June 2018 - Ffmpeg-devel-irc - lists.ffmpeg.org

ffmpeg-devel.log.20180629
by burek 30 Jun '18

30 Jun '18

[01:28:05 CEST] <cone-914> ffmpeg 03Jerome Borsboom 07master:1cf6129d13a3: avcodec/vc1: fix condition guarding overlap filter on I picture [01:28:05 CEST] <cone-914> ffmpeg 03Jerome Borsboom 07master:f92e95e9b5af: avcodec/vc1: change the internal ordering of blocks within a macroblock [01:28:06 CEST] <cone-914> ffmpeg 03Jerome Borsboom 07master:975a1a81b28d: avcodec/vc1: fix overlap filter for frame interlaced pictures [01:28:08 CEST] <cone-914> ffmpeg 03Jerome Borsboom 07master:efde2a70ffe6: avcodec/vc1: fix check for missing CBPTAB [01:35:21 CEST] <cone-914> ffmpeg 03Carl Eugen Hoyos 07master:27662ed681cb: Changelog: Mention vc1 improvements by Jerome Borsboom. [10:41:30 CEST] <kierank> gagandeep: ask durandal_1707 to explain threading [10:42:24 CEST] <gagandeep> should i read pngdec.c also [10:46:02 CEST] <atomnuker> slice threading is trivial [13:01:24 CEST] <atomnuker> is there something like av_clip_uintp2 which zeroes out <bits> MSBs instead of clipping? [13:01:55 CEST] <nevcairiel> you mean like a binary and? [13:02:00 CEST] <nevcairiel> why do you need a function for that? [13:06:02 CEST] <atomnuker> atrac9 weirdness, the RE'd decoder calls this SignExtend32 (which just does (val << bits) >> bits, where bits = 32 - zero_bits) [13:06:42 CEST] <jdarnley> I think I've seen a sign extend macro/func somewhere [13:07:00 CEST] <jdarnley> But that adds 1s [13:10:38 CEST] <atomnuker> its very weird, for some reason the scalefactor delta from get_vlc2() has to be truncated to the maximum length of a code of the huffman table [13:11:11 CEST] <atomnuker> which doesn't make sense because get_vlc2() shouldn't output more than the number of bits of the maximum length of a huffman symbol of the table [13:30:39 CEST] <atomnuker> oh, its because the top bit is the sign bit, and its included as the first bit in the huffman tables [14:49:00 CEST] <durandal_1707> atomnuker: so you are going to complete atrac9? [15:24:26 CEST] <atomnuker> durandal_1707: yeah, I'm on it [18:14:16 CEST] <atomnuker> atrac9 wtf [18:14:42 CEST] <atomnuker> each coefficient isn't sent as a huffman coded value [18:15:13 CEST] <atomnuker> instead you get a huffman coded value which has each coefficient in a small, constant size bitstream [18:15:44 CEST] <atomnuker> usually 4 coefficients with 2 bits each [00:00:00 CEST] --- Sat Jun 30 2018

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ffmpeg.log.20180629
by burek 30 Jun '18

30 Jun '18

[00:08:51 CEST] <Ars> teratorn, every time i have diferent exits [00:09:31 CEST] <Ars> how can i check udp stream availability [00:10:07 CEST] <Ars> am transcoding and multicasting stream like: fmpeg RTMP_Stream udp://225.2.1.11:1300 [00:10:07 CEST] <Ars> Also i have stream check script (ffprobe udp://225.2.1.11:1300), but when executing that command ffmpeg stops. [00:10:08 CEST] <Ars> I understand there is port binding problem. [00:10:08 CEST] <Ars> But how i can check that UDP stream alive or no? How to chek stream on udp://225.2.1.11:1300 ? [01:44:07 CEST] <teratorn> Ars: what is availability going to tell you that monitoring your failures isn't? [02:41:40 CEST] <lemontea> I am having troubles getting a debug build of the ffmpeg with decklink sdk, while release is okay. When --enable-debug --enable-static, I weirdly get this, value 'MT_StaticRelease' doesn't match value 'MTd_StaticDebug' in testfunc.obj C:\dev\ffmpeglib\libavdevice.a(decklink_enc.o) [02:43:20 CEST] <lemontea> The decklink stuff is in MT_StaticRelease code and I am trying to build it with /MTd in visual studio, bothers me that only Release works. Before integrating the decklink SDK, it would work both debug/release. Does anyone know how should I fix this? The MakeFile is not very obvious. [02:43:28 CEST] <adjivas> Hi, please, what version of libquvi need ffmpeg? [02:45:30 CEST] <adjivas> I try to FFmpeg-SIXEL but that not found by pkg-config [12:32:16 CEST] <killown> ffmpeg -i "teste.mp4" -vf "drawtext=text='test ':x=w-tw-20:y=h-th-100:fontsize=0.033*h:fontfile=/scripts/fonts/ttf/Hack-Bold.ttf:fontcolor=white@0.6:shadowcolor=black@0.6:shadowx=2:shadowy=2:box=1: boxcolor=black@0.3:boxborderw=3" "out.mp4" [12:32:27 CEST] <killown> how do I display the test in the top? [12:35:27 CEST] <libertas> Hi, I need to use codec and profile of this line -codec:v h264 -qmin 0 -qmax 40 -crf 18 -profile:v high10 -pix_fmt yuv420p [12:36:16 CEST] <libertas> in a lua script that has different settings: format=yuv444p16,scale=in_color_matrix=$matrix,format=bgr24 [12:37:12 CEST] <libertas> in format, should it be format=yuv420pxx? what should the xx be? or is it not needed? [12:41:05 CEST] <libertas> or put another way, would be nice if someone could build a similar line to the second I wrote using the setting of the first [12:43:46 CEST] <ariyasu_> yes you want 4:2:0 [12:44:12 CEST] <ariyasu_> you probably want -vcodec libx264 [12:50:03 CEST] <libertas> ariyasu_: with the setting in the first line I can view videos in my panasonic tv, with newer profiles I can't, that's why [12:50:35 CEST] <ariyasu_> your tv probably does not support 4:4:4 [13:23:22 CEST] <King_DuckZ> hi, I have this snippet of code https://paste.debian.net/1031274 which is asserting at line 15, the output I get from the commented snipped at line 9 is: https://paste.debian.net/1031273 [13:23:51 CEST] <King_DuckZ> anybody knows what's going on? shouldn't I get some memory allocated by the time the assert is reached? [13:55:45 CEST] <King_DuckZ> picture->buf is also all null :/ [13:55:49 CEST] <King_DuckZ> wth? [14:05:39 CEST] <c_14> picture->data[0] should be 0, no? [14:05:54 CEST] <c_14> and it is so the assertion would obviously get triggered [14:10:12 CEST] <King_DuckZ> shouldn't that function allocate some buffers at least? [14:10:47 CEST] <c_14> it should, yeah [14:16:46 CEST] <c_14> what are format, width and height set to? [14:18:30 CEST] <King_DuckZ> it's printed by my debug code here https://paste.debian.net/1031273 [14:18:45 CEST] <King_DuckZ> I think 0 is YUVsomething [14:25:12 CEST] <c_14> what language is that? [14:30:24 CEST] <c_14> you might want to ask on the user mailing list or wait around here, I'm not really sure what's happening [14:45:44 CEST] <King_DuckZ> <King_DuckZ> hi, I have this snippet of code https://paste.debian.net/1031274 which is asserting at line 15, the output I get from the commented snipped at line 9 is: https://paste.debian.net/1031273 [14:45:47 CEST] <King_DuckZ> c_14: ^^ [14:46:27 CEST] <King_DuckZ> c_14: the first link is c++, the second is just my program's debug output I copy-pasted from my console [14:47:14 CEST] <King_DuckZ> I've changed it to also print buf[0] to 7, they're all null as well [17:46:07 CEST] <dtatulea> Hi. Can ffmpeg show a raw nv12 frame (via ffplay)? [17:59:17 CEST] <DHE> yes, but you'll have to build the command to tell it what the specs are (width, height, format, etc). and maybe set some super-slow framerate to keep it on-screen. like 0.1 fps maybe [17:59:47 CEST] <dtatulea> DHE: thanks. And it doesn't need any container or anything? [18:00:12 CEST] <DHE> you would use the rawvideo "container" [18:00:32 CEST] <DHE> ffplay -f rawvideo -pixfmt nv12 -s 1024x768 -i FILENAME [18:00:56 CEST] <DHE> actually no... let me rewrite that [18:01:13 CEST] <DHE> ffplay -video_size 1024x768 -pixel_format nv12 -framerate 0.1 -i FILENAME [18:07:29 CEST] <if_gaga9> hello guys, i wish implement this scheme https://i.imgur.com/YFKOaYm.png, it's a possible with ffserver? my question is there can ffserver change the source stream on the fly, according external command, which was sended through zmq? [18:26:08 CEST] <dtatulea> DHE: thanks! [19:58:54 CEST] <DHE> I'm using the API to write mpegts streams, but keep getting the error "frame size not set" from the mpegts muxer on startup. only seems to happen on an AAC audio stream. know what the fix for that is? [20:39:44 CEST] <Hello71> did you set the frame size [22:47:15 CEST] <temp> Currently I use "ffmpeg -i <file> -f null -" to read time of an audio file. It shows time=00:00:12.16. How do I increase the precision of the output further? I want time=00:00:12.167 [23:12:30 CEST] <temp> How do i get better time precision when using this command? https://pastebin.com/iTvndR7Q [23:14:10 CEST] <furq> temp: ffprobe -show_entries format=duration file [23:14:50 CEST] <temp> furq: I can't use ffprobe. sigh [23:15:07 CEST] <furq> why not [23:16:18 CEST] <temp> we had some issues before when using ffprobe so we switched to using ffmpeg. Since we already have this in production it would be much easier to get this info from ffmpeg itself (if possible) [00:00:00 CEST] --- Sat Jun 30 2018

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ffmpeg-devel.log.20180628
by burek 29 Jun '18

29 Jun '18

[11:49:08 CEST] <gagandeep> kierank: i now need to start multi threading and have been reading about slice and frame threading [12:10:09 CEST] <kierank> gagandeep: are all samples bitexact in single thread mode [12:16:55 CEST] <gagandeep> other than the mountain sample all samples are bitexact [12:17:17 CEST] <gagandeep> the mountain sample file is not getting decoded in the vlc part [12:17:44 CEST] <gagandeep> i tried to see how to fix that but to not much result [12:17:57 CEST] <gagandeep> though was able to get correct decompanding for codebook 0 [12:18:35 CEST] <gagandeep> the p frame sample i got david is bit exact [12:19:10 CEST] <gagandeep> also mountain sample uses codebook 0 and david told me that they used to encode with it only when they didn't have codebook 1 [12:19:46 CEST] <gagandeep> so i feel right now i should do my other stuff and will try fixing error with codebook 0 alongside [12:41:36 CEST] <kierank> gagandeep: is it a big difference [12:41:50 CEST] <kierank> And did you figure out 3d transformer? [12:41:59 CEST] <kierank> Transform with coefficients from last frame [12:50:47 CEST] <gagandeep> kierank: the quant is like 64 and the vlc code i don't know how reads wrong value to be dequantized to the values a missed by a few integers [12:51:18 CEST] <gagandeep> and also i don't know why vlc part is missing a value between zeros [12:51:38 CEST] <gagandeep> let me send you a screenshot of raw data [12:53:45 CEST] <kierank> You could ask michaelni [12:54:39 CEST] <gagandeep> https://imgur.com/a/BTVrnEj [16:28:41 CEST] <BBB> hey C experts, lets say I have a struct x { int a1, a2, a3, a4, a5, a6; int b1, b2, b3, b4, b5, b6; }; and I want to copy only the a entries in the struct (in reality, they are all different sizes/types/names etc.). I can obviously memcpy(dst, src, offsetof(struct x, b1)), but that relies on b1 not moving within the struct. does C support some kind of labeling so I can add a label halfway (like struct x { int a1, a2, a3, a4, a5, a6; half: int b1, [16:28:41 CEST] <BBB> b3, b4, b5, b6; };), so I can memcpy using offsetof(struct x, half) as size? [16:31:06 CEST] <jdarnley> BBB can you put all the As into one substruct? [16:31:31 CEST] <BBB> everything is possible :-p [16:33:10 CEST] <jdarnley> You might consider "struct x { struct a {}; struct b{} }" then you can just do y.a = x.a [16:33:38 CEST] <ubitux> int field_separator; [16:34:01 CEST] <ubitux> (separator_field if you speak a better english i guess) [16:34:40 CEST] <BBB> right I could obviously just add a separator; does C support zero-sized separators? [16:34:51 CEST] <BBB> I see a unique use case for void [16:35:33 CEST] <jdarnley> Unlikely. I recall something about a zero-length array is possible at the end but really am no expert. [16:39:47 CEST] <BBB> oooo I remember that [16:39:48 CEST] <BBB> brilliant [16:40:09 CEST] <BBB> so evil, I LOVE IT <3 [16:40:32 CEST] <jdarnley> Are you in a C obfuscation contest? [17:00:20 CEST] <atomnuker> BBB: cmon dude I'm itching to work on ffav1 [17:00:27 CEST] <BBB> me too [17:00:36 CEST] <BBB> you know that the holdup is not on my side [17:00:55 CEST] <atomnuker> erm actually no, I thought you started and have been working on it for a month now [17:00:59 CEST] <BBB> I have [17:01:03 CEST] <BBB> its essentially finished [17:01:26 CEST] <atomnuker> like, completely? [17:01:43 CEST] <BBB> a few features are missing, but most things are there and it works correctly [17:01:57 CEST] <atomnuker> with p-frames and loop restoration and global motion [17:02:10 CEST] <BBB> no loop restoration yet [17:02:17 CEST] <BBB> cdef/lpf works [17:02:20 CEST] <BBB> rst not yet [17:02:26 CEST] <atomnuker> rst? [17:03:04 CEST] <BBB> my abbreviation for restoration [17:03:19 CEST] <atomnuker> oh ok, is there simd yet? [17:03:33 CEST] <BBB> some [17:03:57 CEST] <atomnuker> k, I think I know where the holdup is, I'll try to get things moving [17:06:22 CEST] <BBB> all I know, its way above my pay grade [17:08:42 CEST] <atomnuker> BBB: how fast is it compared to aomdec for 1/many threads? [17:08:49 CEST] <BBB> havent tested yet [17:08:59 CEST] <BBB> but I dont think thats a useful question until they are both fully simd'ed [17:09:16 CEST] <BBB> oh look, Im twice as fast <write some simd> oh now the other one is twice as fast [17:09:27 CEST] <BBB> its about design for expected end speed [17:09:34 CEST] <cone-441> ffmpeg 03Gyan Doshi 07master:07987404a8b7: ffmpeg: make loglevel verbose for frame duration warning [17:10:43 CEST] <atomnuker> well, aomdec can barely do 1080p50 realtime, and would be nice for firefox to have 1080p60 playback on reasonably modern machines [17:11:19 CEST] <atomnuker> rumor has it google are planning to roll out av1 on youtube in september [17:12:19 CEST] <BBB> all that is way above my pay grade [17:12:27 CEST] <gnafu> atomnuker: Is that when the first video finishes encoding? [17:12:33 CEST] <BBB> :D [17:12:42 CEST] <BBB> one star for gnafu [17:12:49 CEST] <gnafu> ^^ [17:13:59 CEST] <atomnuker> BBB: anyway, would still be really interesting to know how they compare currently, could you do that if possible? [17:14:02 CEST] <jdarnley> hahaa [17:14:55 CEST] <BBB> not right away, but Ill try in the near future, sure [17:35:40 CEST] <cone-441> ffmpeg 03Mark Thompson 07master:46fb150a81ab: lavfi/colorspace: Add namespace prefix to global functions [17:44:01 CEST] <Gramner> BBB: memcpy(dst, src, offsetof(struct x, a6) + sizeof(dst->a6)) ? [18:12:25 CEST] <jdarnley> Is anyone here familiar with the draw horizontal band feature? Can you answer a few questions of mine? [18:13:31 CEST] <jdarnley> You only call the function for new image data, right? [18:14:19 CEST] <jdarnley> I should specify that I am asking in the context of a decoder, not a library user. [18:15:14 CEST] <jdarnley> You don't call it for lines 0-15 then lines 0-31 but rather 0-15 and 16-31. [18:15:18 CEST] <jdarnley> right? [18:19:11 CEST] <durandal_1707> that misfeature should be removed [18:21:24 CEST] <atomnuker> its alright, I don't mind the API [18:21:44 CEST] <jdarnley> Maybe. I might prefer returning chunks of frames as they're completed but the feature already exists. [18:22:37 CEST] <atomnuker> eh, it makes more sense and is faster to work with rows rather than blocks [18:23:21 CEST] <jdarnley> I do mean whole lines. [18:23:45 CEST] <jdarnley> I am trying to stop using "slice" when I don't mean specific codec features. [18:35:57 CEST] <jdarnley> Another question. If a decoder is using draw_horiz_band() what is it supposed to do at the end of a frame? Can it just return it as normal and the user is supposed to handle it all? [18:54:19 CEST] <jdarnley> Why would attach sample files for common formats for which we already have decoders? [21:37:50 CEST] <cone-914> ffmpeg 03Michael Niedermayer 07master:75a2db552423: avcodec/dpx: Check elements in 12bps planar path [21:37:50 CEST] <cone-914> ffmpeg 03Michael Niedermayer 07master:ed22dc22216f: avformat/movenc: Check that frame_types other than EAC3_FRAME_TYPE_INDEPENDENT have a supported substream id [21:37:50 CEST] <cone-914> ffmpeg 03Michael Niedermayer 07master:00e8181bd97c: avcodec/ac3_parser: Check init_get_bits8() for failure [21:37:50 CEST] <cone-914> ffmpeg 03Michael Niedermayer 07master:95556e27e2c1: avformat/movenc: Do not pass AVCodecParameters in avpriv_request_sample [21:37:50 CEST] <cone-914> ffmpeg 03Michael Niedermayer 07master:2aa9047486db: avcodec/mpeg4videodec: Check read profile before setting it [21:37:51 CEST] <cone-914> ffmpeg 03Michael Niedermayer 07master:310a49f71df7: tools/target_dec_fuzzer: Also optionally fuzz with a parser [21:37:51 CEST] <cone-914> ffmpeg 03Michael Niedermayer 07master:872ea3dfe565: avformat/movenc: Use mov->fc consistently for av_log() [21:37:52 CEST] <cone-914> ffmpeg 03John Stebbins 07master:08b1d1d81225: h264_slice: Fix return of incomplete frames from decoder [23:28:16 CEST] <kierank> jdarnley: iiuc there is a test for draw_horiz_band [23:37:24 CEST] <cone-914> ffmpeg 03Jun Zhao 07master:2730a0fdf47c: configure: fix check for opencl_vaapi_intel_media. [23:37:25 CEST] <cone-914> ffmpeg 03Jun Zhao 07master:e6ac328b79f4: hwcontext_opencl: Remove the opencl_device_init in opencl_device_derive [23:37:26 CEST] <cone-914> ffmpeg 03Jun Zhao 07master:6b453a81bcb5: hwcontext_opencl: remove an unused variable [00:00:00 CEST] --- Fri Jun 29 2018

