [21:34:17 CET] <furq> yes [21:34:19 CET] <Llamageddon> Shit x.x [21:34:49 CET] <furq> that's why you need to extract the image [21:34:56 CET] <furq> that option is specifically an input option for image files [21:35:03 CET] <Llamageddon> I wish any of the resources I found online said that the order fucking matters [21:35:04 CET] <Llamageddon> X_X [21:35:34 CET] <furq> it's a bit late to say this but 99% of online resources for ffmpeg are outdated or wrong or just stupid [21:35:40 CET] <furq> other than ffmpeg.org [21:35:43 CET] <Llamageddon> Yeah I noticed that [21:35:53 CET] <furq> it's better to just ask in here [21:35:59 CET] <Llamageddon> Okay the thing works [21:36:22 CET] <furq> i still can't get the loop filter to work so that's my best suggestion thus far [21:36:26 CET] <Llamageddon> Could I also reduce the framerate/bitrate/something since the video is a static image? [21:36:32 CET] <furq> add -tune stillimage again [21:36:44 CET] <furq> and you can adjust the framerate (-r 6 or whatever) [21:36:47 CET] <furq> those won't make a huge difference though [21:36:50 CET] <Llamageddon> Anywhere in the line? [21:36:57 CET] <furq> as an output option [21:37:12 CET] <furq> [input 1 options] -i input1 [input 2 options] -i input2 [output optionsp output [21:37:14 CET] <furq> [input 1 options] -i input1 [input 2 options] -i input2 [output options] output [21:38:41 CET] <Llamageddon> Okay huge thanks, it all works perfect now, and stillimage and -r 6 did speed the encoding up considerably [21:38:51 CET] <Llamageddon> Not like it takes a long time, but since it makes no difference anyway [21:38:52 CET] <furq> you can add -preset superfast as well if you just want to speed up encoding [21:38:58 CET] <furq> (output option) [21:39:10 CET] <furq> it's a still image so the video stream will be tiny anyway [21:39:37 CET] <furq> also if this is for youtube you probably want to scale the image to 720p so that you get the highest quality audio [21:39:43 CET] <furq> if it isn't already [21:40:01 CET] <Llamageddon> That'll stretch the cover though, hm [21:40:12 CET] <furq> you can keep the ar, just as long as the height is 720 [21:40:20 CET] <furq> -vf scale=-2:720 as an output option [21:40:28 CET] <Llamageddon> What does the -2 stand for? [21:40:51 CET] <furq> -1 scales to the correct value for the AR and height, -2 does the same thing but ensures it's an even number [21:41:45 CET] <furq> your output is yuv420p which means the width and height need to be even numbers [21:42:03 CET] <Llamageddon> I see I see. Really huge thanks man. [21:42:20 CET] <Llamageddon> I've gotta note this down for future reference if I ever end up wanting to transcode videos [22:11:00 CET] <Llamageddon> furq: One more question: What switch and where should I use if i want a less verbose input(e.g. only 1-4 lines)? [22:13:44 CET] <mike_check> -hide_banner [22:15:08 CET] <metalhead33> Hello everyone [22:16:38 CET] <metalhead33> Not exactly sure about the newest FFmpeg, but as a user of FFmpeg 3.3, I looked into the source code, and found out that there can not be muxing to xwm, only demuxing fromk it. [22:17:43 CET] <Llamageddon> mike_check: Hmm maybe I'm adding it in the wrong spot but it doesn't seem to reduce the amount of stdout ffmpeg outputs substantially [22:21:05 CET] <mike_check> Llamageddon: can you post the command/output in pastbin? [22:21:21 CET] <Llamageddon> mike_check: I could but anyway, I found what I needed, which is `-loglevel panic` [22:22:59 CET] <furq> i'm still here trying to get this to work without extracting the image [22:23:03 CET] <furq> i don't think it's possible any more [22:25:32 CET] <Llamageddon> The workaround with overlaying the image posted by the other guy worked, dunno if that counts [22:26:01 CET] <SilenceDogood> Thanks for your help furq. Final product is here: https://youtu.be/1LSadYPuB3A [22:26:03 CET] <furq> it does not [22:26:48 CET] <SilenceDogood> OK I gotta go, cya! :D [22:26:59 CET] <furq> this is really annoying [22:27:04 CET] <furq> the duration is two seconds out and i have no idea why [22:30:08 CET] <furq> oh wtf [22:30:27 CET] <furq> it actually works with -c:a copy even though ffmpeg and x264 both report the wrong number of frames written [22:35:00 CET] <furq> http://vpaste.net/dwIOQ [22:35:08 CET] <furq> i have no idea what's going on there [22:35:12 CET] <furq> and also i think i'm done with this shit now [22:35:33 CET] <Llamageddon> furq: Report the bug perhaps, if ffmpeg has a bugtracker/mailing list? [22:35:39 CET] <furq> there is [22:35:44 CET] <furq> idk if this is a bug or if i'm using it wrong [22:35:50 CET] <furq> it's normally the latter [22:47:58 CET] <tM^caoX> damnit, bashing my head around using ffmpeg with jpeg's, trying to preserve original exif metadata, is this truly impossible, to date? say simple 'ffmpeg -i img_0001.jpg output.jpg" where i want the exif to be copied as-is into the output..? [22:48:27 CET] <furq> does -map_metadata 0 work [22:48:50 CET] <tM^caoX> no, tried it. also tried "-map_metadata -1" ... :/ [22:49:28 CET] <tM^caoX> right