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July 2016

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ffmpeg-devel.log.20160720
by burek 21 Jul '16

21 Jul '16

[00:04:23 CEST] <cone-171> ffmpeg 03Petru Rares Sincraian 07master:68f991d9090d: fate: add test for avfft [00:29:43 CEST] <cone-171> ffmpeg 03Michael Niedermayer 07master:fd1588919d2c: avcodec/h264: remove list_count and ref_count clearing [00:29:44 CEST] <cone-171> ffmpeg 03Michael Niedermayer 07master:9bf3fdcd3c21: avcodec/h264_slice: Move h264_init_ps() call into h264_field_start() [01:47:11 CEST] <cone-171> ffmpeg 03James Almer 07master:114870dbc939: configure: add support for new CPUs [02:22:09 CEST] <cone-171> ffmpeg 03James Almer 07master:5adfbd391847: avformat/oggparsevp8: fix pts calculation on pages ending with an invisible frame [02:22:10 CEST] <cone-171> ffmpeg 03James Almer 07master:b3820cabad87: avformat/oggenc: fix page duration calculation when granule differs from timestamp [02:47:38 CEST] <cone-171> ffmpeg 03James Almer 07master:1582e306a479: avformat/avlanguage: make av_convert_lang_to() internal [03:05:47 CEST] <cone-171> ffmpeg 03Petru Rares Sincraian 07master:2383618a4019: fate: add test for chorus filter [06:04:00 CEST] <cone-171> ffmpeg 03Vignesh Venkatasubramanian 07master:134fe28981cd: libavcodec/libvpx: Add VPx alpha decode support [13:01:46 CEST] <cone-761> ffmpeg 03Jun Zhao 07master:a06acfff76ca: ffmpeg_opt: Delete duplicate hwaccel_output_format option. [14:56:25 CEST] <cone-761> ffmpeg 03Mark Reid 07master:41d7642a7be5: libavcodec/dnxhd: add dnxhr profiles [15:33:46 CEST] <BBB> ubitux: do you want to review the vp9 loopfilter simd additions? [15:33:51 CEST] <BBB> ubitux: I think you wrote the original, right? [15:36:11 CEST] <ubitux> i don't think i'll have much time sorry [15:38:16 CEST] <BBB> ok, np [18:45:35 CEST] <cone-761> ffmpeg 03James Almer 07master:7a15cf42ee17: x86/diracdsp: make ff_put_signed_rect_clamped_10_sse4 work on x86_32 [23:12:51 CEST] <cone-921> ffmpeg 03Petru Rares Sincraian 07master:7403dcc34e0f: fate: add test for dcshift filter [23:12:52 CEST] <cone-921> ffmpeg 03Petru Rares Sincraian 07master:3fd2ef922eab: fate: add test for earwax filter [23:53:52 CEST] <sanmarcos> in H264, when there isn't a keyframe and you begin decoding, are you supposed to see green? [23:54:42 CEST] <iive> green, gray or nothing at all. [23:55:09 CEST] <iive> green is what you get when chroma channels contain zeros. [23:55:12 CEST] <nevcairiel> the software decoder typically fills the frames gray [23:55:24 CEST] <nevcairiel> hardware is typically green because the frames are not initialized [23:55:29 CEST] <nevcairiel> (ie. zero) [23:59:07 CEST] <sanmarcos> zero in YUV is green [23:59:19 CEST] <sanmarcos> is there a way to force this to be black [23:59:26 CEST] <nevcairiel> no [23:59:49 CEST] <sanmarcos> I am working with PJSIP and OpenH264 here, and I am getting some green frames initally when doing video conferencing [00:00:00 CEST] --- Thu Jul 21 2016

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ffmpeg.log.20160720
by burek 21 Jul '16

21 Jul '16