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ffmpeg.log.20180628
by burek 29 Jun '18

29 Jun '18

[00:59:13 CEST] <Ars> Hi guys, pls help me. [00:59:13 CEST] <Ars> I am transcoding and multicasting stream like: fmpeg RTMP_Stream udp://225.2.1.11:1300 [00:59:13 CEST] <Ars> Also i have stream check script (ffprobe udp://225.2.1.11:1300), but when executing that command ffmpeg stops. [00:59:13 CEST] <Ars> I understand there is port binding problem. [00:59:13 CEST] <Ars> But how i can check that UDP stream alive or no? How to chek stream on udp://225.2.1.11:1300 ? [01:03:02 CEST] <dmc> should I transcode TrueHD to DTS or AC3? I'm using S/PDIF (only have support for passing those through, apparently) [01:03:52 CEST] <dmc> ffmpeg -i <file with DD+ or TrueHD> -f matroska -vcodec copy -acodec ac3 - | vlc - [01:06:06 CEST] <dmc> hopefully that's the best way to do it [01:12:04 CEST] <dmc> same question for FLAC [01:14:09 CEST] <dmc> "ffmpeg ac3 encoder is better than DTS" [01:24:29 CEST] <iive> transcoding will not improve quality [01:24:54 CEST] <iive> flac is lossless, so it will not lower quality... [01:27:03 CEST] <iive> oh, truehd is lossless too. and is dolby's format [02:35:51 CEST] <dmc> didn't mention quality :p [02:36:03 CEST] <dmc> S/PDIF doesn't support those [02:40:56 CEST] <furq> just encode a sample with both and see if you can tell the difference [02:42:41 CEST] <furq> the ac3 encoder is definitely pretty good quality [02:42:55 CEST] <furq> i have no idea about the dts encoder, iirc it's a really old fork of dcaenc [02:43:14 CEST] <furq> but idk if it's been worked on significantly since then [02:44:31 CEST] <dmc> yeah, figured I should just experiment [03:35:05 CEST] <Hello71> what's considered the least worst media container nowadays? matroska? [03:44:24 CEST] <furq> Hello71: probably [03:53:30 CEST] <ChChCherryBom1> mp4 :D [03:54:05 CEST] <Hello71> mp4 seems to have lower overhead, but maybe I'm using matroska wrong [03:54:51 CEST] <ChChCherryBom1> I find mp4 with fast start and ac3 just works on like everything really nicely [03:55:14 CEST] <ChChCherryBom1> most dlna clients expect it properly handle metadata too if you aren't using custom media servers [03:55:16 CEST] <furq> mp4 is fine if you've got supported codecs [03:55:26 CEST] <furq> and it doesn't have the stupid timebase issue that matroska has [03:55:52 CEST] <furq> and it's obviously much better supported [04:10:45 CEST] <Hello71> how come remuxing my video from mp4 to mp4 increases the size by 20k (~0.17%)? I added -fflags +bitexact and -map_metadata -1, so if anything shouldn't it get smaller? [04:11:16 CEST] <Hello71> also the duration changed... odd... [04:14:55 CEST] <Hello71> and the "Total: x demuxed" and x muxed are the same numbers... [04:16:14 CEST] <Hello71> looks like the input streams have slightly different duration, that's probably why [10:46:52 CEST] <drkop> Hello! then i try code from smb:// to smb:// - got an error https://pastebin.com/zuruF7w6 and one second target video. if local to smb:// or vice versa - all is fine. what can i do? thank you! [12:33:56 CEST] <ksj> hello, is it possible to make av codec drop frames if they are out of order? [13:38:03 CEST] <roxlu> hi, probably a silly question :$ but I saw that there is a `delete_filler` option to remote filler nals. I'm curious how to actually use this? (adding -delete_filler clearly isn't the way to go). See https://gist.github.com/roxlu/998b7b3e014e49633d66052d74222337#file-gistfil… [13:54:43 CEST] <roxlu> ah nvm, got it [16:28:41 CEST] <reinaldorauch> Hello. I'm trying to convert mp4 files with audio streams only tagged with MusicBrainz Picard to mp4 files converting the audio stream to aac. But it seems that the codec does not support writing all the tags (the ones that picard adds) into the outfile. I tryied with libmp3lame and it added the metadata right. Is something that I'm missing? [16:33:37 CEST] <reinaldorauch> The commands that I tryed: https://pastebin.com/C6LfAJAx [16:58:36 CEST] <benlieb> Im trying to join two videos together that apparently have different SARs: Input link in1:v0 parameters (size 720x480, SAR 10:11) do not match the corresponding output link in0:v0 parameters (720x480, SAR 1:1) [16:58:36 CEST] <benlieb> [Parsed_concat_4 @ 0x7ff1fd103e80] Failed to configure output pad on Parsed_concat_4 Error reinitializing filters! [16:58:37 CEST] <benlieb> Failed to inject frame into filter network: Invalid argument Error while processing the decoded data for stream #0:0 Conversion failed! [16:58:41 CEST] <benlieb> this is the command: ffmpeg -y -i file1.mp4 -i file.mp4 -filter_complex '[1:v]fade=t=in:st=0:d=2,fade=t=out:st=510:d=2[v];[1:a]afade=t=in:st=0:d=2,afade=t=out:st=510:d=2[a];[0:v][0:a][v][a]concat=a=1 [v][a]' -c:v libx264 -preset slow -crf 26 -r 30 -map '[v]' -map '[a]' file3.mp4 [16:58:52 CEST] <benlieb> can anyone point me in the right direction? [17:26:46 CEST] <atbd> hi, i'm using ffmpeg api to receive an udp stream and save images from it. I let the program run in background and it reboots periodically every 27h30min with "DTS %f / %f out of order" message. I saw this message in compute_pkt_fields but did not see how to correct my bug, furthermore it just mean the bug is after this but maybe not linked... Some of you have an idea on this please ? [17:46:37 CEST] <kepstin> atbd: mpeg-ts in udp? [17:47:18 CEST] <kepstin> that's the max value the mpeg-ts timestamps can have before they roll over [17:47:42 CEST] <pomaranc> that's 26.5h [17:47:54 CEST] <kepstin> er, well, it's pretty close anyways [17:48:22 CEST] <atbd> yes it's mpegts [17:48:56 CEST] <atbd> and when it roll over everything hang? [17:49:53 CEST] <kepstin> applications just have to be designed to deal with the rollover and keep going :/ [17:51:12 CEST] <kepstin> one option is to use a larger timestamp value, and every time it rolls over or jumps back, save the previous value and then rewrite each new packet's pts to be prev_value + pts as [17:51:43 CEST] <kepstin> (ffmpeg's normal pts values are 64bits, so that should work for long enough) [17:52:29 CEST] <atbd> okay! thanks for your explanation [17:52:37 CEST] <atbd> i'll try your solution [18:52:25 CEST] <Chiefmas> Hello, I recently got a system with a Kaby Lake CPU, and I'm trying to get an idea for how much of the work of a transcode with a re-size I should be able to offload onto Quicksync. I had been hoping that I would have much lower CPU time used then I've been able to get. Example command I'm trying: [18:52:55 CEST] <Chiefmas> ffmpeg -hwaccel qsv -init_hw_device qsv=qsv:MFX_IMPL_hw_any -filter_hw_device qsv -i "..\I Am Legend - Trailer.mp4" -vf "format=nv12,hwupload=extra_hw_frames=100,scale_qsv=640:360" -b:v 800k -c:v h264_qsv -c:a copy -y "..\test.mkv" [18:54:14 CEST] <Chiefmas> so taking a 1008P source video, scale down and re-encode as h.264 again. This still seems to use 90%+ on CPU, although I am seeing a good amount of activity on the GPU offload (as reported by task manager). [20:05:05 CEST] <bruce-> hey. should avdevice_list_input_sources return with a negative return value (-78) when querying the "avfoundation" device or am I doing something wrong? [20:12:55 CEST] <teratorn> bruce-: probably [20:13:14 CEST] <teratorn> bruce-: pastebin minimal example code plz [20:32:09 CEST] <lemontea> I have a question about incorporating the DeckLink SDK into the build. I am trying to create build a static DLL, but when building on windows visual studio 64bit, I get this => "Error LNK2038 mismatch detected for '_ITERATOR_DEBUG_LEVEL': value '0' doesn't match value '2' in MyTestFunction.obj ffmpegFuncDLL C:\dev\testffmpeg\libavdevice.a(decklink_enc.o)", when building it, i was using the following configure flags,... [20:32:14 CEST] <roxlu> is it possible to show the packet number with `ffprobe` ? I have to inspect a specific nal and trying to find it :) [20:40:16 CEST] <atomnuker> -show_packets [20:41:53 CEST] <roxlu> atomnuker: that has no counter [20:43:52 CEST] <atomnuker> welp, you can parse the json then and count 'em [20:45:42 CEST] <roxlu> yeah I see there is also -read_intervals ... but specifying a range like: +START_OFFSET|+END_OFFSET (e.g. #200%#1) doesn't work [20:45:56 CEST] <roxlu> I'm trying to figure out what the correct syntax is [20:50:10 CEST] <lemontea> Anyone know about decklink SDK and building it in MSVC, then using the static lib files for visual studio project? [20:51:33 CEST] <roxlu> lemontea: you want to use the decklink sdk? From what I remember there is a .idl file(?) that is used to generate a headers; it's been years that I looked at that [20:52:49 CEST] <lemontea> roxlu: You are correct, i had to use midl from visual studio to generate the big header file on Windows. Used it within Ffmpeg builds fine, and all, but weirdly only the decklink_enc.o is built with a different flag or something? [20:53:35 CEST] <roxlu> what error do you get? [21:01:06 CEST] <lemontea> roxlu: "Error LNK2038 mismatch detected for '_ITERATOR_DEBUG_LEVEL': value '0' doesn't match value '2' in MyTestFunction.obj ffmpegFuncDLL C:\dev\testffmpeg\libavdevice.a(decklink_enc.o)" [21:02:18 CEST] <lemontea> I was trying to compile MSVC with /MD flag, and got that error, and I will try it with /MT, but so many errors. I think I am on the right path. But, I have to figure out how to do that with configure options building ffmpeg [21:02:49 CEST] <roxlu> Ah I think that's due to a debug / release build [21:03:15 CEST] <roxlu> Also check if the architecture of the library is similar as your project [21:03:42 CEST] <roxlu> lemontea: at this page I've got some snippets that show how to do that http://roxlu.com/2013/017/working-set [21:03:42 CEST] <bruce-> teratorn: that's a bit tricky because I am using ffmpeg through javacpp bindings. in the meanwhile I found http://ffmpeg.org/pipermail/libav-user/2018-March/010969.html which makes it look like it is not implemented. [21:06:22 CEST] <lemontea> roxlu: nice page, like your use of Kalman Filter, I have used that before on an Arduino MCU. :) [21:07:21 CEST] <lemontea> It is more like --enable-static --enable-debug sorta thing, but how to differentiate /MT and /MD for MSVC in configure scripts...hmmmm [21:09:05 CEST] <roxlu> lemontea: there is probably a makefile which has those flags somewhere [21:09:15 CEST] <roxlu> maybe try a "grep" [21:14:02 CEST] <Ars> I am transcoding and multicasting stream like: fmpeg RTMP_Stream udp://225.2.1.11:1300 [21:14:02 CEST] <Ars> Also i have stream check script (ffprobe udp://225.2.1.11:1300), but when executing that command ffmpeg stops. [21:14:02 CEST] <Ars> I understand there is port binding problem. [21:14:02 CEST] <Ars> But how i can check that UDP stream alive or no? How to chek stream on udp://225.2.1.11:1300 ? [21:16:45 CEST] <lvlscape> Hi everyone, odd question. We've been learnign ffmpeg for the last week or two in our hackerspace. Curious for any recomended readings /books / reference materials relating to using which codecs for what. I've been web-searchign and grabbign other peoples config recomendations, tryign them out, and taco-belling them together. [21:23:57 CEST] <lemontea> lvlscape: inside ffmpeg docs/examples is a very good place to start, pull from latest git, they updated their examples [21:26:27 CEST] <lvlscape> @lemontea thanks, i'll dig down that direction. thankfully all the machiens are linux so capturing has been consistant with x11grab and v4l2 for grabbing the xorg desktops / webcams in our space... [21:27:25 CEST] <lvlscape> related/unrelated question.. what do peopel hook into to stream the screen output of a headless server, thinking a tty-session, or a hdmi output.. was trying to research that but was lackign terms. [21:30:13 CEST] <lemontea> lvlscape: I guess a simple approach is stream it out to a RTMP URL, and perhaps you have a rtmpdump server somewhere and people on the network can play it through vlc [21:32:01 CEST] <lvlscape> yeah, already have a RTMP server setup, i got v4l2 to conenct to webcam devices, but unsure where to hook it to the terminal, web-searched for 2-4 hours yesterday ont he topic and was about to dive into that, wasnt sure if iw as even approachign it the right way or which 'f is the right tool, im like 90% sure its v4l2 [21:36:28 CEST] <lemontea> If you're using the API, avformat_open_input(rtmpurlhere); like this? [21:45:36 CEST] <lvlscape> lemontea: can avformat access the hdmi-out / tty session and stream that correctly as video? .... and as i was typign this i was still web-searcign and i think my mind was just blown... i think x11grab doesnt just grab xserver stuff, and may be able to do what i need.. im goign to go read a lot more, order one of these books, check the docs, and keep hanging here, thanks for the direction [22:10:47 CEST] <kepstin> lvlscape: with some graphics drivers on linux you can use 'kmsgrab' input device to grab the entire screen output (including console display) [22:12:00 CEST] <lvlscape> kepstin: oh wow, i hadnt read anything about that one yet. thank you, im goign to test that too. [22:12:14 CEST] <lvlscape> read about that now + test that today * [22:21:08 CEST] <Ars> guys, how you monitoring ffmpeg process ? [22:21:23 CEST] <Ars> and restart it ? [22:21:56 CEST] <teratorn> Ars: exit code? [22:23:12 CEST] <Ars> some time process die without exit, i am monitoring with monit but have problems [22:24:42 CEST] <teratorn> dies how? [22:24:59 CEST] <teratorn> you can use "-progress -" on the command line to get parsable progress output [22:25:21 CEST] <BtbN> there is always an exit code if a proces quits [23:03:51 CEST] <lemontea> roxlu: still trying to find flags and hoping what im replacing in these two lines are making a difference. not a pro at building [23:33:27 CEST] <lemontea> If anyone is interested, I solved my build problem by using /MT flag for importing the decklink SDK, and then making sure I am linking these files according to UCRT changes => https://blogs.msdn.microsoft.com/vcblog/2015/03/03/introducing-the-universa… [00:00:00 CEST] --- Fri Jun 29 2018