now i'm using exiftool (the -tagsFromFile switch) causing an extra step in my process, would be awesome finding a way to eliminate the need for this. [22:52:17 CET] <Llamageddon> furq: I ended up using `ffmpeg -i "$FILE_IN" -loop 1 -i "$FILE_IMG" -c:a copy -c:v libx264 -shortest -tune stillimage -r 6 -preset superfast -vf scale=-2:720 "$FILE_OUT"`, which works perfectly, although on my weak rig, inside a Linux VM, it takes almost 90s for a 4m50s long song, do you have any quick ideas about speeding it up? [22:52:47 CET] <sfan5> you can pick faster presets but I'm not sure if that's a good idea [22:52:49 CET] <sfan5> also less fps [22:54:01 CET] <Llamageddon> sfan5: I tried -r 1 but that breaks seeking... which is probably not important because YT likely converts it anyway, but ah well. How do I try faster presets, and why is it unlikely to be a good idea? [22:54:35 CET] <sfan5> -preset fast / -preset faster / -preset ultrafast [22:54:43 CET] <sfan5> because it can reduce quality [22:54:50 CET] <furq> he's already using superfast [22:55:07 CET] <furq> you can try ultrafast [22:55:11 CET] <Llamageddon> I'm copying the audio stream and the video is just a single image, I don't think quality matters :P [22:55:51 CET] <Llamageddon> furq: Is that an output option, or should I add it to the video stream part? [22:57:07 CET] <metalhead33> Okay, now I know why muxing to xwm is not possible [22:57:40 CET] <furq> Llamageddon: change superfast to ultrafast [22:58:49 CET] <Llamageddon> Hmm, don't think it makes a difference with what I'm doing. Still, thanks again for all the help! [23:29:43 CET] <FishPencil> On a bluray disk are only the m2ts files encrypted and not the other files such as the mpls playlists or the java menus? [23:42:55 CET] <foul_owl> Having an issue trying to cut videos. The start time is off by several seconds [23:43:23 CET] <foul_owl> What is the most accurate way to cut a video? ie, the closest I can get to my timestamps? [23:45:27 CET] <klaxa> if you are using -c copy you are limited to cutting at i-frames at the beginning (and under some circumstances at the end as well afaik) [23:45:30 CET] <kepstin> foul_owl: if you're copying the video (-c:v copy), the closest you can get is the nearest keyframe, so it depends on how the original video was encoded. [23:45:41 CET] <kepstin> foul_owl: if you re-encode the video, ffmpeg's seeking is frame-accurate [23:46:46 CET] <foul_owl> How often do keyframes happen? Like every few seconds or so? [23:48:42 CET] <klaxa> depends on the codec and the encoding settings [23:48:52 CET] <foul_owl> Gotcha [23:48:59 CET] <foul_owl> Ok so sounds like reencoding is my best option here [23:49:23 CET] <foul_owl> What is the best way to reencode that doesn't cause any quality loss from the original? [23:49:29 CET] <foul_owl> Or at least minimizes it [23:50:24 CET] <kepstin> depends what you plan to do with the video afterwards, really [23:50:53 CET] <foul_owl> I basically just want to keep the bitrate / codec / quality of the original clip [23:50:59 CET] <foul_owl> But cut it to frame accuracy [23:51:19 CET] <foul_owl> Is there a flag to tell ffmpeg "keep the original codec but reencode because I need frame accurate cuts" ? [23:51:22 CET] <klaxa> but do you want to keep editing it or do you want to share it directly? [23:51:30 CET] <foul_owl> I don't need to edit it further [23:51:30 CET] <klaxa> there is no such flag [23:51:34 CET] <foul_owl> I just need to cut [23:52:07 CET] <klaxa> your best bet is probably just using -c:v libx264 and experiment with -crf values [23:52:12 CET] <klaxa> afaik the default is 23 [23:52:16 CET] <klaxa> lower means higher quality [23:52:22 CET] <foul_owl> Gotcha! [23:52:36 CET] <foul_owl> That seems like a pretty good standard, I just have no idea :) [23:52:50 CET] <klaxa> so: ffmpeg -i input.mp4 -ss 01:23:45.67 -c:v libx264 -crf 23 -t 10 output.mp4 [23:52:52 CET] <kepstin> hmm. there are some tools out there iirc which can cut h264 videos by only re-encoding the first (&last?) gops to cut it frame accurate, but copy the middle bit [23:53:00 CET] Action: kepstin can't remember offhand what could do that [23:53:20 CET] <foul_owl> -ss after -i ? [23:53:47 CET] <klaxa> since a few years (?) it doesn't matter [23:54:14 CET] <klaxa> https://trac.ffmpeg.org/wiki/Seeking [23:54:34 CET] <foul_owl> Hmm, it seems to make a large difference in the amount of time it takes to encode on my system [23:54:41 CET] <klaxa> > As of FFmpeg 2.1, combined seeking is still possible but I have yet to find a valid use case for it since -ss as an input option is now both fast and accurate. [23:54:43 CET] <kepstin> if you put -ss after -i it'll be a lot slower, yes [23:54:51 CET] <foul_owl> But I'm on Debian also, so the packages are probably old as hell [23:54:56 CET] <kepstin> since in that case it has to decode then throw out the entire video [23:55:13 CET] <klaxa> oh, i'm a dumbo, nevermind me [23:55:46 CET] <foul_owl> Thanks everyone! [00:00:00 CET] --- Tue Jan 2 2018

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