[00:23:23 CEST] <CFS-MP3> I'm experimenting a bit with complex filters... anyway, I have a large (well, 4x2) mosaic that shows 8 TV channels at the same time [00:23:31 CEST] <CFS-MP3> it works OK, except that it hiccups from time to time [00:24:02 CEST] <CFS-MP3> ffmpeg says speed 0.8 or so, which I guess means it can't keep up [00:24:16 CEST] <CFS-MP3> but system load is around 40%, and I'm OK using 100% for it [00:24:26 CEST] <CFS-MP3> input and output are network streams, nothing goes to disk [00:24:31 CEST] <CFS-MP3> Any suggestion? [00:25:49 CEST] <furq> decoding multiple inputs is done sequentially iirc [00:26:24 CEST] <furq> i don't think there's a way to do it in parallel short of spawning multiple processes [00:30:02 CEST] <klaxa> and then? pass the decoded raw stream as a pipe to another instance to unify it? will that really speed it up? or am i misunderstanding? [00:30:14 CEST] <furq> https://trac.ffmpeg.org/ticket/1663 [00:30:15 CEST] <furq> oh well that's fun [00:30:31 CEST] <klaxa> hmm, decoding is slower than muxing so it probably will [00:30:52 CEST] <furq> i'm not sure how you'd work around it if that bug is still unfixed [00:31:33 CEST] <klaxa> huh? i am pretty sure i used multiple named pipes with ffmpeg [00:31:34 CEST] <klaxa> let me check [00:33:40 CEST] <klaxa> https://klaxa1337.blogspot.de/2012/09/streaming-videos-over-http-while.html [00:33:48 CEST] <klaxa> too bad i didn't put any output to see the version i used [00:33:53 CEST] <klaxa> but this was late 2012 [00:36:21 CEST] <furq> i wonder if it's an issue with multiple outputs on named pipes [00:36:54 CEST] <c_14> it is [00:37:18 CEST] <c_14> At least, I've run into the same issue in several cases. [00:38:27 CEST] <furq> http://vpaste.net/96A4T [00:38:32 CEST] <furq> i guess something like that should work then [00:38:36 CEST] <furq> it's worth a try anyway [00:41:09 CEST] <CFS-MP3> furq was this "decoding multiple inputs is done sequentially iirc" addressed to me? If it was I'm sorry [00:41:14 CEST] <furq> es [00:41:16 CEST] <furq> yes [00:41:19 CEST] <CFS-MP3> OK [00:41:19 CEST] <furq> all of this was addressed to you [00:41:35 CEST] <CFS-MP3> Thanks, I saw klaxa reply to something so I wasn't sure [00:43:04 CEST] <furq> if you have 40% cpu usage but ffmpeg isn't keeping up with the inputs then either decoding or filtering is to blame [00:43:15 CEST] <furq> splitting that across multiple processes might help [00:43:26 CEST] <furq> if it works at all, that is [00:43:46 CEST] <CFS-MP3> furq I don't know if it's the inputs, or the output, or maybe the network... can be anything I guess. That's what I'd like to figure out [00:43:58 CEST] <furq> i guess it could be the network too [00:44:09 CEST] <furq> but yeah try what i pasted [00:44:16 CEST] <CFS-MP3> I mean, if replacing the i5 with a i7 is the simplest solution I can do that [00:45:15 CEST] <CFS-MP3> Your solution is more fun though [00:45:20 CEST] <furq> http://vpaste.net/ng93U [00:45:21 CEST] <furq> or that, even [00:45:37 CEST] <furq> if it works then you can send me the i7 [00:47:35 CEST] <CFS-MP3> I'll get your shipping details in a bit [00:48:37 CEST] <CFS-MP3> anyway this takes 4 input files and sends the decoded that to a pipe how? I mean, is the data separate in different streams? [00:49:21 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#hstack [00:49:29 CEST] <furq> pipe1 is the top row, pipe2 is the bottom row [00:50:56 CEST] <furq> if you need padding or something then it gets a bit