1 0

ffmpeg-devel.log.20180627
by burek 28 Jun '18

28 Jun '18

[08:43:17 CEST] <cone-185> ffmpeg 03Lukas Rusak 07master:7395f13df94e: v4l2_m2m: decoder: fix memory leak [11:11:58 CEST] <Thirumalai_K> anyone had issues with compiling latest ffmpeg master with libaom? [11:13:02 CEST] <Thirumalai_K> a0ac49e38ee1d1011c394d7be67d0f08b2281526 causes issue when the aom is compiled from the source archive [11:14:51 CEST] <Thirumalai_K> libaom's version.cmake typically tries to generates the version code from "git describe". I think, If the source path is not git, then it's retrieved from changelog [11:15:45 CEST] <Thirumalai_K> so, pkg-config file aom.pc always has version code as 0.1.0 if the source path is not git [11:16:09 CEST] <nevcairiel> sounds like a problem they need to solve [11:18:59 CEST] <Thirumalai_K> We can revert the minimum version code requirement till it's resolved by them [11:19:32 CEST] <nevcairiel> but there is a reason its set like that [11:19:39 CEST] <nevcairiel> we dont want to link against old versions [11:23:19 CEST] <Thirumalai_K> @nevcairiel Understood. Thank You [11:25:11 CEST] <nevcairiel> noone ever expects a 1.0.0 release of anything to be actually usable. :D its probably the first actual file release they did, so they never ran into this particular problem before [11:28:43 CEST] <Thirumalai_K> Yes. It's still in development. So, The version code issue may not be fixed in priority. [11:29:20 CEST] <Thirumalai_K> I will change my build scripts to always clone rather than getting the archive. [11:29:44 CEST] <nevcairiel> did you let them know about this problem in the archive? [11:30:13 CEST] <Thirumalai_K> not yet. I will try to create a ticket. [13:51:56 CEST] <cone-185> ffmpeg 03Jacob Trimble 07master:ad2ae21010c4: avutil/encryption_info: Fix documentation problem. [13:51:57 CEST] <cone-185> ffmpeg 03Jacob Trimble 07master:96281ed68904: avutil/gitignore: Ignore integer test binary. [13:51:58 CEST] <cone-185> ffmpeg 03Jacob Trimble 07master:69bd73b3ff87: libavutil/encryption_info: Add unit tests. [13:51:59 CEST] <cone-185> ffmpeg 03Michael Niedermayer 07master:b31189881a4c: avcodec/ra144: Fix integer overflow in ff_eval_refl() [13:52:00 CEST] <cone-185> ffmpeg 03Michael Niedermayer 07master:71ba79702d33: avcodec/dvbsubdec: Compute scoretab without iterating over pixels multiple times in compute_default_clut() [13:52:01 CEST] <cone-185> ffmpeg 03Michael Niedermayer 07master:f59c4e43915e: avcodec/escape124: Fix spelling errors in comment [16:05:45 CEST] <cone-185> ffmpeg 03Timo Rothenpieler 07master:3f953379e1c7: avcodec/qsvenc: fix version detection on cygwin [00:00:00 CEST] --- Thu Jun 28 2018