more complicated [00:52:23 CEST] <CFS-MP3> I don't need padding, but I'm overlays subtitles in each of the videos [00:53:00 CEST] <CFS-MP3> This is what I have at the moment [00:54:10 CEST] <CFS-MP3> -thread_queue_size 512 \ [00:54:10 CEST] <CFS-MP3> -i udp://239.168.100.1:5000?fifo_size=1000000 \ [00:54:10 CEST] <CFS-MP3> -thread_queue_size 512 \ [00:54:10 CEST] <CFS-MP3> -i udp://239.168.100.2:5000?fifo_size=1000000 \ [00:54:10 CEST] <CFS-MP3> -thread_queue_size 512 \ [00:54:10 CEST] <CFS-MP3> -i udp://239.168.100.4:5000?fifo_size=1000000 \ [00:54:11 CEST] <CFS-MP3> -thread_queue_size 512 \ [00:54:11 CEST] <CFS-MP3> -i udp://239.168.100.7:5000?fifo_size=1000000 \ [00:54:12 CEST] <CFS-MP3> -thread_queue_size 512 \ [00:54:12 CEST] <CFS-MP3> -i udp://239.168.100.8:5000?fifo_size=1000000 \ [00:54:13 CEST] <CFS-MP3> -thread_queue_size 512 \ [00:54:13 CEST] <CFS-MP3> -i udp://239.168.100.9:5000?fifo_size=1000000 \ [00:54:14 CEST] <CFS-MP3> -thread_queue_size 512 \ [00:54:14 CEST] <CFS-MP3> -i udp://239.168.100.10:5000?fifo_size=1000000 \ [00:54:40 CEST] <CFS-MP3> admitedly I just started today playing with this stuff and I pasted from tutorials [00:54:47 CEST] <CFS-MP3> anyway that works except for what I said [00:56:47 CEST] <furq> http://vpaste.net/lJK5u [00:56:51 CEST] <furq> something like that then [00:58:31 CEST] <CFS-MP3> Damn, looks so simple [00:58:39 CEST] <CFS-MP3> Let's try [01:00:05 CEST] <furq> http://vpaste.net/f5l5n [01:00:07 CEST] <furq> or that [01:06:01 CEST] <CFS-MP3> I need my subs from DVB though (instead of file) so I do have to use that overlay first [01:07:31 CEST] <CFS-MP3> $ ffmpeg -i udp://239.168.100.1:5000?fifo_size=1000000 -i udp://239.168.100.2:5000?fifo_size=1000000 -i udp://239.168.100.4:5000?fifo_size=1000000 -i udp://239.168.100.7:5000?fifo_size=1000000 -filter_complex "[0:v]overlay[out1]; [1:v]overlay[out2]; [2:v]overlay[out3]; [3:v]overlay[out4]; [out1][out2][out3][out4] hstack=inputs=4 [out]" -c:v rawvideo /tmp/pipe1 & [01:07:37 CEST] <CFS-MP3> Does that look reasonable to you? [01:08:15 CEST] <furq> overlay takes two inputs [01:08:32 CEST] <furq> i assume it'd be [0:v][0:s] overlay [out1] etc [01:08:38 CEST] <CFS-MP3> ah yes [01:08:39 CEST] <CFS-MP3> thanks [01:15:54 CEST] <CFS-MP3> [NULL @ 0x38af100] Unable to find a suitable output format for '/tmp/pipe1' [01:16:01 CEST] <CFS-MP3> I must say I cheated [01:16:12 CEST] <CFS-MP3> ah, the -c:v must be [01:16:34 CEST] <CFS-MP3> ah no, it's there [01:16:36 CEST] <CFS-MP3> ffmpeg -i udp://239.168.100.1:5000?fifo_size=1000000 -i udp://239.168.100.2:5000?fifo_size=1000000 -i udp://239.168.100.4:5000?fifo_size=1000000 -i udp://239.168.100.7:5000?fifo_size=1000000 -filter_complex "[0:v][0:s]overlay[out1]; [1:v][0:s]overlay[out2]; [2:v][2:s]overlay[out3]; [3:v][3:s]overlay[out4]; [out1][out2][out3][out4] hstack=inputs=4 [out]" -c:v rawvideo /tmp/pipe1 [01:17:25 CEST] <CFS-MP3> trying with pipe1.ts [01:17:47 CEST] <furq> -f nut /tmp/pipe1 [01:18:18 CEST] <CFS-MP3> File '/tmp/pipe1.ts' already exists. Overwrite ? [y/N] Not overwriting - exiting [01:18:33 CEST] <CFS-MP3> let me try -f nut [01:25:51 CEST] <CFS-MP3> OK so two things are ahppening, one that it asks about overwritting the pipe, and the second it that it says that hstack has an unconnected output [01:26:00 CEST] <CFS-MP3> I guess the 2nd one I can solve with -map [out] [01:39:37 CEST] <CFS-MP3> about pipe, someone suggests to use pipe:// in a forum but my terminal went apeshit :-) [04:14:53 CEST] <jrm> Hi. I have a old script I use for screencasts: http://ftfl.ca/paste/screencast.html. I just ran it for the first time in a long time and I noticed x11grab wasn't compiled in, so when I tried to recompile, I got a not about x11grab begin "legacy". What should I be using instead of x11grab? [04:16:04 CEST] <jrm> s/not/note/ s/begin/being/ [05:53:56 CEST] <Plan_B> How to save EIA_608 subtitles in encoded video (h264.ts) please? [07:00:46 CEST] <Ashish_> Hello All [07:01:10 CEST] <Ashish_> Myself Ashish and i need help to install FFMPEG on centos server. [07:02:24 CEST] <Ashish_> i am installing ffmpeg on centos on root account. i can install it successfully. but when i check on cpanel sub accounts of server, ffmpeg is not installed on it [07:03:27 CEST] <Ashish_> can anyone help me on how to migrate ffmpeg on my cpanel sub accounts. [07:05:17 CEST] <relaxed> Ashish_: where did you install the ffmpeg binary? [07:08:13 CEST] <Ashish_> i am installing it using root ssh account [07:08:24 CEST] <Ashish_> so installed in root accounts. [07:08:41 CEST] <Ashish_> so installed in root account [07:09:24 CEST] <relaxed> did you compile it yourself? [07:09:32 CEST] <Ashish_> Yes [07:09:57 CEST] <Ashish_> i have follow this tutorial. https://trac.ffmpeg.org/wiki/CompilationGuide/Centos [07:10:09 CEST] <relaxed> where is the ffmpeg binary? [07:11:42 CEST] <relaxed> ls -l $HOME/ffmpeg_build/bin/ffmpeg [07:11:52 CEST] <relaxed> what does that return ^^^ ? [07:13:44 CEST] <Ashish_> No such file or directory [07:16:10 CEST] <relaxed> The guide says to use "--prefix="$HOME/ffmpeg_build" when compiling ffmpeg. Did you follow the guide? [07:17:49 CEST] <Ashish_> Yes [07:18:33 CEST] <Ashish_> i can't see ffmpeg in ffmpeg_build/bin directory [07:19:17 CEST] <relaxed> look in the ffmpeg source dir [07:20:43 CEST] <Ashish_> Where i can see or find this directory. [07:24:08 CEST] <relaxed> ffmpeg_sources/ffmpeg [07:24:50 CEST] <relaxed> I'm guessing in /root [07:25:44 CEST] <Ashish_> Yes it is [07:25:54 CEST] <Ashish_> what should i do here ? [07:26:58 CEST] <relaxed> run /root/ffmpeg_sources/ffmpeg/ffmpeg [07:29:14 CEST] <Ashish_> What will this do ? [07:29:23 CEST] <relaxed> run ffmpeg if it's there [07:30:20 CEST] <relaxed> or try /root/ffmpeg_build/bin/ffmpeg [07:31:11 CEST] <Ashish_> it is working fine. but how to run it on my sub accounts of server. [07:31:38 CEST] <relaxed> what is working fine? [07:32:03 CEST] <Ashish_> when i run ffmpeg on root ssh account then it is giving version information. [07:33:04 CEST] <Ashish_> so ffmpeg commands are working fine in root ssh account. but it is giving command not found error when i run it from server sub accounts. [07:35:26 CEST] <relaxed> Ashish_: run this--> whereis ffmpeg [07:37:45 CEST] <Ashish_> ffmpeg: /usr/bin/ffmpeg /usr/local/bin/ffmpeg /usr/share/ffmpeg /usr/share/man/man1/ffmpeg.1.gz [07:44:36 CEST] <limbo> and what is it when you run it from the subaccounts? [07:45:13 CEST] <limbo> also, does it run when you use the full path? What's the other account's PATH variable set to? [07:46:24 CEST] <Ashish_> it is giving same