1 0

ffmpeg.log.20180627
by burek 28 Jun '18

28 Jun '18

[00:43:01 CEST] <ChChCherryBom1> https://pastebin.com/pXgVtuWR [00:43:23 CEST] <ChChCherryBom1> forced_subs_only and disposition forced seem to not be working, any ideas? [01:34:21 CEST] <darkdrgn2k> hi all [01:34:43 CEST] <darkdrgn2k> if im receivng an RTMP stream, and the source feed to the RTMP server stops, then starts up again [01:34:59 CEST] <darkdrgn2k> is tehre a wya to have the ffmpeg that is accessing the stream from the RTMP server continue to work [01:35:28 CEST] <darkdrgn2k> i get "invalid dropping st:0" [03:44:02 CEST] <acoctres> Finally home to try out that USB capture device. [04:18:53 CEST] <Dude> does anyone have sha256 fingerprint for latest ffmpeg download? [04:19:00 CEST] <Dude> where are those usually posted at? [05:27:49 CEST] <kepstin> Dude is gone, but the source downloads have pgp signatures... made with some key, I suppose, I have no idea how to validate those :/ [05:35:08 CEST] <acos> Hmmm. Wow not sure [07:57:54 CEST] <hendry> what is the output format when i want my output to be "mpegts"? [08:44:12 CEST] <furq> hendry: mpegts [08:44:21 CEST] <furq> if you mean the file extension then .ts [08:46:05 CEST] <hendry> furq: ah [08:46:30 CEST] <hendry> hoping FCPX will understand that over .mkv (which it doesn't) [08:50:14 CEST] <furq> if you've got h264 then you probably want mp4 [08:50:28 CEST] <furq> if you don't have h264 then good luck in general i guess [08:52:38 CEST] <ritsuka> or set it directly to .mov [09:40:37 CEST] <blayze> av1! [10:34:50 CEST] <hendry> ritsuka: hmm, sometimes MacOS has issues with the .mov that ffmpeg produces. Urgh [10:35:12 CEST] <ritsuka> for example? [10:57:40 CEST] <hendry> ritsuka: can't playback this file at all https://s.natalian.org/2018-06-27/1530086668.mov [10:57:55 CEST] <hendry> ritsuka: mpv on Archlinux plays it back. Warning I sneeze early on. [10:59:47 CEST] <ritsuka> QuickTime can't playback Hi444PP h.264 [11:05:01 CEST] <hendry> ritsuka: i just set the output file as .mov. What's a good file format to target from x11grab so that MacOS can play it back? [11:05:38 CEST] <JEEB> if you want RGB lossless, use Ut Video and grab the QT component from the format's author's site [11:05:53 CEST] <ritsuka> if you want to edit it in final cut and you have some free disk space, prores (if the ffmpeg encoder is fast enough) [11:06:34 CEST] <JEEB> if you are going that big might as well do the Ut Video way [11:06:36 CEST] <JEEB> and get lossless [11:06:43 CEST] <JEEB> but yes, prores is the other thing [11:06:59 CEST] <ritsuka> I don't know if final cut supports the classic 32bit quicktime components [11:07:10 CEST] <JEEB> didn't he publish 64bit as well? [11:07:19 CEST] <JEEB> not that I've used it, I just know it's getting used in the mac area [11:07:33 CEST] <ritsuka> there is no public api for core media components, so you can't make a 64bit component/plugin [11:28:23 CEST] <pagios> hello, any idea how to calculate the average RAM consumption of a given stream? Say an 720p standard SD stream consuming 1GB/hour [11:30:18 CEST] <BtbN> RAM consumption of a stream? [11:30:44 CEST] <BtbN> Every proper player won't keep more than a few frames in memory. [11:30:58 CEST] <BtbN> And a few seconds of buffer [11:30:58 CEST] <pagios> on the server side i mean [11:31:00 CEST] <pagios> serving the stream [11:31:16 CEST] <BtbN> What kind of stream? [11:31:23 CEST] <pagios> HD stream video/audio [11:31:26 CEST] <BtbN> ... [11:31:30 CEST] <BtbN> HLS, RTMP, RTP, ... [11:31:34 CEST] <pagios> hls [11:31:46 CEST] <BtbN> I don't see why HLS would need RAM on the server. [11:32:10 CEST] <pagios> rtmp needs? [11:32:39 CEST] <BtbN> It's a stream, the whole point is to not buffer it, but to send it. [11:32:50 CEST] <pagios> BtbN, so say a server is running 2GB of ram, how many clients can it serves concurrently? [11:33:10 CEST] <pagios> 100?1000? [11:33:11 CEST] <BtbN> That's impossible to say. [11:33:54 CEST] <BtbN> I'd guess it hits other bottlenecks way before even getting close to the RAM limit. [11:34:22 CEST] <BtbN> Most prominently it will clog its Network rather quickly [11:34:35 CEST] <pagios> even if on 2GB? [11:34:45 CEST] <BtbN> ...? [11:34:58 CEST] <BtbN> Adding more RAM doesn't magically make the network faster. [11:35:48 CEST] <pagios> i mean if you have 1TB network bandwidth [11:35:54 CEST] <pagios> i am focusing on the ram size [11:35:56 CEST] <BtbN> You don't [11:36:03 CEST] <BtbN> If you're lucky you have 1Gbit [11:37:17 CEST] <BtbN> And with a ~2.5Mbit/s stream like you're describing, the theoretical limit is 400 parallel streams. In practice probably notably less [11:37:31 CEST] <BtbN> Due to overhead and other traffic [11:37:52 CEST] <pagios> 400 parallel if over 1gbit?! [11:38:18 CEST] <BtbN> Just do the math. 1000Mbps/2.5Mbps -> 400 [11:40:43 CEST] <pagios> i am assuming 1 sd stream is 1GB / hour [11:40:56 CEST] <BtbN> yes, that's what I calculated with. [11:42:06 CEST] <BtbN> 1GB/h is 1024*8 Mbit/h. Is 2.28Mbit/s. [11:42:39 CEST] <BtbN> And 1000Mbit/s of a gigabit connection divided by those 2.28Mbit/s gives ~440 [11:43:18 CEST] <BtbN> And that's very idealized. In practice with HTTP overhead and other stuff happening on the connection, you're probably lucky if you manage 200 stable streams in parallel. [11:49:17 CEST] <pagios> so 1GB/h for a stream recorded with a server running 1Gbps can serve 200 concurrent user [11:50:31 CEST] <pagios> right BtbN ? [11:50:41 CEST] <BtbN> There is no hard rule like you wish for [11:51:04 CEST] <BtbN> 200 seems realistic to me, but who knows what other factors will come up in the real world [14:38:47 CEST] <Guest47278> Hi, I'm using custom I/O for avformat_open_input(), the custom I/O is reading from UDP, the avformat_open_input() gets blacked and keeps calling the UDP receiving function. Where without custom I/O it's not blocked, any suggestions how to unblock it? Thanks [14:41:41 CEST] <JEEB> by default UDP doesn't have an I/O timeout if that's what you mean [14:41:48 CEST] <JEEB> if you look into libavformat/udp.c [14:42:36 CEST] <Guest47278> I implemented my own udp receiving function. [14:43:06 CEST] <JEEB> yes [14:43:17 CEST] <Guest47278> Without custom I/O, but also udp input, it's not blocked. [14:43:25 CEST] <JEEB> oh it was that way [14:43:36 CEST] <JEEB> I understood that as "my custom thing works, the default doesn't2 [14:44:33 CEST] <Guest47278> It's "my custome thing blocks, the default doesnt" [14:44:49 CEST] <Guest47278> Both are udp input [14:46:15 CEST] <Guest47278> https://ffmpeg.org/pipermail/libav-user/2012-July/002495.html [14:46:21 CEST] <JEEB> the default one has a separate thread [14:46:26 CEST] <JEEB> which blocks [14:46:34 CEST] <Guest47278> I see [14:46:35 CEST] <JEEB> and keeps reading the input [14:46:57 CEST] <JEEB> then you have a circular buffer which the lavf API keeps reading [14:47:59 CEST] <Guest47278> so I have to put my custom I/O into a separate thread? [15:02:33 CEST] <Guest47278> where are the answers for questions in such post: (https://ffmpeg.org/pipermail/libav-user/2012-July/002495.html) [15:02:54 CEST] <Guest47278> Or it's not answered yet? [15:04:19 CEST] <JEEB> https://ffmpeg.org/pipermail/libav-user/2012-July/thread.html#2495 [15:04:22 CEST] <JEEB> see the thread [15:04:38 CEST] <JEEB> ctrl+F "MOV and custom IO" [15:06:55 CEST] <Guest47278> Thanks [16:03:40 CEST] <pagios> ? [16:05:02 CEST] <Mathieu_Du> hrm, https://github.com/FFmpeg/FFmpeg/blob/master/libavutil/opt.c#L217-L218 causes av_opt_set (otp, name, NULL, flags); to return ENOMEM, is this really the expected behaviour ? [16:20:08 CEST] <fnstudio> hi, this page https://www.qubes-os.org/doc/live-usb/ says "...creating a USB boot drive for testing purposes..."; "testing" would make me think that usb-live mode is still somewhat experimental? [16:20:55 CEST] <fnstudio> however, if i look at https://www.qubes-os.org/doc/installation-guide/#installing-to-a-usb-drive it seems usb-live mode is now fully supported? [16:21:32 CEST] <fnstudio> no mention of "testing purposes" in the new installation guide [16:22:04 CEST] <fnstudio> (ooops wrong chat - sorry! :) ) [16:33:00 CEST] <ChChCherryBom1> https://pastebin.com/KqkNn3N5 [16:33:16 CEST] <ChChCherryBom1> when converting subs to mov_text, they lose the timestamps or w/e [16:33:30 CEST] <ChChCherryBom1> the forced language subs all play at the opening scene [16:33:42 CEST] <ChChCherryBom1> but they're fine in the source file [16:43:52 CEST] <kubast2> Hey I'm looking at kdenlive preset ,but I don't quiet understand what this portion does: bf 0 -vb 0 ,I know that -bf 0 disables b frames [16:43:58 CEST] <kubast2> but idk what does -vb 0 do [16:44:14 CEST] <termos> I'm trying to push some HLS stream to an HTTP entry point, but there is an error where FFmpeg is not waiting for HTTP responses and just closes the HTTP connection prematurely. Is there a way to make FFmpeg wait for responses and not just close like that? [16:44:19 CEST] <kubast2> encoding proxy profile* [16:44:33 CEST] <kubast2> tried to do a search on ffmpeg documentation [16:44:38 CEST] <kubast2> but the only option I found ws vbr [16:44:54 CEST] <kubast2> or variations on vbr ,none of them describe "-vb" behaviour [16:48:01 CEST] <kubast2> ah [16:48:04 CEST] <kubast2> bitrate [16:59:49 CEST] <tuna> So I setup my hardware decoder, similar to my hardware encoder...the only difference being I setup a software decoder first to parse the incoming data off the RTSP stream so I can get my height and width of the image...then I setup a hardware decoder (and its frame contexts) with that width height, as well as the cached packet off of the first setup (of the software decoder)...what am I missing? When I "avcodec_open2" the codec context [17:00:33 CEST] <tuna> failed reading 0xFFFFFFF location..... [17:01:07 CEST] <tuna> I went through the c and frames context structs via MSVC debugger, but I dont see anything telling [17:01:20 CEST] <tuna> Maybe I am missing something that is causing this to happen [17:23:57 CEST] <tuna> Happens in both release and debug mode [17:25:48 CEST] <BtbN> There are probe functions that find that stuff for you. No need for an extra decoder [17:26:37 CEST] <ChChCherryBom1> im trying with -sub_charenc UTF-8 for giggles [17:30:55 CEST] <tuna> BtbN: How do I setup the AVHWFramesContext without width and height...will it just work? [17:33:27 CEST] <tuna> Seems to just die when I do not include width/height....for the hwframe [17:37:04 CEST] <saml_> slack is down lols [17:40:16 CEST] <BtbN> tuna, you don't. You probe the stream for those parameters first. [17:41:52 CEST] <acos> Hmmm not sure [17:43:54 CEST] <tuna> BtbN: Could you point me toward those probe functions? [17:52:17 CEST] <BtbN> Primarily avformat_find_stream_info [17:55:36 CEST] <saml_> tuna, what are you building? just curious [18:04:18 CEST] <tuna> BtbN: How does that work when you are reading in packets vs using a file? [18:04:24 CEST] <tuna> All I see is file related stuff [18:04:42 CEST] <tuna> saml_: I am setting up an RTSP stream that need as low latency as possible [18:04:57 CEST] <tuna> both encoding and decoding side...I cannot really say what it is for [18:05:00 CEST] <BtbN> It works with anything avformat can read [18:06:13 CEST] <BtbN> you call avformat_find_stream_info on it, and after that the contextx streams array is populated, and in the codecpar of each stream you find the info you want [18:06:23 CEST] <ChChCherryBom1> https://pastebin.com/1FWdrpPb [18:06:34 CEST] <ChChCherryBom1> ok, so I tried to just convert the subs only instead and see what happens [18:06:50 CEST] <ChChCherryBom1> [3gp @ 0x7f98f6000000] Application provided duration: 2999288000 / timestamp: 2999288000 is out of range for mov/mp4 format [3gp @ 0x7f98f6000000] pts has no value [18:11:41 CEST] <saml_> thats good [18:12:00 CEST] <saml_> i just use ffmeg command [18:19:42 CEST] <saml_> ChChCherryBom1, maybe add -c:v libx264 [18:20:23 CEST] <ChChCherryBom1> this is the video stream [18:20:25 CEST] <ChChCherryBom1> Stream #0:0(eng): Video: h264 (High), yuv420p(progressive), 1920x824, SAR 1:1 DAR 240:103, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) [18:20:35 CEST] <ChChCherryBom1> I don't think thats related is it? [18:20:36 CEST] <saml_> ah i see [18:20:40 CEST] <saml_> yeah sorry my bad [18:20:44 CEST] <ChChCherryBom1> its a timing issue from set to mov_text [18:20:50 CEST] <ChChCherryBom1> srt* [18:21:14 CEST] <JEEB> so it didn't like the timestamp/duration given from the input [18:21:18 CEST] <JEEB> for one reason or another [18:21:28 CEST] <ChChCherryBom1> yeah [18:21:50 CEST] <JEEB> movenc.c has various "sanity" checks, some of which are dubious, some are not [18:22:01 CEST] <JEEB> (movenc.c handles mov, mp4, 3gpp etc) [18:22:18 CEST] <saml_> you got input files available online? [18:22:43 CEST] <JEEB> it doesn't exactly help when mov and mp4 f.ex. sometimes have a biiiiit different specification [18:22:47 CEST] <ChChCherryBom1> the srt? [18:23:11 CEST] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/movenc.c;h=e2c56… [18:23:23 CEST] <ChChCherryBom1> https://github.com/XeresRazor/subler this app uses ffmpeg, and converts them fine [18:23:51 CEST] <JEEB> ok, so if DTS is smaller than reference OR duration is larger or equals INT_MAX [18:24:11 CEST] <JEEB> ChChCherryBom1: probably however it does that ends up with a different time base? [18:24:17 CEST] <JEEB> this sounds like a really huge time base [18:24:27 CEST] <JEEB> if it has a time base of 2999288000 [18:24:32 CEST] <JEEB> uhh, not time base [18:24:33 CEST] <ChChCherryBom1> oh let me see [18:24:35 CEST] <JEEB> but duration [18:24:39 CEST] <ChChCherryBom1> I didn't think of checking its encoded subs [18:25:41 CEST] <JEEB> also I think durations are one of the things where it's not simple to have a single sanity check for both MOV and MP4 [18:25:46 CEST] <JEEB> since you can either have 32bit or 64bit values [18:25:53 CEST] <JEEB> but still, it should print the time base there too [18:25:58 CEST] <JEEB> 'cause that value is huge [18:26:01 CEST] <JEEB> for a duration [18:26:18 CEST] <JEEB> like you'd expect SRT input have a time base of... 1000 or so? [18:26:27 CEST] <JEEB> because you've got .000 usually [18:26:35 CEST] <JEEB> and thus 1000 ticks is one second [18:26:35 CEST] <ChChCherryBom1> 1 00:49:59,288 --> 00:50:00,789 Going somewhere, Solo? [18:26:39 CEST] <ChChCherryBom1> thats the srt [18:27:19 CEST] <JEEB> uhh [18:27:24 CEST] <JEEB> aren't those supposed to be dots? [18:27:44 CEST] <FishPencil> So AOM is 1.0.0 now, does that mean it's usable? Is the bitstream finally done? [18:27:56 CEST] <ChChCherryBom1> ffmpeg -i Star\ Wars\ A\ New\ Hope.mkv -hide_banner -map 0:2 out.srt [18:27:59 CEST] <ChChCherryBom1> thats what it produced [18:28:20 CEST] <saml_> what's the video? .3gp? [18:28:22 CEST] <JEEB> FishPencil: it /should/ be normatively frozen now except for bugs [18:28:44 CEST] <ChChCherryBom1> mkv/h264/dts/srt input [18:28:53 CEST] <JEEB> FishPencil: they're still working on them matroska and mp4 specs though :D [18:29:04 CEST] <JEEB> so have fun waiting for them to get that done now [18:29:12 CEST] <saml_> ChChCherryBom1, maybe try -movflags +genpts [18:29:24 CEST] <FishPencil> Have they moved onto making it faster yet? [18:29:58 CEST] <JEEB> ChChCherryBom1: interesting. is there some really long line there? [18:29:58 CEST] <saml_> ffmpeg -fflags +genpts sorry. not -movflags [18:30:17 CEST] <JEEB> ChChCherryBom1: because it sounds like some subtitle line has a very very long duration [18:30:25 CEST] <JEEB> (or the time base goes bonkers) [18:31:14 CEST] <JEEB> and the time base for SRT indeed should be 1000 [18:31:21 CEST] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/srtdec.c;h=56bd0… [18:31:45 CEST] <saml_> and srt usually is 1\n<timestamp> --> <timestamp>\n<text> [18:31:53 CEST] <benlieb> Im returning to a project I started about 4 years ago. Running some of the commands I get this warning: [18:31:54 CEST] <benlieb> -qscale is ignored, -crf is recommended [18:32:25 CEST] <JEEB> benlieb: not sure which encoders utilize qscale [18:32:34 CEST] <saml_> -crf 23 [18:32:37 CEST] <JEEB> every encoder tends to have its own parameters [18:32:40 CEST] <benlieb> ffmpeg -y -ss 0.0 -t 512.34 -i file.mp4 -q:v 8 -vf scale=104:-2 -aspect 720:480 'file.0-512.34.mp4' [18:32:59 CEST] <JEEB> benlieb: that most likely picks libx264 as the video encoder or something [18:33:00 CEST] <benlieb> I looked for -crf on this page, but didnt see it: https://www.ffmpeg.org/ffmpeg.html [18:33:09 CEST] <JEEB> ffmpeg-all.html [18:33:19 CEST] <JEEB> it's an AVOption for a specific encoder [18:33:33 CEST] <JEEB> for the mpeg4 (part 2) encoder -q:v is the best you can get :P [18:33:39 CEST] <JEEB> for anything more modern there most likely is something better [18:33:43 CEST] <benlieb> ah ok [18:33:57 CEST] <JEEB> you didn't set an encoder, and thus you got whatever boba fett set [18:34:00 CEST] <benlieb> should I specify the encoder specifically? Id like to use h264 [18:34:05 CEST] <JEEB> yes, you should [18:34:10 CEST] <JEEB> -c:v libx264 for H.264 [18:34:17 CEST] <JEEB> (and your FFmpeg has to be built with x264) [18:34:28 CEST] <benlieb> Ok Ill give that a go! [18:34:37 CEST] <benlieb> thats an output option right? [18:34:57 CEST] <JEEB> yes, after input [18:35:35 CEST] <ChChCherryBom1> its made no difference [18:35:53 CEST] <JEEB> ChChCherryBom1: saml_ seems to have been just throwing various stuff at the wall [18:35:55 CEST] <benlieb> I was never super good at ffmpeg, so theres probably a number of things that need adjustment. Thanks for being such a friendly channel ! [18:36:09 CEST] <JEEB> ChChCherryBom1: as I noted, is there some super long subtitle? [18:36:30 CEST] <JEEB> unless somehow ffmpeg.c happens to generate a really weird time base for the subtitle track? [18:36:47 CEST] <JEEB> ChChCherryBom1: can you post the full terminal log and the command line onto a pastebin and link it? [18:36:54 CEST] <JEEB> where you are trying to create the movtext track [18:36:59 CEST] <ChChCherryBom1> I just checked, theres no empty space [18:37:00 CEST] <JEEB> which is where the error is happening [18:37:07 CEST] <ChChCherryBom1> yeah one moment [18:37:11 CEST] <JEEB> ChChCherryBom1: not empty space, a long line. in time. [18:37:34 CEST] <ChChCherryBom1> oh like a 10 minute subtitle segment? [18:37:51 CEST] <JEEB> that really sounds even longer than that [18:37:59 CEST] <benlieb> where is the link to ffmpeg-all.html? [18:38:01 CEST] <JEEB> although those values by themselves are rather useless [18:38:06 CEST] <JEEB> https://www.ffmpeg.org/ffmpeg-all.html [18:38:11 CEST] <JEEB> ^ benlieb [18:38:19 CEST] <benlieb> I can go there manually, but Id like to know how to navigate back from the ohome page [18:38:27 CEST] <JEEB> no idea, I just always google it [18:38:41 CEST] <JEEB> or you can in ffmpeg.c query the AVOptions for a specific encoder [18:38:44 CEST] <JEEB> from the command line [18:39:07 CEST] <JEEB> ffmpeg -h encoder=libx264 [18:39:09 CEST] <JEEB> for example [18:39:46 CEST] <benlieb> ah, theres a teeny tiny link under 7 See Also [18:48:02 CEST] <baschdel__> I know this isn't ffmpeg specific but: does somebody know What codec/format would work on a 6 year old TV (That can read digital video)? [18:48:14 CEST] <JEEB> it can really be random [18:48:42 CEST] <JEEB> I would first test with mp4 with its index in the front, AVC/H.264 video (main profile, level 4.0) and AAC audio [18:49:17 CEST] <JEEB> and resolution being something like 1280x720 or 1920x1080. [18:56:16 CEST] <saml_> ah, -fflags +ptsgen was from https://superuser.com/questions/710008/how-to-get-rid-of-ffmpeg-pts-has-no-… [18:56:40 CEST] <saml_> i'm randomly googling and spitting out things [18:56:50 CEST] <JEEB> anyways, I'd wait for the full log and command [18:57:11 CEST] <JEEB> anyways, according to the quoted part it just seems that either something is making the subtitle time base be insanely large [18:57:19 CEST] <JEEB> or there's a subtitle that goes on for a long, long time [18:57:36 CEST] <ChChCherryBom1> terminal output https://pastebin.com/rz5iX3sg [18:57:39 CEST] <ChChCherryBom1> srt file https://pastebin.com/JFQ3DNYt [18:58:18 CEST] <JEEB> ah yes [18:58:43 CEST] <JEEB> chapters get made as movtext because MOV was weird like that [18:59:49 CEST] <JEEB> so unless you have a long subtitle line then I guess something is making the time base be insanely high there [19:00:08 CEST] <JEEB> (thank you movenc for not telling which) [19:01:31 CEST] <JEEB> ChChCherryBom1: if you want I can give you a patch that would add some extra logging into that error but I want to have a run in the forest first before I start hacking on that :P [19:02:26 CEST] <ChChCherryBom1> no problem [19:02:49 CEST] <JEEB> ChChCherryBom1: aaah your input is straight out of the matroska :P and the first timestamp is at 50min D: [19:02:54 CEST] <ChChCherryBom1> im just wondering if its possible to -map_chapters 0 then map them later on as another track [19:03:01 CEST] <JEEB> no, it's probably not chapters [19:03:15 CEST] <JEEB> just noticed the subtitles starting at 49min mark :P [19:03:42 CEST] <ChChCherryBom1> no it is [19:03:46 CEST] <ChChCherryBom1> thats whats its doing [19:04:09 CEST] <ChChCherryBom1> https://pastebin.com/mNfx4HrB [19:04:19 CEST] <ChChCherryBom1> thats the srt to mov_text [19:04:29 CEST] <ChChCherryBom1> its chapter timestamps and raw subtitle dumb without timestamps [19:04:34 CEST] <ChChCherryBom1> dump* [19:05:11 CEST] <JEEB> anyways, running for me [19:05:13 CEST] <donics> I'm having trouble really understanding how the order of command line options works [19:05:18 CEST] <donics> is there a page on that somewhere [19:13:50 CEST] <ChChCherryBom1> afaik it will tell you in error output if the command is in wrong order [19:24:12 CEST] <ChChCherryBom1> srt to mov_text log: https://pastebin.com/d0Z4XCqL [19:24:31 CEST] <furq> donics: ffmpeg [input1 options] -i input1 [input2 options] -i input2 [...] [output1 options] output1 [output2 options] output2 [...] [19:24:54 CEST] <furq> and then global options whereever [19:25:12 CEST] <furq> also input options aren't inherited from earlier inputs, and the same with outputs [19:25:18 CEST] <donics> ok it's delimited with -i that makes sense [19:29:06 CEST] <ntd> if i'm using an ffmpeg filtergraph to generate a grid view of 9 inputs and then play it with mplayer, this doesn't happen (a delay builds up between the feeds though). using mplayer and -geometry to have each grid be it's own processm this happens: [19:29:08 CEST] <ntd> https://pastebin.mozilla.org/9088615 [19:29:23 CEST] <ntd> nouveau borks, display output broken until reboot [19:29:54 CEST] <ntd> happens every other week on a quadro 5000, every third hour on an NVS290. not exactly on-topic, but any ideas? [19:30:22 CEST] <ntd> exact same setup/install with intel or amdgpu: no problemo [19:52:24 CEST] <kepstin> sounds like a driver bug to me. Unlike the intel and amd drivers, nouveau is all reverse-engineered, so it'll have bugs like that more often. Consider using the nvidia proprietary drivers instead. [19:53:52 CEST] <ntd> yeah... bigger footprint, dkms hell, etc, etc [19:54:32 CEST] <ntd> also, some of the boxes have ancient gpus (like the NVS 290), afaik deprecated in terms of driver support [20:18:35 CEST] <furq> you could maybe consider using mpv as well [20:18:48 CEST] <furq> i have no idea if it's better with nouveau but it's worth a shot [20:19:38 CEST] <ntd> right [20:46:58 CEST] <owner1> exit [20:47:02 CEST] <owner1> quit [21:25:00 CEST] <benlieb> running some commands from an old project and getting this error: No such filter: 'ass' [21:25:18 CEST] <benlieb> running with mac homebrew [21:25:40 CEST] <benlieb> tried reinstalling with --enable-libass but getting this during re-install: ffmpeg: this formula has no --enable-libass option so it will be ignored! [21:25:50 CEST] <benlieb> how can I get the ass filter running? [21:28:02 CEST] <benlieb> hm, maybe I need with-libass [21:30:40 CEST] <ChChCherryBom1> https://trac.ffmpeg.org/ticket/6021 [21:30:44 CEST] <ChChCherryBom1> I wonder if this is related [21:31:04 CEST] <ChChCherryBom1> im going to compare valid srt->mov_text against what ffmpeg is outputting and find the diff [21:55:26 CEST] <im_dumb> which libx264 param is the option 'rc=cbr' mapped to? or is there a way i can find out? [21:58:29 CEST] <kepstin> "rc=cbr" isn't an option, it's just something printed in the log output, I think. [21:58:51 CEST] <kepstin> cbr mode is activated by using vbv controls with minrate=maxrate=bitrate, iirc. [21:59:19 CEST] <kepstin> (you may additionally need other options) [21:59:55 CEST] <kepstin> I think that's -nal-hrd cbr [22:00:18 CEST] <im_dumb> ah ok [22:00:27 CEST] <kepstin> of course, in general you want to not use CBR unless you are required to do so to meet some spec [22:00:46 CEST] <im_dumb> i have to meet a spec sadly :( [22:01:39 CEST] <im_dumb> trying to get as low latency as possible out of a custom encoder we built - of course ffmpeg is miles ahead and with a few simple command line options the latency is like .5s, but using our custom built encoder it's like 1.5 sec... so just trying to match what ffmpeg does in totality with the stuff we have here [22:02:03 CEST] <im_dumb> what about things like 'ip_ratio=0' and 'aq=0'? [22:04:07 CEST] <kepstin> im_dumb: those are just some quality things (related to how it chooses quantizers). Nothing you should touch manually in most cases, and it shouldn't affect latency. [22:05:25 CEST] <kepstin> in general, when using x264, you should set the preset (speed option), options relating to rate control (either bitrate settings or crf), and if needed add a tune setting and/or set the profile limit. [22:06:12 CEST] <im_dumb> ah ok [22:06:33 CEST] <im_dumb> i added ultrafast/zerolatency for my preset+tune settings, but they aren't that good [22:08:02 CEST] <kepstin> zerolatency makes the x264 encoder go one frame in -> one frame out, with hopefully just a few ms of time spent encoding the frame [22:08:10 CEST] <kepstin> so if you have latency, it's somewhere other than the encoder [22:09:24 CEST] <im_dumb> i think it's all the other cruft we try to do when synchronizing the encoded video and audio [22:09:36 CEST] <im_dumb> but sadly i did not develop the application so i am quite unaware to the internals of it :( [22:09:41 CEST] <im_dumb> but ffmpeg has it beat... easily... [22:09:59 CEST] <kepstin> and the ffmpeg cli tool is designed for batch encoding, not low latency realtime :/ [22:12:36 CEST] <lesshaste> how could I download the audio of https://www.podbean.com/media/player/3ui6b-9412c8-pb?vjs=1 ? [22:14:55 CEST] <kepstin> I'd suggest youtube-dl, but I don't recognize that site and I don't see it in their list. Regardless, the people there have a lot of experience in downloading web video. [22:15:33 CEST] <kepstin> (of course, if it's just a straightforwards playback url or m3u8, you can use that as input to ffmpeg directly and tell ffmpeg to save the audio somewhere) [22:17:47 CEST] <im_dumb> is there anything in the ffmpeg toolkit that is good for low latency/realtime use cases? [22:19:05 CEST] <JEEB> you can configure the framework to low latency, but the defaults are not for that [22:19:24 CEST] <kepstin> there's nothing wrong with using the ffmpeg formats & codecs as libraries in a tool designed for low latency/realtime. Some of the protocols/io stuff even work fine (although you might need to use threads to work around blocking issues with some of them) [22:20:19 CEST] <kepstin> the ffmpeg cli tools is almost completely single-threaded, which is where some of the issues come in, since if anything blocks it'll hold up something else [22:20:46 CEST] <JEEB> ffmpeg.c is just an example client which has with time had a crapload of stuff added to it [22:20:58 CEST] <JEEB> so you can do scary many things with it that the API lets you [22:21:26 CEST] <JEEB> and in theory you could tweak various paramters and make it relatively low latency, but at that point I'd think it would make sense to just start doing your own API client for your own use case [22:25:02 CEST] <lesshaste> kepstin, thanks.. that seemed to work! [22:25:28 CEST] <lesshaste> kepstin, is just straightforward playback but I don't know how to get ffmeg to download it [23:24:45 CEST] <kubast2_> Hey what's a -crf value at which vp9 produces fairlly transparent value? [23:25:00 CEST] <kubast2_> Tried -crf 17 but that was a miss [23:25:25 CEST] <kubast2_> libvpx-vp9 [23:34:08 CEST] <kepstin> dunno, you'd have to try a few values and see [23:34:19 CEST] <kepstin> lower numbers are higher quality, and i think the default for vp9 is 10 [23:37:21 CEST] <kepstin> (the numbers have no relation to the numbers in any other codec, e.g. x264) [23:39:25 CEST] <kubast2_> Yeah I noticed that which is perhaps why I'm asking(besides the fact I couldn't find an answer by googling 'vp9 crf transparent value or vp9 transparent encoding or vp9 transparent encode settings') [23:43:20 CEST] <kepstin> "transparent" is a perceptual thing, so you won't find any agreement on a single number for any codec :) [23:43:46 CEST] <kepstin> well, i suppose you can show a lower bound with abx testing [23:49:06 CEST] <kubast2_> Yeah I sorta did a ghetto abx just ,that I looked at the source after I looked at the encode ,-crf 17 wasn't bad ,would say that on a mobile phone it would look allright [23:52:28 CEST] <kubast2_> as in I looked at the reencode and thought even with 1080P scale down it probablly doesn't look right versus 50GB 60Mb/s 4k hevc source [23:55:32 CEST] <kubast2_> I did wanted to test out a couple of encoders(namelly vp9 and the 2 hevc implementations ,maybe even experimental daala) but I will wait until I get my multi pc encoding setup working [23:56:35 CEST] <kubast2_> I will most likelly set on using dve script but modify it so it accounts for slower pcs when there are less jobs pending than devices avalible [00:00:00 CEST] --- Thu Jun 28 2018

1 0

ffmpeg-devel.log.20180626
by burek 27 Jun '18

27 Jun '18

[00:25:08 CEST] <cone-357> ffmpeg 03Carl Eugen Hoyos 07master:52e8a0d96da7: lavc/libopenjpeg: Support GRAY10, GRAY12 and GRAY14. [02:06:01 CEST] <klaxa> close to having patches ready for the ML for ffserver hls and dash, almost done cleaning up commits and then i'll add documentation [02:43:53 CEST] <rcombs> could lavu hwcontexts expose an API to list available devices and their capabilities [02:45:26 CEST] <rcombs> like, on windows you call GetAdapterCount() and iterate from 1 to that, and the id is just the number; on linux you list /dev/dri/ and filter on renderD[&] [02:47:19 CEST] <atomnuker> sure, I don't see why not [06:18:29 CEST] <cone-093> ffmpeg 03Gyan Doshi 07master:4ac88ba5487e: fftools/ffmpeg: check sseof value and clash with ss [14:49:38 CEST] <gagandeep> kierank: THAT MOUNTAIN SAMPLE is not bit exact with run length code part [14:49:48 CEST] <gagandeep> finally managed to run in testcfhd [14:50:01 CEST] <gagandeep> was reading multithreading in cormen [14:50:22 CEST] <gagandeep> also sent the patch for codebook 0 decompanding previously [14:50:53 CEST] <gagandeep> i now need to start working on multithreading [14:51:05 CEST] <gagandeep> https://github.com/gamer97gagan/FFmpeg [14:51:13 CEST] <gagandeep> this is the fork you asked previously [18:40:18 CEST] <j-b> Speaking of AV1, https://github.com/Matroska-Org/matroska-specification/blob/av1-mappin/code… [18:42:49 CEST] <JEEB> oh I missed the OBU deal [18:42:52 CEST] <JEEB> with AV1 [18:47:58 CEST] <TD-Linux> j-b, oh excellent, was just talking to cyril about making the mkv and mp4 mappings similar [18:48:18 CEST] <j-b> TD-Linux: we (with robux and mosu) are on it. [18:48:34 CEST] <j-b> TD-Linux: encryption is a bit undefined, of course, because I don't care much [18:48:49 CEST] <j-b> But, at least, we have the beginning of a mapping [18:48:52 CEST] <jamrial> so raw OBUs? cool [18:49:03 CEST] <j-b> TD-Linux: I need a volunteer though, for AV1 in AVI [18:49:16 CEST] <gnafu> {:-] [18:49:42 CEST] <JEEB> -28 [18:49:44 CEST] <j-b> :D :D [18:49:47 CEST] <j-b> JEEB: killjoy. [18:49:47 CEST] <TD-Linux> j-b, well at least there are none of those pesky b-frames :^) [18:49:53 CEST] <j-b> TD-Linux: exactly! [18:50:02 CEST] <JEEB> j-b: that was a typo :D [19:01:29 CEST] <atomnuker> JEEB: could you like, dunno, sneak in another timestamp or something [19:01:55 CEST] <atomnuker> or or maybe even a timebase, wouldn't mind that either [19:02:05 CEST] <atomnuker> anything to escape the 1ms precision [19:02:31 CEST] <atomnuker> calculators in the 60's could count more divisions in a second! [19:07:44 CEST] <jamrial> "The AV1 bitstream MUST be in [Annex B] format which is more suitable for 8-bits alignment" [19:07:46 CEST] <jamrial> WTF, no [19:08:05 CEST] <jamrial> i though we agreed annex-b would not be in mp4/mkv or any other container [19:08:08 CEST] <atomnuker> ^^ [19:08:33 CEST] <j-b> jamrial: as you can see, it is in {} which means it's being debated. [19:09:03 CEST] <nevcairiel> just copy paste the mp4 format [19:09:07 CEST] <jamrial> by all means, please don't do it [19:09:08 CEST] <nevcairiel> thats what mkv does usually anyway [19:09:13 CEST] <JEEB> yes [19:09:20 CEST] <jamrial> AV1 annex-b is its own pseudo container thing [19:09:26 CEST] <jamrial> don't put that inside another container [19:09:30 CEST] <nevcairiel> we already got rid of annexb in mp4 :p [19:09:51 CEST] <JEEB> just some details regarding things that are mp4 specific if there's any (14496-15 for HEVC had some cases of that) [19:10:00 CEST] <JEEB> but I guess AV1 didn't have too much of that [19:10:17 CEST] <j-b> jamrial: I will forward your comment. [19:10:35 CEST] <nevcairiel> that "spec" up there is totally overspecified if you ask me, all those color properties are not codec-specific, and nor should they be, for example [19:11:01 CEST] <JEEB> yea, that too [19:11:28 CEST] <JEEB> one would think those would be some generic entity for colorspace information in case the contained format doesn't have that [19:15:37 CEST] <January> AV1 is a container now? [19:16:47 CEST] <January> mmh, I actually haven't read the spec... [19:17:07 CEST] <JEEB> the thing that it can be encapsulated in for stuff without packet structure [19:22:51 CEST] <BBB> I want to see av1-in-ogg-in-avi [19:23:05 CEST] <BBB> if someone can get that working in windows media player, Ill buy him a beer at the next VDD [19:23:12 CEST] <BBB> <3 [19:24:57 CEST] <JEEB> :s [19:25:28 CEST] <gnafu> BBB: I think something like that would call for a stronger drink so the person could forget. [19:25:39 CEST] <JEEB> that's just scary because I can think of ways [19:35:54 CEST] <TD-Linux> I'm happy to add official support to libogg [19:36:45 CEST] <January> gnafu: gonna need more than alcohol to exorcise this [19:37:30 CEST] <BBB> TD-Linux: thats one half [19:37:32 CEST] <TD-Linux> JEEB, those are generic color properties. the problem is that AV1 also can specify them, so this says how to copy them into the container [19:38:00 CEST] <TD-Linux> BBB, sadly I only know of implementations of avi-in-ogg, not the other way around :( [19:38:16 CEST] <BBB> fine, av1-in-avi-in-ogg also qualifies [19:38:31 CEST] <BBB> this is getting kind of frankensteinish, though [19:38:47 CEST] <BBB> but it has to work in windows media player [19:39:25 CEST] <TD-Linux> if we can get a libaom VfW filter and an ogm player we should be set [19:39:43 CEST] <BBB> cool \o/ [19:39:48 CEST] <BBB> creepy, also [19:39:50 CEST] <BBB> but cool [19:42:31 CEST] <JEEB> if it's AVI then you could make the MS filter export the OGG, and then stick a VFW/DShow/MF filter at that that demuxes the resulting OGG into AV1 packets [19:42:36 CEST] <JEEB> and then stick a decoder at that [19:42:58 CEST] <Gramner> y'all need to work in powerpoint somewhere there too [19:43:11 CEST] <JEEB> played in powerpoint of course [19:43:17 CEST] <JEEB> because MS Office and media [19:45:19 CEST] <jamrial> BBB: is one beer worth the pain and missery this would bring to the world? [19:45:41 CEST] <BBB> Gramner: mss1, right? [19:45:51 CEST] <BBB> (mss1 = vc1 with powerpoint sprites, IIRC) [19:45:59 CEST] <JEEB> ah yes :D [19:46:13 CEST] <BBB> the glory days of multimedia [19:46:43 CEST] <BBB> jamrial: theres only one way to answer that question: test it out and see what happens! :D [20:09:47 CEST] <nevcairiel> how would you ever double-pack a container [20:17:24 CEST] <TD-Linux> I thought I at least had docs for ogm and ogg-in-wav but can't find either [20:22:00 CEST] <atomnuker> nevcairiel: you call it by some already taken name like "srt" [21:04:03 CEST] <gnafu> I just taught a PowerPoint class the other day where we looked at embedding audio and video, and 2016 handles media a lot better than PowerPoint used to. [21:04:21 CEST] <gnafu> But I don't know how many formats it actually support these days; they might lock it down to just WMV, MP4, and MOV. [21:04:42 CEST] <gnafu> (Container-level restriction, and then probably only WMV and H.264 for codecs.) [21:04:54 CEST] <gnafu> But I haven't played with it. I should try a VP9+Opus WebM file. [21:05:37 CEST] <nevcairiel> powerpoint just uses your systems directshow setup [21:05:45 CEST] <nevcairiel> so it can play virtually anything if you install codecs [21:08:58 CEST] <gnafu> That's good to know. For some reason, I thought they had artificially restricted it to encourage people to only use common codecs for better portability. [21:09:05 CEST] <gnafu> At least, that's what I recommend :-P. [21:09:25 CEST] <gnafu> Since you don't know what codecs a presentation computer may or may not have, just stick to the most common./ [21:09:50 CEST] <gnafu> Or better, use PDF presentations and separate video files with a portable copy of VLC ;-D. [21:10:59 CEST] <j-b> gnafu: as a joke, I made a PDF plugin for VLC... [21:11:22 CEST] <j-b> gnafu: the sad part is that some big US company is actually using that to display presentations... [21:11:30 CEST] <j-b> >< [21:11:57 CEST] <JEEB> :D [21:12:04 CEST] <JEEB> yes, that is a thing that /will/ happen [21:12:18 CEST] <JEEB> "this is totally a joke" -> "oh, someone's using /that/ in production?!" [21:12:53 CEST] <atomnuker> its alright, we have a .psd decoder, which has actually been quite useful despite big concerns some people had early on [21:13:16 CEST] <atomnuker> and an svg rasterizer as well, which too has been useful to me and quite a few people on the support channel [21:13:18 CEST] <j-b> JEEB: yeah, i was afraid of doing the same for pps. [21:13:26 CEST] <atomnuker> some of these experiments do work out [21:14:02 CEST] <JEEB> I've once used the PSD thing to check wtf those random PSDs on my old disks were [21:14:07 CEST] <JEEB> when I was lazy to poke at GIMP [21:14:35 CEST] <atomnuker> same [21:14:46 CEST] <JEEB> j-b: I still remember looking at the docs for the DVD specs as PDF. > this is a PDF > oh wait you need this acrobat binary plugin to open it [21:15:19 CEST] <atomnuker> does windows still not ship a pdf viewer by default? [21:15:25 CEST] <JEEB> it does now [21:15:39 CEST] <JEEB> or well, for a few years now [21:16:36 CEST] <nevcairiel> yeah its browser just does it now [21:16:36 CEST] <atomnuker> I think 7 only had that .xps or whatever vector format viewer which could save to pdf [21:28:41 CEST] <j-b> libavfilter/dnn_espcn.h | 12637 +++++++++++++++++++++++++++++++++++++ [21:28:44 CEST] <j-b> wow. [22:09:48 CEST] <BBB> j-b: isnt that the neural network filter? [22:09:55 CEST] <BBB> 12k is nothing [22:10:32 CEST] <j-b> Well, 12k of unexplained array... [22:12:21 CEST] <gnafu> j-b: Wait, PDF in VLC? You shouldn't have told me that; I'll have to try it now ;-). [22:13:00 CEST] <j-b> gnafu: I'm afraid I never made this plugin public [22:13:09 CEST] <j-b> the shame would be forever attached to my name, [22:13:10 CEST] <gnafu> Aah, good for you. [22:13:33 CEST] <gnafu> I appreciate that you're a scientist, but not a mad one (at least not on weekdays). [22:14:45 CEST] <gnafu> I use OpenLP to display presentations and videos, and it uses VLC for media. It would've been interesting to just use a VLC playlist for all of it ;-D. [22:18:17 CEST] <j-b> TIL OpenLP [22:22:08 CEST] <nevcairiel> you know thats not what i ever expected OpenLP to be [22:27:16 CEST] <gnafu> I was happy to talk my church into switching to OpenLP instead of paying for a more expensive commercial license for a program we were using on Mac when we moved to a Windows PC. It works really well even just for PowerPoints and videos, but of course also for song lyrics. [22:28:07 CEST] <nevcairiel> i havent been to church in decades, in my mind they are still absent any modern tech [22:28:09 CEST] <JEEB> glad to know there's hackers interested in that niche [22:28:48 CEST] <gnafu> nevcairiel: Hehe, yeah. I have fond memories of overhead projectors and flannelgraphs from my childhood, but "we" have definitely advanced. [22:28:52 CEST] <j-b> Worship is a HUGE market for multimedia [22:29:03 CEST] <nevcairiel> its probably an american thing [22:29:10 CEST] <j-b> There are always large sections at NAB and IBC for that. [22:29:22 CEST] <j-b> very american, yes. [22:29:43 CEST] <gnafu> Now that I've proven the value of OpenLP, I need to talk my church into adding a donation into the budget so the OLP team actually benefits :-P. [22:30:07 CEST] <j-b> Well, often I see people who want to restream cheaply their church [22:30:10 CEST] <gnafu> Since we saved $300-400 compared to the old solution, it's only fair we give them something. [22:30:48 CEST] <gnafu> j-b: Streaming is something we're starting to work out. I've seen churches that have multiple satellite locations, but that seems weird to me. Definitely a market, very America, and kinda weird to me. [22:31:04 CEST] <gnafu> I said "weird to me" twice because of how strongly I feel, not because my brain is tired. [22:31:10 CEST] <gnafu> Or maybe that second thing. [23:05:16 CEST] <cone-589> ffmpeg 03Carl Eugen Hoyos 07master:7becc70375de: lavf/amr: Fix heuristic to avoid detection of repeated bytes. [23:45:30 CEST] <jdarnley> I really should unsubscibe from ffmpeg-user. The message traffic there isn't good for me. [00:00:00 CEST] --- Wed Jun 27 2018