output when run from [07:46:29 CEST] <Ashish_> it is giving same output when run from sub accounts [07:48:54 CEST] <limbo> can you put the output of this command on a pastebin? [07:49:02 CEST] <limbo> ls -lah /usr/bin/ffmpeg /usr/local/bin/ffmpeg /usr/share/ffmpeg [07:49:16 CEST] <Ashish_> ffmpeg: /usr/bin/ffmpeg /usr/local/bin/ffmpeg /usr/share/ffmpeg /usr/share/man/man1/ffmpeg.1.gz [07:49:46 CEST] <relaxed> maybe you have to give cpanel the absolute path to ffmpeg [07:51:00 CEST] <Ashish_> can you show us how ? [07:51:55 CEST] <relaxed> is there an input box in cpanel where you can add ffmpeg's location? [07:52:33 CEST] <Ashish_> i think no. we have follow this tutorial https://trac.ffmpeg.org/wiki/CompilationGuide/Centos to compile it manually [07:53:43 CEST] <Anonaly> hi, I have a question regarding these instructions on how to install FFMPEG [07:53:44 CEST] <Anonaly> https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [07:54:05 CEST] <Anonaly> the final install ends up in my home directory (/home/me/bin) ? [07:54:27 CEST] <Anonaly> should I move that to my /usr/bin folder ? [08:09:19 CEST] <darkapex> Anonaly: Rather than polluting your /usr/bin which used system-wide and which might interfere with your package manager, I suggest just add /home/me/bin to your $PATH [08:10:12 CEST] <Anonaly> ok [08:14:49 CEST] <Ashish_> any one can help on my issue ? [08:26:32 CEST] <Ashish_> anyone can let me know how to uninstall ffmpeg installed it using this tutorial ? https://trac.ffmpeg.org/wiki/CompilationGuide/Centos#FFmpeg [08:29:49 CEST] <darkapex> Ashish_: Try to run `make uninstall` from the directory you ran the installation from [11:21:05 CEST] <whald> is there some function which allows me to tell a freshly av_frame_alloc'd AVFrame apart from an AVFrame which actually holds some data (aka buffers)? i'm currently doing this with my own flag an wonder if I can get rid of it. [11:22:16 CEST] <whald> something like av_frame_has_buffers or the like, or do i have to poke into the avframe myself? [14:40:39 CEST] <friki2015> hi, i'm testing ffmpeg to encode 1080p50 live video (h264 at 5mbps). After some tests I can't get a good quality output. Is there any tip? [14:43:00 CEST] <friki2015> It seems that ffmpeg only use 12 of 24 CPU [14:46:26 CEST] <BtbN> That's barely enough bitrate for that resolution and framerate. [14:46:34 CEST] <BtbN> Some atrifacts are to be expexted. [14:48:48 CEST] <friki2015> i think the result is worst than a 250kbps... i'll try 10mbps [14:49:00 CEST] <iive> friki2015: try to encode based on quality - qp or crf . find what number looks ok (lower number -> better picture, bigger bitrate). [14:49:55 CEST] <iive> this way you can find the bitrate that looks good. round it up. [14:50:20 CEST] <iive> also, i hope you do use libx264 encoder, it's the best encoder outthere [14:52:33 CEST] <iuppiter> got it [14:54:05 CEST] <friki2015> fixing -qp 24 result in 20mbps [14:56:47 CEST] <friki2015> in fact the original video signal is h264 20mbps 1080i [15:00:32 CEST] <friki2015> BtbN iive: thanks, i'll keep working on it [16:13:12 CEST] <Mutantebirl> hi everyone [16:13:16 CEST] <Mutantebirl> i need a little help [16:14:32 CEST] <Mutantebirl> i cant convert to mp3 in ffmpeg, give me a error of codec [16:14:36 CEST] <Mutantebirl> "Encoder (codec mp3) not found for output stream #0:0" [16:15:02 CEST] <Mutantebirl> i get the same error in youtube-dl when i try to download the sound of video in mp3 [16:15:15 CEST] <DHE> ffmpeg doesn't directly include an MP3 encoder. it uses lame as a helper library. your copy wasn't built with it [16:15:16 CEST] <Mutantebirl> "ERROR: audio conversion failed: Unknown encoder 'libmp3lame'" [16:15:39 CEST] <Mutantebirl> yeah DHE, this happen after the update i made yeasterday [16:15:43 CEST] <Mutantebirl> in arch linux [16:17:03 CEST] <Mutantebirl> i think thats happens when updatede ffmpeg and not compile limmp3lame together [16:18:01 CEST] <Mutantebirl> can you help me? [16:18:42 CEST] <DHE> if you did a package manager upgrade and lame broke like that, I'm thinking it's an Arch issue [16:19:42 CEST] <Mutantebirl> i think it is, i just upgrade whit pacman... do you know what can i do to compile that again? [16:20:09 CEST] <furq> Mutantebirl: https://wiki.archlinux.org/index.php/Arch_Build_System [16:21:22 CEST] <Mutantebirl> thanks, i will search about it, and try in my other arch [16:21:41 CEST] <furq> https://www.archlinux.org/packages/extra/x86_64/ffmpeg/ [16:21:48 CEST] <furq> that says it depends on lame, so maybe something else is broken [16:22:25 CEST] <furq> ffmpeg -version | grep libmp3lame [16:22:27 CEST] <furq> does that return anything [16:25:31 CEST] <iive> isn't mp3 patent supposed to expire soon? [16:26:33 CEST] <furq> decoding is already patent-free apparently [16:26:38 CEST] <furq> encoding in 2017 [16:26:49 CEST] <furq> http://en.swpat.org/wiki/Mp3#Final_expiration:_2015_or_2017 [16:27:43 CEST] <Mutantebirl> oh, thanks everyone [16:27:48 CEST] <Mutantebirl> thats is a upgrade error [16:28:05 CEST] <Mutantebirl> in my ohter arch i just dont dont update and test, and work fine [16:28:27 CEST] <Mutantebirl> and now i update him and get the same issue, i will get that to the arch forum [16:28:29 CEST] <Mutantebirl> :) thanks [16:28:36 CEST] <furq> https://git.archlinux.org/svntogit/packages.git/tree/trunk/PKGBUILD?h=packa… [16:28:43 CEST] <furq> that sure looks like it's including lame [16:29:17 CEST] <Mutantebirl> yes, i see... i will make some tests here [16:31:47 CEST] <Mutantebirl> strange... now i see and work fine in ohter arch [16:32:37 CEST] <iive> 30 dec 2017 ... for standard published 1991 [16:33:09 CEST] <furq> just be glad it wasn't invented by walt disney [16:40:04 CEST] <Mutantebirl> thanks everyone [17:58:44 CEST] <Threads> was wondering is it possible to start encoding from a second input time but use the first as a jump point [17:59:25 CEST] <Threads> lets say video starts at 00:07:52.00 but i have to keep adding 2seconds to get it on point [17:59:44 CEST] <Threads> could i do -ss 00:05:52.00 -ss 00:00:02.00 [18:00:02 CEST] <Threads> could i do -ss 00:07:52.00 -ss 00:00:02.00 [18:08:21 CEST] <kepstin> Threads: no, you can't use -ss multiple times (if you do, I think the later one just overwrites the earlier one) [18:09:08 CEST] <kepstin> well, I suppose you can use both -ss as an input option and as an output option, which would sort of do that, but it's not quite the same thing (the output version behaves differently) [18:39:46 CEST] <soulshock> when generating PSNR, is there a reason most guides use "-f mp4 /dev/null" ? I mean it would be a lot faster using "-f rawvideo /dev/null" [18:40:25 CEST] <soulshock> calculating PSNR on an already encoded file that is [18:41:07 CEST] <kepstin> well, fastest is using "-f null" :) [18:41:52 CEST] <kepstin> but if you're doing encoding and psnr measuring in one command, the overhead of putting stuff in a container is probably pretty minimal [18:42:28 CEST] <soulshock> yeah. I was trying to calculate PSNR after encoding, but I guess I might as well do it while encoding [18:43:15 CEST] <kepstin> hmm. actually.. i don't think you can measure it while encoding in ffmpeg [18:43:23 CEST] <kepstin> it'll have to be a separate step after [18:44:06 CEST] <soulshock> ah ok. so -f null it is [18:45:02 CEST] <DHE> psnr is a filter. it gets 2 input decoded (raw) video frames to compare. what you do with the output is arbitrary, right? [18:45:42 CEST] <soulshock> that's my understanding [18:45:55 CEST] <soulshock> but you have to specify an output [18:45:59 CEST] <kepstin> keep in mind that -f null still needs a filename, you can use anything, it's just ignored (/dev/null is fine) [18:46:14 CEST] <DHE> actually, is the nullsink "filter" usable here? discard the video, then have it just pass audio through? [18:46:54 CEST] <DHE> never used nullsink and the description says its mainly intended for C programming use, so this might not be practical [18:47:09 CEST] <kepstin> the issue with using '-f mp4 /dev/null' when measuring psnr between two input videos is that it'll re-encode the video afterwards with x264 before throwing out the output [18:47:14 CEST] <kepstin> so it'll be slower :/ [18:47:51 CEST] <soulshock> -i result.mp4 -i source.mov -filter_complex "scale=1920:1080,format=yuv420p,psnr" -f null -an NUL does indeed work [18:58:08 CEST] <soulshock> seems like it can actually calculate psnr during encoding [19:00:09 CEST] <DHE> libx264 does have an option to do that and output it to the log [19:03:30 CEST] <soulshock> yeah ffmpeg supports it with -psnr [19:28:35 CEST] <CFS-MP3> What's the option to ignore a missing stream? I'd swear I saw it somewhere but I can't find it [19:28:52 CEST] <CFS-MP3> I mean, as if you have 3 input streams from UDP but one of htem is down and you want to just display the other 2 [19:29:00 CEST] <CFS-MP3> instead of having the whole output down [19:44:52 CEST] <DelphiWorld> yo [19:45:00 CEST] <DelphiWorld> will CUDA help with live stream transcoding? [19:45:57 CEST] Action: DelphiWorld still dreaming QSV [19:46:24 CEST] <basisbit> what does "yo" mean? [19:46:45 CEST] <DelphiWorld> basisbit: still trying to make quickSync work on linux but no luck [19:47:13 CEST] <basisbit> :-\ [19:51:26 CEST] <kepstin> in most cases, CUDA is unused for video transcoding; on nvidia cards, you normally use 'nvenc', which uses a dedicated video codec block rather than compute resources. [19:52:16 CEST] <DelphiWorld> kepstin: ok, and how many stream do you think i can transcode [19:52:38 CEST] <furq> two if you use nvenc unless you buy a quadro [19:52:41 CEST] <kepstin> DelphiWorld: depends on the card. Consumer nvidia cards are limited - iirc to 2 streams? [19:52:46 CEST] <furq> ^ [19:52:53 CEST] <DHE> most nvidia GPUs are locked to 2 streams. but it'll do two 1080p streams without issue [19:53:00 CEST] <DHE> realtime at 