1 0

ffmpeg.log.20180626
by burek 27 Jun '18

27 Jun '18

[00:32:43 CEST] <WeiJunLi> ffmpeg -i input.mkv -pix_fmt yuv420p -f rawvideo - | vpxenc --codec=vp9 --good --cpu-used=2 --threads=8 --row-mt=1 --webm --tune-content=film --end-usage=q --kf-max-dist=250 --bias-pct=100 --cq-level=40 ----width=1920 --height=800 - -o test_vp9.webm [00:32:58 CEST] <WeiJunLi> Im using that pipe but when i try to vlc test_vp9.webm [00:33:15 CEST] <WeiJunLi> I have a `mkv demux error: cannot find KaxSegment or missing mandatory KaxInfo` [00:33:34 CEST] <WeiJunLi> ffmpeg -i input.mkv -pix_fmt yuv420p -f rawvideo - | vpxenc --codec=vp9 --good --cpu-used=2 --threads=8 --row-mt=1 --webm --tune-content=film --end-usage=q --kf-max-dist=250 --bias-pct=100 --cq-level=40 --width=1920 --height=800 - -o test_vp9.webm [00:33:35 CEST] <WeiJunLi> That's the right pipe [00:33:50 CEST] <WeiJunLi> Anyone have an idea of what's missing on the cfg [00:36:37 CEST] <wfbarksdale> does anyone know of a good way to recover an MP4 which has media data (mdat box) but no other meta information? [00:37:14 CEST] <wfbarksdale> I have other videos encoded with the same camera, same frame rate and everything, so seems like i could reconstruct / copy the meta information so salvage? [00:37:46 CEST] <wfbarksdale> interwebs is only turning up this recover_mp4.exe but hoping to find a linux/mac utility [00:46:12 CEST] <Hello71> try prefixing your search with "open source" [00:46:15 CEST] <Hello71> works pretty well [01:50:10 CEST] <Cracki> wfbarksdale, if that files has anything other than the video stream, you might be out of luck [01:50:18 CEST] <Cracki> the interleaving of the container would be lost [01:50:45 CEST] <Cracki> there's a chance you can find patterns in the mdat box that hint at whether it's H.264 or AAC (or whatever codecs were used) [01:51:38 CEST] <furq> there's a oss tool called untrunc [01:51:45 CEST] <furq> but it has some weird dependency on internal libav headers [01:51:50 CEST] <furq> so i can imagine it'll be a pain to build [02:12:31 CEST] <nicolas17> https://trac.ffmpeg.org/wiki/Create%20a%20thumbnail%20image%20every%20X%20s… [02:12:50 CEST] <nicolas17> thumbnails... if I get one image every N seconds, it will have to decode everything, which will be slow, so I want i-frames only [02:13:12 CEST] <nicolas17> if I use the pict_type command shown there, it gets *every* i-frame [02:13:48 CEST] <nicolas17> can I get "one image every N iframes" or "one image roughly every N seconds rounding to the nearest iframe"? [02:18:03 CEST] <nicolas17> okay nevermind :| iframes in this video have terrible quality [04:18:39 CEST] <ChChCherryBom> https://pastebin.com/tjBiNtqj [04:19:07 CEST] <ChChCherryBom> im converting mkv/x264/dts to mp4/ac3/aac but I can't figure out how to specify number of audio channels with mapping like this [04:22:50 CEST] <furq> ChChCherryBom: -ac:a:1 [04:23:58 CEST] <furq> also you can just do `for f` instead of `for f in "$@"` [04:25:23 CEST] <ChChCherryBom> thanks [04:25:31 CEST] <ChChCherryBom> is that still valid for an automator script? [04:25:42 CEST] <ChChCherryBom> im passing in files as arguments not looping through a folder in terminal [04:25:48 CEST] <furq> they do the same thing [04:25:54 CEST] <ChChCherryBom> sweet [04:25:56 CEST] <furq> `for f` will just iterate over all positional arguments [04:30:55 CEST] <ChChCherryBom> does this look valid? https://pastebin.com/U3TxfZX5 [04:31:08 CEST] <ChChCherryBom> oh woops [04:31:16 CEST] <ChChCherryBom> I need to change the audio [04:31:50 CEST] <ChChCherryBom> https://pastebin.com/jpZYyMVA [04:41:42 CEST] <furq> don't put language metadata on the video stream [04:41:57 CEST] <furq> other than that, sure [04:42:54 CEST] <ChChCherryBom> oh makes sense [04:52:21 CEST] <nicolas17> why not? [04:54:23 CEST] <furq> it generally doesn't make much sense [04:54:34 CEST] <furq> but also because ffmpeg -map doesn't let you combine stream type and metadata selectors [04:54:57 CEST] <furq> so if you have a bulk script that does -map 0:v -map 0:m:language:eng then you end up double mapping the video stream [04:55:03 CEST] <furq> that's no fun [04:55:52 CEST] <furq> but it's also just a weird thing to do unless you have multiple video streams [05:13:09 CEST] <ChChCherryBom> one last question I think, I haven't been able to find any info on this [05:13:24 CEST] <ChChCherryBom> when converting subtitles from say ass/srt to mov_text, how do I copy the forced flag [05:14:59 CEST] <ChChCherryBom> if there is one [05:21:54 CEST] <crazybit> vidstabdetect: is there any reason to use accuracy < 15 if video processing speed is not an issue? [05:27:34 CEST] <crazybit> also, what is a "local measurement field" that's mentioned in docs vidstabdetect -> mincontrast? [06:06:05 CEST] <nicolas17> how can I split a video into multiple segments? [06:06:43 CEST] <nicolas17> not for hls/dash streaming... I just need to split the video [06:54:17 CEST] <Cracki> nicolas17, -ss and -t [06:54:39 CEST] <Cracki> behave differently in front and after -i [06:55:07 CEST] <Cracki> be aware that "-c copy" would only be able to cut at keyframes [06:57:14 CEST] <nicolas17> I found the 'segment' muxer, seems to work well [13:41:47 CEST] <Blacker47> hello. i want apply some filters (like deinterlace and crop) to a few videos and safe relults as y4m and then concatenate them (later with a few still-image-snippets between) to a big file (and then encode it). but ffmpeg seems not to support concatenate y4m files - any ideas how I should do this? [13:56:11 CEST] <anyvision> Hey guys, I'm looking for a way to copy an rtsp's stream original pts (if such information exists). I've tried to use -copyts and -ts mono2abs but to no avail (pts appears to be relative rather than absolute). [14:09:01 CEST] <anyvision> Hey guys, I'm looking for a way to get an rtsp stream's original (absolute) pts. I've tried -copyts & -ts mono2abs but to no avail. I'm not sure this information even exists but if it does, how can I get it? thanks [15:30:00 CEST] <konnex> hey guys, just wondering if you can use expressions like random(x) in the output filename? doesn't seem to work for me [15:30:48 CEST] <BtbN> Use your shell for stuff like that. [15:31:40 CEST] <konnex> hmm okay, just thought it would be possible, but thanks anyways =) [15:36:07 CEST] <konnex> but what about this: I am outputting to hls and want the filenames to be uuids and not just counting up... any way to do this? [15:54:29 CEST] <Celmor> does ffmpeg behave differently on SIGINT (Ctrl+C) than pressing q? [15:54:48 CEST] <Celmor> using former I get "Exiting normally, received signal 2." [17:00:21 CEST] <King_DuckZ> hi, I'm porting some old code that is adding some options into a dictionary - options like sc_threshold, coder, 8x8dct, b_strategy, b-pyramid... is there a list of all such options with their valid ranges and the default values? [17:15:27 CEST] <furq> King_DuckZ: https://gitlab.com/qruf/x264settings/blob/master/settings.md [17:15:56 CEST] <furq> and http://www.chaneru.com/Roku/HLS/X264_Settings.htm [17:16:02 CEST] <furq> i think that's the most up-to-date one [17:21:50 CEST] <King_DuckZ> furq: I need to know what the default would be tho, like if I didn't put b-pyramid in the dictionary I pass to avcodec_open2, what would it default to behind the scenes? [17:23:02 CEST] <King_DuckZ> unless, is there a particular column I should be looking at, in your link? [17:23:47 CEST] <Fenrirthviti> default is listed for every option in the second link [17:24:15 CEST] <King_DuckZ> ahh sorry, it looked like one long link [17:24:25 CEST] <Fenrirthviti> second link is probably better to use [17:26:30 CEST] <King_DuckZ> it looks very useful, thanks! [17:27:20 CEST] <King_DuckZ> btw when the second link says preset's default is medium, it's the "medium" column from the first link that applies, correct? [17:34:46 CEST] <furq> right [17:49:31 CEST] <saml> is nut file good? [17:50:57 CEST] <atomnuker> sehr gut [18:10:40 CEST] <King_DuckZ> when I do av_dict_get(my_dict, blah blah); will that automatically create the dictionary if my_dict is null? [18:11:32 CEST] <King_DuckZ> or av_dict_set() does it maybe? [18:29:32 CEST] <crazybit> besides ffmpeg source code and official docs, does anyone have a source for a more detailed explanation of vidstab filters? [18:29:33 CEST] <acos> Are USB devices supported for video capture? [18:31:42 CEST] <crazybit> acos: $ ffmpeg -devices [18:33:11 CEST] <crazybit> acos: and yes you can, at least if you're on Linux [18:34:55 CEST] <acos> Coom. trying to capture old videos. Almost got a pcie device [18:35:28 CEST] <crazybit> you just need to know the /dev entry [18:35:46 CEST] <bencoh> .59 [18:35:50 CEST] <bencoh> (woops) [18:35:50 CEST] <crazybit> ffmpeg -i /dev/whatever [18:35:59 CEST] <crazybit> beauty of Linux [18:36:41 CEST] <crazybit> AFAIK you do have to tell ffmpeg what format it is though, so you would add -input_format mjpeg or whatever [18:37:57 CEST] <crazybit> ^ actually i'm not sure that's required, maybe only for some types of hardware i don't know [18:39:44 CEST] <acos> Hmmm its older [18:39:50 CEST] <acos> Its not that good hardware. [18:40:14 CEST] <acos> Might have to inline deinterlace [18:48:04 CEST] <acos> What linux distro is best for ffmpeg? Can I live boot USB ubuntu? [18:50:24 CEST] <crazybit> if you want the newest features, use a rolling release distro since it'll have the newest release of ffmpeg... if you don't care, it does not matter much so long as you don't use some outdated distro version [18:50:34 CEST] <tdr> acos running it from usb would work, but you'd need to find one that has it. [18:50:45 CEST] <acos> Just want to capture like 60 tapes [18:51:04 CEST] <crazybit> acos: don't use a usb live distro if you wan't to also use usb for capture [18:51:17 CEST] <acos> Oh :( hmmm. I got more USB ports [18:51:45 CEST] <crazybit> well ok [18:52:10 CEST] <acos> I guess I can dial boot on this laptop [18:52:24 CEST] <kepstin> just make sure the storage you're using and the capture device aren't on ports that share bandwidth (which is sometimes hard to tell) [18:52:52 CEST] <acos> Well it will be a sata data disk [18:52:58 CEST] <crazybit> acos: better yet, if you do boot distro via usb, boot it to memory if you have more than enough that way it won't use usb [18:53:03 CEST] <crazybit> ah [18:53:16 CEST] <acos> Hmmm 16gb ram in that laptop [18:53:26 CEST] <kepstin> yeah, shouldn't be a problem if you're writing to a sata disk [18:55:35 CEST] <acos> Hope so [18:55:45 CEST] <acos> Not sure what format to save the video [18:56:01 CEST] <crazybit> mp4 probably [18:56:09 CEST] <acos> Kinda dont want h264 [18:58:20 CEST] <kepstin> don't use mp4 for live captures, it's unreadable if there's a crash during capture [18:58:30 CEST] <kepstin> (mkv is good, you can always remux later) [19:06:31 CEST] <acos> What's the data rate [19:07:24 CEST] <kepstin> whatever you set it to when encoding [19:07:28 CEST] <acos> Hmmm [19:07:32 CEST] <acos> Not sure. [19:07:47 CEST] <acos> What's datarate of uncompressed again? [19:07:54 CEST] <kepstin> way too high [19:08:10 CEST] <acos> Hmm. Ya not sure what settings to choose for this then [19:08:13 CEST] <kepstin> well, if you're not sure, a good option is to use libx264 encoder with "-crf" mode, setting a value that gives you acceptable quality. [19:08:46 CEST] <kepstin> (the default crf is 23, most folks think 18 is very high quality. lower numbers are higher quality) [19:09:44 CEST] <acos> But I want to re edit this later. Would hate generation loss of exporting h264. [19:10:29 CEST] <kepstin> if you really want to avoid generation loss, consider using a lossless codec, e.g. ffv1 or x264 in lossless mode (option "-qp 0") [19:12:21 CEST] <acos> Good idea. Wish I could use dnhx [19:12:34 CEST] <acos> I hear those are good intemrdite codecs [19:13:03 CEST] <kepstin> most intermediate codecs are actually lossy [19:13:27 CEST] <kepstin> they're usually designed as I-frame only (or very small gop) to make non-linear editing/cutting easier [19:14:34 CEST] <kepstin> (i mean, not all, and some have both lossy and lossless modes) [20:07:04 CEST] <furq> 18:07:47 ( acos) What's datarate of uncompressed again? [20:07:22 CEST] <furq> acos: for reference, (bpp * width * height * fps) / 1024 [20:07:57 CEST] <furq> and probably / 1024 again because this is going to be in megabits per second [20:08:13 CEST] <JEEB> bits are generally calculated in / 1000 [20:08:20 CEST] <JEEB> in video encoders etc [20:08:25 CEST] <JEEB> so a kilobit is 1000 bits [20:08:31 CEST] <furq> damn SI [20:08:45 CEST] <JEEB> of course after you divide by eight and make into bytes, then you go into 1024 area again [20:09:01 CEST] <bencoh> :) [20:09:11 CEST] <furq> also you can find the bpp for your pixel format in ffmpeg -pix_fmts [20:09:37 CEST] <acos> Hmmm thanks you smart [20:09:56 CEST] <acos> Not sure what reisltuion vhs video tapes are 420x240? [20:10:05 CEST] <furq> depends on your capture device [20:13:26 CEST] <acos> It's some old USB thing [20:13:38 CEST] <acos> With the yellow red white rca [20:13:54 CEST] <im_dumb> SI is the worst thing since the imperial system [20:16:06 CEST] <bencoh> I'd expect it to depend on pal/ntsc as well [20:16:51 CEST] <acos> Ntsc [20:16:55 CEST] <acos> Not in palland [20:26:38 CEST] <ChChCherryBom> why does the duration of tracks change when I change the container from mkv to mp4? [20:27:32 CEST] <kerio> with -c:v copy -c:a copy? [20:30:22 CEST] <kepstin> acos: most capture devices will use the standard/typical size for the video standard, so you'll probably get an interlaced 720x480 signal from that. [20:33:41 CEST] <ChChCherryBom> yes [20:34:31 CEST] <ChChCherryBom> -c:v copy changes from 2:22:27:873 to 2:22:27:872 chapters go from :873 to :831 [20:36:13 CEST] <acos> Hmmm kepstin thanks [20:36:30 CEST] <ChChCherryBom> subtitles go from :873 to 0:06:01:347 (being converted to mov_text, but that length is weird) [20:40:28 CEST] <furq> .873 to .872 is probably just because of a different timebase [20:40:44 CEST] <furq> not sure about the other stuff though [20:41:16 CEST] <furq> are the subtitles actually desynced [20:43:37 CEST] <ChChCherryBom> I need to try it on another set to see, I can't find the forced subs part [21:04:29 CEST] <Hello71> VHS is analog though, isn't it? so theoretically you can get higher resolution than NTSC/PAL [21:05:28 CEST] <Hello71> like, it doesn't technically have horizontal pixels, only lines [21:17:45 CEST] <acos> Hmmm cool [23:43:53 CEST] <Ars> Hi guys, pls help me. [23:43:53 CEST] <Ars> I am transcoding and multicasting stream like: fmpeg RTMP_Stream udp://225.2.1.11:1300 [23:43:53 CEST] <Ars> Also i have stream check script (ffprobe udp://225.2.1.11:1300), but when executing that command ffmpeg stops. [23:43:53 CEST] <Ars> I understand there is port binding problem. [23:43:53 CEST] <Ars> But how i can check that UDP stream alive or no? How to chek stream on udp://225.2.1.11:1300 ? [00:00:00 CEST] --- Wed Jun 27 2018

1 0

ffmpeg-devel.log.20180625
by burek 26 Jun '18

26 Jun '18

[00:06:43 CEST] <c3r1c3-Win> I have not heard ANYTHING about them moving to h.264. Given the recet release of their 3.5 SDK I would be greatly surprised if they did. [00:09:31 CEST] <c3r1c3-Win> Also there's not really much to reverse engineer. They're following certain standards (Zeroconf+SpeedHQ+TCP/IP+Some others) so there really isn't much (if anything) to reverse engineer. [00:13:33 CEST] <kierank> they have some homemade encryption [00:13:37 CEST] <kierank> and framing [02:12:38 CEST] <cone-039> ffmpeg 03Michael Niedermayer 07master:7719b8ccc790: avcodec/magicyuv: Check bits left in flags&1 branch [02:12:38 CEST] <cone-039> ffmpeg 03Michael Niedermayer 07master:d25c94524797: avcodec/h264_parser: Reduce needed history for parsing mb index [02:31:02 CEST] <kierank> This fixes a bug/regression with very small packets [02:31:02 CEST] <kierank> Fixes: output_file [02:31:03 CEST] <kierank> wow [02:31:07 CEST] <kierank> such helpful [08:08:04 CEST] <cone-043> ffmpeg 03Steven Liu 07master:387464bea5bd: from RTCTIME to N*23, step is 23 fix ticket: 7225 [10:51:37 CEST] <haasn> atomnuker: is it intended that your ffmpeg vulkan filter code stalls the pipeline on every upload? [10:52:18 CEST] <haasn> also if that's your intent, you're probably better off not bothering with the COMPUTE/TRANSFER qfs and just using GRAPHICS for the upload always [11:06:53 CEST] <atomnuker> stalls the queue? as in waiting for the fence [11:08:08 CEST] <haasn> yeah [11:08:16 CEST] <haasn> you dispatch an upload and then wait for it to complete before doing anything else [11:08:24 CEST] <haasn> during which time the GPU could be doing useful work [11:08:39 CEST] <haasn> also the transfer queues (DMA engine) are generally slower than uploading directly from the graphics queue [11:08:52 CEST] <haasn> they only exist _because_ they can run at the same time as the GPU is doing work [11:09:19 CEST] <haasn> you also stall on the download, so it's pretty much the worst case scenario [11:09:29 CEST] <haasn> I think you can even run the upload and the download simultaneously, in theory [11:09:42 CEST] <atomnuker> you can't not stall on download, can you? [11:09:45 CEST] <haasn> so you can download frame 0, filter frame 1 and upload frame 2 at the same time [11:09:47 CEST] <haasn> sure you can [11:10:00 CEST] <haasn> dispatch the download ahead of time, then do useful work in the meantime [11:10:07 CEST] <haasn> well maybe not for ffmpeg's filter design [11:10:23 CEST] <haasn> if ffmpeg expects filters to produce output in lockstep before it will give you more frames then you have to stall on the download [11:10:53 CEST] <haasn> in e.g. mpv's filter chain you can receive multiple frames before producing one frame of output [11:11:44 CEST] <atomnuker> you can do that here too, but there's no way to inform other parts of the code when a frame's ready, its always assume to be completed by the time its given off to non-hardware filters [11:11:56 CEST] <atomnuker> *assumed [11:14:19 CEST] <haasn> right, you'd have to maintain your own download queue [11:14:24 CEST] <haasn> and only hand them off once the fence passes [11:14:34 CEST] <haasn> blocking in case the queue would get too long [11:15:32 CEST] <haasn> it depends on the filter though [11:16:09 CEST] <haasn> for a filter that's primarily bottlenecked on PCIe bandwidth (e.g. RGB->YUV conversion or something simple like that) the difference will probably be moot [11:16:22 CEST] <haasn> since the amount of "useful work" you're missing out on amounts to a few µs at best per frame [11:16:42 CEST] <haasn> but for something like a more involved compute shader you'd definitely be wasting a lot of GPU power by waiting on transfers [11:37:14 CEST] <atomnuker> wouldn't be that much though, you have to block somewhere, at most you can move the blocking to the next time you submit a command queue for execution [11:43:42 CEST] <haasn> atomnuker: blocking and pipeline stalls are not the same thing [11:44:14 CEST] <haasn> pipeline stall = blocking when you have too little work queued for the GPU to be fully loaded [11:44:50 CEST] <atomnuker> oh, right [11:47:40 CEST] <atomnuker> haasn: so the correct way to deal with this would be to use per-frame semaphores on which to wait/signal during queue submissions, and only use a single fence at the end when you download (because there's no other choice here) [12:49:12 CEST] <haasn> atomnuker: yeah, or events (if they're on the same queue family) [12:49:20 CEST] <haasn> s/queue family/queue/ [12:49:35 CEST] <haasn> although in your case just a pipeline barrier would be enough in the single-queue case [12:52:05 CEST] <atomnuker> events? but you can't submit anything other than semaphores in VkSubmitInfo [12:52:48 CEST] <atomnuker> oh, you'd do that via a command [12:53:45 CEST] <atomnuker> semaphores would be cleaner here, we don't have complex multistage commands, just usually a single shader [12:54:18 CEST] <atomnuker> btw did moltenvk get support for events? [12:55:49 CEST] <haasn> I mean events are cheaper than semaphores [12:55:51 CEST] <haasn> and no [12:55:59 CEST] <haasn> semaphores require splitting your stuff into separate command buffers [12:56:09 CEST] <haasn> in fact I'd argue that semaphores are precisely what you need for "complex multistage commands" [12:56:18 CEST] <haasn> oh wait you mean -within- a buffer [12:56:21 CEST] <haasn> yeah fine [13:43:48 CEST] <atomnuker> yeah, each filter has its own command queue [13:44:23 CEST] <atomnuker> didn't know events were cheaper then semaphores, I though it would be the other way around [13:45:05 CEST] <atomnuker> since events are more flexible and can occur at any point in any command buffer [13:57:44 CEST] <haasn> command buffer scheduling is pretty expensive [13:57:48 CEST] <haasn> cheaper to have one big command buffer [13:57:53 CEST] <haasn> than many little small ones [20:51:36 CEST] <jamrial> whoa, libaom 1.0.0 [20:53:21 CEST] <JEEB> yup [20:53:39 CEST] <jamrial> is the library stable enough for this? [20:54:16 CEST] <jamrial> one thing is codec bitstream freeze, another is the reference encoder being stable for such a release [20:54:26 CEST] <JEEB> that's a good question [20:54:42 CEST] <BBB> were there stability issueS? [20:54:56 CEST] <jamrial> no idea, just wondering [20:55:21 CEST] <jamrial> i at least see a couple open issues reporting assertion failures and valgrind stuff [20:56:15 CEST] <jamrial> i'm going to bump the required version for the wrapper so we stop creating broken files, but will not remove the experimental flag for now [20:57:10 CEST] <atomnuker> BBB: MAKE [20:58:17 CEST] <BBB> <3 [20:59:14 CEST] <gnafu> :-o :-D [21:02:49 CEST] <January> atomnuker: rav1e tho [21:10:24 CEST] <atomnuker> decoder, not encoder [21:40:46 CEST] <thardin> ffav1 or xav1? [21:41:16 CEST] <gnafu> thardin: ffav1 for decoding, rav1e for encoding ;-). [21:41:32 CEST] <gnafu> Or EVE, I suppose. Right, BBB? [21:44:03 CEST] <January> atomnuker: time for rav1d then [21:44:52 CEST] <nevcairiel> thats what we call ffav1 =p [21:45:37 CEST] <January> nevcairiel: but ffav1 isnt rust [21:45:48 CEST] <Compn> our next codec needs to be uck... :P [21:45:56 CEST] <nevcairiel> January: and thats a bonus [21:46:06 CEST] <January> :d [21:53:16 CEST] <cone-357> ffmpeg 03James Almer 07master:a0ac49e38ee1: configure: bump minimum required version of libaom [21:53:31 CEST] <j-b> Can we do it in JS and then transpile it to C? :D [21:53:44 CEST] <haasn> ffav1 is dangerously close to ffv1 [21:54:25 CEST] <gnafu> j-b: I don't see why not. [21:54:43 CEST] <gnafu> I suspect it would get a lot more attention on HN than a regular ol' C version. [21:54:47 CEST] <nevcairiel> ffav1 is not actually a name we use [21:54:53 CEST] <nevcairiel> it would just be called "av1" [21:55:02 CEST] <j-b> to be sure to confuse it with av1 [21:55:05 CEST] <j-b> avi [22:01:29 CEST] <atomnuker> January: its not a bad thing; rust sucks [22:02:02 CEST] <j-b> Is the MP4/AV1 spec frozen or not yet? [22:02:54 CEST] <atomnuker> ff<codec> is just something people use to refer to ffmpeg implementations of codecs, I'm not aware of anyone using it for containers, let alone av1 [22:03:00 CEST] <atomnuker> erm *avi [22:47:43 CEST] <cone-357> ffmpeg 03Mark Thompson 07master:bd02b2c05df1: lavfi/vf_tonemap_opencl: Mark local tables static [22:47:44 CEST] <cone-357> ffmpeg 03Mark Thompson 07master:e7c0b44e747b: lavf/mpjpegdec: Mark local variable static [23:08:43 CEST] <cone-357> ffmpeg 03Michael Niedermayer 07master:6677c9862648: avcodec/escape124: Check buf_size against num_superblocks [23:08:44 CEST] <cone-357> ffmpeg 03Michael Niedermayer 07master:84bbdc255a12: avcodec/ffv1dec: Check state transition table [23:08:45 CEST] <cone-357> ffmpeg 03Michael Niedermayer 07master:78167b498f53: avcodec/cscd: Check output buffer size for lzo. [00:00:00 CEST] --- Tue Jun 26 2018

1 0

ffmpeg.log.20180625
by burek 26 Jun '18

26 Jun '18

[00:30:02 CEST] <teratorn> asked this a couple times so sorry if I missed any answers... [00:30:11 CEST] <teratorn> any tips of tricks to profile an ffmpeg filter-graph execution? [00:30:15 CEST] <teratorn> s/of/or/ [01:27:50 CEST] <kepstin> teratorn: what are you trying to accomplish that makes you think profiling an ffmpeg filter-graphi is the solution? [01:30:23 CEST] <teratorn> I want to sanity-check my particular solution that combines multiple videos, still images and various effects using a fairly complicated filter-graph... just want to make sure I'm not screwing up badly somewhere and making it much slower than a straight transcode [01:31:04 CEST] <Cracki> >much slower than a straight transcode [01:31:10 CEST] <Cracki> they're filters. they cost something. [01:31:28 CEST] <teratorn> knowing how much cpu time each node takes would give me some insight [01:31:30 CEST] <Cracki> and you use them because you have to. [01:31:42 CEST] <teratorn> sure, I get that [01:31:45 CEST] <Cracki> yes true, I understand the desire to see which costs what [01:32:20 CEST] <Cracki> I have no clue, but perhaps looking at the threads and their call stacks could be insightful. [01:32:31 CEST] <Cracki> that is, if filter graphs are modeled as a thread per filter... [01:33:14 CEST] <teratorn> could be.. I'm not sure what graph execution model is [01:33:21 CEST] <teratorn> *what the [01:37:09 CEST] <kepstin> they're not really modeled like that. for the most part, ffmpeg filters are all single-threaded and run in the thread of whatever created/calls into the filter graph [01:38:34 CEST] <teratorn> I'm using the front-end [01:40:11 CEST] <teratorn> I guess I can just time a transcode of my output, vs how long it took my invocation of ffmpeg to produce that output [01:40:59 CEST] <kepstin> anyways, ffmpeg doesn't contain any code for measuring execution time per filter instance. Profiling it with a regular code profiling tool would be tricky, particularly if you have multiple instances of a filter. [01:42:25 CEST] <kepstin> something that might actually be useful is running ffmpeg with -v verbose and make sure that you're not getting a lot of auto-inserted scale filters for pixel format conversions between filters. [01:43:14 CEST] <kepstin> other things to keep an eye out is to avoid using overlay filters when you can use hstack or vstack. [01:44:44 CEST] <kepstin> other than that, presumably you put in a filter because you want to use it. So if it makes things slow, well, :/ [01:54:49 CEST] <furq> teratorn: run it with a null output [01:54:56 CEST] <furq> -f null - [01:55:32 CEST] <furq> it's not really a great profiling tool but it will at least tell you how fast the full filterchain processing is [01:55:37 CEST] <teratorn> kepstin: ok, I /am/ doing some overlay filters [01:56:10 CEST] <teratorn> this is what I mean... yeah as far as /I know/ i'm not using any filters I don't have to.. but maybe there are other ways to skin the same cat [01:56:31 CEST] <teratorn> furq: good tips, thanks [01:56:41 CEST] <furq> if you're using overlay to put videos side-by-side then that's wrong [01:56:46 CEST] <furq> otherwise it's probably fine [01:57:02 CEST] <teratorn> ah, cool [01:57:18 CEST] <furq> just pastebin the filterchain if you want people to look over it [02:05:28 CEST] <teratorn> sure if you're curious, I'm doing something like this: https://hastebin.com/raw/tahuyeyupe which produces output something like this: https://docs.zoho.com/file/0cpqed0d0433728d14865b43654a09eed8059 [02:29:01 CEST] <Cracki> that is one huge complicated filter [02:30:28 CEST] <Cracki> maybe you want to use a video editing program for that [02:40:42 CEST] <furq> you know you can have spaces and newlines in the filterchain, right [02:40:46 CEST] <furq> this seems like a good time to make use of that [03:01:30 CEST] <teratorn> Cracki, furq: it's programmatically generated to produce effects like in the sample output I pasted :) [03:01:40 CEST] <teratorn> furq: no I wasn't aware of that; noted [03:04:00 CEST] <kepstin> for large programmatic filters you can also write them to a separate file then use the -filter_complex_script option. That avoids command line length issues, and sometimes makes it easier to read/debug. [03:19:09 CEST] <teratorn> kepstin: cool beans [05:13:26 CEST] <galex-713_> hi [05:13:37 CEST] <galex-713_> is stereo3d filter interleaved mode notably broken? [05:13:44 CEST] <galex-713_> because Ive the impression it is [05:13:52 CEST] <galex-713_> I just tested bino, and it worked where mpv didnt [06:37:19 CEST] <TheAMM> nvenc_h264 takes a small moment to warm up, and on realtime sources this results in a small pause/lag at the start of the output video [06:37:40 CEST] <TheAMM> Is there some way I could initialize the encoder without using the first output frame? [06:38:40 CEST] <TheAMM> I'm not at the machine to test it right now, but would vf trim work, or do I have to go wild with nut-piping into another ffmpeg that would drop the first (premature) frame? [06:40:14 CEST] <TheAMM> Actually nevermind, that won't work, since I can't drop the frame without re-encoding which defeats the purpose of using nvenc for realtime encoding [08:29:32 CEST] <pragomer> I have a video file called webm.dat7 any chance to convert this to something readable? "mediainfo" tells me nothing about that file [08:34:11 CEST] <Cracki> where is it from [08:34:27 CEST] <Cracki> be forthcoming with relevant info [08:34:39 CEST] <Cracki> or else it's the spanish inquisition for you [08:35:58 CEST] <pragomer> LOL ... its a webvideo in chromes browser cache... [08:37:33 CEST] <Cracki> share it if you want. or open it with a hex editor and show the first page [08:37:48 CEST] <Cracki> ffprobe it [08:38:23 CEST] <pragomer> mm. unfort. I cannot share it as it is part of an investigation in our company :-(... but I surely can post you the first hex-view page [08:38:30 CEST] <pragomer> just a moment (and thank you already) [08:39:52 CEST] <pragomer> hex-view [08:39:53 CEST] <pragomer> https://snag.gy/f6oJ3A.jpg [08:40:02 CEST] <pragomer> I know.. this doesnt look like any video format [08:41:02 CEST] <Cracki> indeed. could be a fragment. [08:41:11 CEST] <Cracki> I'd expect dat6 and dat8 to exist? [08:43:37 CEST] <Cracki> take the file size, represent in hex, see if you can find this (or the reverse) in the hex view [08:43:50 CEST] <Cracki> the first 1-2 lines look not-very-random [08:44:08 CEST] <pragomer> I think too, that it is just a fragment.. so... I think it wont be worth the efforts. [08:44:15 CEST] <Cracki> potentially some kind of header. that's all I would hope to get out of that file alone. [08:44:16 CEST] <pragomer> but thank you in any way for your support [08:45:25 CEST] <Cracki> there might be ways to reassemble that browser cache [08:54:10 CEST] <furq> pragomer: does ffprobe give anything useful [08:55:11 CEST] <pragomer> webm: Invalid data found when processing input [08:56:05 CEST] <pragomer> ps: originally the file has no extension, just "webm"... the dat7 extension was "defined" from our forensics tool "x-ways" [11:35:13 CEST] <Celmor> "No such filter: 'zscale'" meh [11:43:34 CEST] <haystack> Hi, I would like to share my screen using ffmpeg but i'm not able to host the streaming server within this company. Has anyone solved this problem already? Or can recommend a service that accept encrypted ffmpeg streams? [11:45:00 CEST] <JEEB> Celmor: yea it bases on a separate library [11:45:00 CEST] <JEEB> zimg [11:45:05 CEST] <JEEB> https://github.com/sekrit-twc/zimg/ [11:50:31 CEST] <Celmor> which I need to first "install"? [11:50:49 CEST] <JEEB> yes, then you build FFmpeg with it [11:51:01 CEST] <Celmor> according to my package manager I have zimg installed, does ffmpeg not recognize that?> [11:52:11 CEST] <JEEB> if it was not built with it, then it will not have it [11:52:23 CEST] <JEEB> since the module only gets built if it was enabled during build time [13:51:59 CEST] <YokoBR> hi there [13:52:14 CEST] <YokoBR> please, how can I get a remote m3u8 (hls) and re-stream it? [13:52:37 CEST] <YokoBR> i've tried -c copy but it didn't worked [13:53:09 CEST] <YokoBR> I want to receive an hls stream and push it to a rtmp server [13:59:41 CEST] <kepstin> assuming the codecs are compatible, i'd expect that to work. [14:04:26 CEST] <YokoBR> https://pastebin.com/4KA6CFwm [14:04:31 CEST] <YokoBR> I've done this so far [14:05:14 CEST] <ozette> can ffmpeg follow that link? [14:08:12 CEST] <ariyasu> don't need the "ffmpeg -i ffmpeg -i" just on of them [14:08:27 CEST] <ariyasu> other than that it looks ok i guess, can you paste bin the console output [14:08:39 CEST] <ariyasu> one* [14:11:46 CEST] <YokoBR> ariyasu: it's only one of them at once [14:11:55 CEST] <YokoBR> but I can't forward it yet [14:25:09 CEST] <Franciman> Hi [14:25:23 CEST] <Franciman> av_frame_get_best_effort_timestamp is now deprecated, what should I use instead? [15:27:48 CEST] <King_DuckZ> hi, I'm porting some old code that does something based on a codec id, eg if (codec_id == AV_CODEC_ID_DNXHD) { //blah }, but in my new code I only have the codec string name and no direct access to the ffmpeg library to convert it to a codec_id [15:28:26 CEST] <King_DuckZ> my question is is it safe to do the same if over the string name? is there anything I should watch out for? [15:30:27 CEST] <andreas> Hi, I want to make a slideshow from a number of images (AMB_00001.gif, AMB_00002.gif, ...). When using this command: [15:30:27 CEST] <andreas> ffmpeg -i AMB_%05d.gif output.mp4 [15:30:27 CEST] <andreas> I get the following output: [15:30:27 CEST] <andreas> ffmpeg version 2.8.14 Copyright (c) 2000-2018 the FFmpeg developers [15:30:27 CEST] <andreas> built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-16) [15:30:28 CEST] <JEEB> just keep in mind that there can be multiple decoders or encoders for a format, although their names tend to be different :) [15:30:29 CEST] <andreas> configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-versio [15:30:34 CEST] <andreas> n3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11g [15:30:41 CEST] <andreas> rab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect [15:30:44 CEST] <andreas> libavutil 54. 31.100 / 54. 31.100 [15:30:46 CEST] <andreas> libavcodec 56. 60.100 / 56. 60.100 [15:30:48 CEST] <andreas> libavformat 56. 40.101 / 56. 40.101 [15:30:50 CEST] <andreas> libavdevice 56. 4.100 / 56. 4.100 [15:30:52 CEST] <andreas> libavfilter 5. 40.101 / 5. 40.101 [15:30:54 CEST] <andreas> libavresample 2. 1. 0 / 2. 1. 0 [15:30:56 CEST] <andreas> libswscale 3. 1.101 / 3. 1.101 [15:30:58 CEST] <andreas> libswresample 1. 2.101 / 1. 2.101 [15:31:00 CEST] <andreas> libpostproc 53. 3.100 / 53. 3.100 [15:31:02 CEST] <andreas> *.gif: No such file or directory [15:36:28 CEST] <King_DuckZ> geez why are you spamming like that [15:38:37 CEST] <andreas> realized that it would be annoying when I saw it myself, sry [15:40:05 CEST] <JEEB> /33 [16:01:32 CEST] <King_DuckZ> I'm looking at how find_codec_by_name() is implemented, it seems to be just looping over a list and stopping at the first match, so I should be ok right? [16:01:53 CEST] <King_DuckZ> I can't really find where that list comes from, but I don't think it matters at this point [16:19:52 CEST] <King_DuckZ> anyone who can help with my question please? [16:20:22 CEST] <JEEB> I actualyl replied in the middle of that spam by that other guy [16:21:02 CEST] <King_DuckZ> JEEB: ah I didn't see that, it got lost in the spam sorry! [16:39:18 CEST] <King_DuckZ> idk if this is correct but I ran this: ffmpeg -encoders | ag 'V.{5}\s' | sed -E 's/^\s+//' | sed -E 's/\s+/\t/g' | cut -sf2 | sort | column [16:40:02 CEST] <King_DuckZ> and I do see several codecs with the same name [16:40:05 CEST] <King_DuckZ> or maybe not :s [16:42:19 CEST] <King_DuckZ> uhm not sure what I did earlier but I can't see any duplicates right now [16:44:14 CEST] <JEEB> yea the names shouldn't have any [16:48:47 CEST] <King_DuckZ> JEEB: I'm confused... that commands prints h264_nvenc h264_omx h264__v4l2m2m h264_vaapi, would the codec_id be the same for all them? is that what you meant? [16:49:11 CEST] <JEEB> yes, that is what I meant. that if you ask for a decoder or encoder for a codec id, you might get something you don't expect [16:49:25 CEST] <King_DuckZ> dammit [16:52:25 CEST] <King_DuckZ> my problem is that I have this case: my_program -> my_lib -> ffmpeg, and ffmpeg is private to my_lib, but this logic has to be inside my_lib [16:52:42 CEST] <King_DuckZ> so I can't access any ffmpeg function from there, not even the AVCodecID type [17:11:13 CEST] <tuna> Hello again, I previously was here asking about hardware encoding...that works great now, however I'd like to implement hardware decoding now. My question is where should I start with the decoding...I have a software decoder setup and running...will I need a hardware frame context like with the encoder? [17:17:48 CEST] <atomnuker> yes, it should be identical as the encoder contexts [17:19:14 CEST] <tuna> So, I should be able to essentially copy paste that part (maybe some minor changes like size/format) but then the hardware frame that I alloc with ffmpeg, I can then expect to be filled by the nvdec? [17:46:01 CEST] <Celmor> if there were only an ffmpeg version downloadable with zimg any other optional libs included, tried looking for docker images for ffmpeg but they don't list zimg as far as I can see [17:46:20 CEST] <Celmor> recompiling just for library support is a hassle [18:01:10 CEST] <Hello71> install gentoo [18:04:25 CEST] <Celmor> arch is hard-core enough for me [19:17:41 CEST] <Celmor> can I interrupt a conversion, play the output file to see if it's as I want it and then continue the conversion afterward? [19:18:41 CEST] <klaxa> in some shells ctrl+z puts a process in pause, depending on your output format you can just play the output, that wouldn't require pausing the encode though [19:19:03 CEST] <klaxa> you can put the process back in the foregrund by running the "fg" command [19:19:06 CEST] <Celmor> yeah, I know about that, but often the output isn't playable at that point [19:19:34 CEST] <klaxa> like i said, depends on the format, e.g. mp4 is usually not playable until it is fully written [19:19:51 CEST] <Celmor> usually using mkv [19:20:16 CEST] <klaxa> you should be able to play that without problems i think [19:20:24 CEST] <klaxa> (maybe with some obscure settings not) [19:22:07 CEST] <ntd> anyone know of a no-nonsense x/xorg program that can/will just display N video inputs in a grid view? [19:22:16 CEST] <ntd> had a look at mythtv, not a fan of the forced ui/background [19:23:07 CEST] <ntd> been using a cfgraph, works swell if all inputs have the same fps, surely there must be some software that simply does this? [19:23:08 CEST] <Celmor> I'd use i3wm + mpv [19:24:05 CEST] <ntd> will mpv handle say twelve diff inputs (rtsp, http mjpeg, v4l), diff fps at the same time? [19:24:18 CEST] <Hello71> maybe google i3wm first [19:24:28 CEST] <Celmor> juse use different mpv instances [19:24:28 CEST] <ntd> i had a look at it before [19:25:50 CEST] <Celmor> then what made you drop it as the solution [19:28:16 CEST] <ntd> grid was out of sync [19:28:48 CEST] <Celmor> you mean the video outputs? [19:29:00 CEST] <Celmor> or mpv instances [19:29:31 CEST] <ntd> latter [19:29:47 CEST] <benlieb> is it possible to change the seek duration (left / right keys) using ffplay? For what Im doing 10 second jumps are way too big... [19:29:48 CEST] <Hello71> --video-sync=display-x [19:29:59 CEST] <Hello71> benlieb: consider using a real media player [19:31:00 CEST] <Cracki> Real[tm] Media Player [19:31:21 CEST] <benlieb> @Hello71 I need to capture time frames down to the milisecond the way ffplay allows. Do you know of a media player that does this (and then allows the time to be copied or written to a file)? [19:31:28 CEST] <benlieb> Im ussing the ffplay log. [19:31:53 CEST] <Cracki> any video editor where you can set keyframes or markers [19:31:56 CEST] <benlieb> Am I to assume that the answer is that this 10 second value isnt configurable. [19:32:10 CEST] <Hello71> did you read the ffplay man page [19:32:19 CEST] <benlieb> Hello71: yup [19:32:25 CEST] <Hello71> well then I guess not [19:32:58 CEST] <Cracki> you can single step frames [19:33:03 CEST] <benlieb> @Hello71 Is that because we can always assume that documentation is fully representative of whats possible? [19:33:13 CEST] <Cracki> just singlestep the frames [19:33:17 CEST] <Celmor> ntd, maybe this video helps somewhat https://youtu.be/TtOqqj7RhZY [19:34:18 CEST] <ntd> scripting this is just asking for trouble afaik. some sw must be able to read N inputs, decide on a common "start" and just display them? [19:35:09 CEST] <Cracki> if you have a "wheel/jog" input device (e.g., contour shuttle, or some DIY USB HID), map the singlestep keys to your wheel [19:35:10 CEST] <Celmor> well, how do you decide on a "common start" if all you get are video fits, they would need a marker, there are hardware solutions that do that [19:35:33 CEST] <Hello71> I mean, if you want them all to be in the same picture, then just use n fps filters [19:54:44 CEST] <tuna> For decoding (via hardware) off of a RTSP stream....Earlier you (i forgot the user name and I got detatched so I lost the history) said that I need the same context setup as the hardware encode. However, to init the hwframescontext I need width and height of the frame....but I do not have that until I recieve the NAL unit from the stream....is it possible to init the hwframe without knowing the size until I get that NAL unit? [19:56:37 CEST] <ntd> Hello71, go on? [19:57:18 CEST] <Hello71> tuna: probably not [19:57:19 CEST] <ntd> https://unix.stackexchange.com/questions/434868/play-4-rtsp-streams-in-full… <--- googling ""linux" " video" "grid view"" [19:57:28 CEST] <ntd> one relevant result, no stack answers.. [19:57:29 CEST] <Hello71> I mean, you also don't have the codec information [19:57:49 CEST] <Hello71> usually video-sync=audio works pretty well [22:05:04 CEST] <acos> Howdy all. Trying to find out if a blackmagic hardware capture card will work in this software. What's the cheapest one? Thanks in advance. [22:23:55 CEST] <ThugAim> hey guys. Trying to get 4.0 installed on Tahrpup 6.0.6 [22:25:41 CEST] <ThugAim> compiled and installed with checkinstall, but even when when installing the 'supposedly' 64 bit version it gives the /usr/bin/sensible-pager no such file or directory error. [23:21:19 CEST] <BtbN> That sounds like you are using some random script, and it wants you to replace that with, wenn, a sensible pager. [23:21:23 CEST] <BtbN> *well [23:55:30 CEST] <Cracki> acos, I hear the decklinks work just fine in linux. I would not recommend the "intensity shuttle". check out magewell (chinese), they make proper stuff to capture analog, hdmi, sdi, either over usb, or via pcie [23:56:45 CEST] <Cracki> as for "will it work", that's a matter of device interoperating with the OS. ffmpeg uses standard apis (linux v4l2, windows dshow/msmf/...) [00:00:00 CEST] --- Tue Jun 26 2018

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