30fps [19:53:10 CEST] <furq> not sure if quicksync is artificially limited [19:53:16 CEST] <furq> other than the fact that nobody seems to be able to get it to work [19:53:22 CEST] <DHE> even the quadros require a high-end variant. usually they cost $1000 or more for unlocked streams [19:53:24 CEST] <DelphiWorld> fuck quicksync [19:53:26 CEST] <kepstin> i've gotten it working, but only via the vaapi interface [19:54:04 CEST] <kepstin> and the benefits were... well, it's a bit faster than x264 ultrafast on my laptop, i guess? [19:54:43 CEST] <DelphiWorld> kepstin: so you saying i could just use libx264? [19:54:48 CEST] <DHE> x264's opencl support is hit and miss as to whether it will speed up encoding. sometimes it slows it down [19:55:03 CEST] <DelphiWorld> ... [19:55:07 CEST] <furq> doesn't it cause a slight quality hit as well [19:55:10 CEST] <DHE> at the end of the day, x264 provides the best quality. if image quality matters, use it. [19:55:22 CEST] <DHE> yes, it does. most video cards only support 32-bit floating point numbers [19:55:38 CEST] <kepstin> yeah. only use a hardware encoder if your cpus are busy doing something else :/ [19:55:42 CEST] <furq> ^ [19:56:04 CEST] <furq> if x264 ultrafast isn't good enough for you in terms of quality then you're going to be disappointed by quicksync/nvenc anyway [19:56:42 CEST] <furq> i guess it might be useful for decoding [19:56:45 CEST] <DHE> that said, nvenc is handled by dedicated video encoding hardware. the CPU only has to copy the frames into the video cards and collect the encoded output. and when you're doing shadowplay even the first step is handled by the GPU [19:56:52 CEST] <DelphiWorld> furq: i want max number of stream with basic quality, i dont want holywood style movi [19:57:33 CEST] <DHE> then you'll be spending a grand on a Quadro video card. I estimate you'll get around 5 distinct 1080p30 encodes running on it at once for fairly minimal CPU [19:58:03 CEST] <furq> you could probably get more than that for less money from an 8-core xeon [19:58:04 CEST] <DHE> power usage should be fairly low since you won't be using the compute features [19:58:07 CEST] <DelphiWorld> DHE: what about several 720x576 stream? [19:58:24 CEST] <jkqxz> QuickSync is useful for low-power cases as well - you can transcode 300fps or so of 1080p with negligible power overhead. [19:58:29 CEST] <DHE> that low? suddenly I'm disappointed [19:58:37 CEST] <kepstin> or just get some big intel multi-core cpus, and use a bunch of x264 ultrafasts. You shouldn't have any issue doing (possibly even more than one) stream per core at SD resolution... [19:58:56 CEST] <furq> yeah [19:59:05 CEST] <DelphiWorld> hold on guys, rebooting [19:59:08 CEST] <DHE> I've got a first gen e5-2690 which will easily run 10 at that resolution. and the new chips are only better [19:59:29 CEST] <DHE> not talking about the 2698v4 capabilities. :) [19:59:42 CEST] <furq> i get 300fps with 576p at ultrafast on a first gen i7 [19:59:54 CEST] <DHE> sounds about right [20:00:03 CEST] <furq> a recent 8-core xeon should be pushing 20 streams easily [20:00:27 CEST] <DHE> with hyperthreading it actually improves reasonably well. [20:01:11 CEST] <jkqxz> Output from ultrafast is significantly worse than quicksync/nvenc will do, though. Compare to superfast/veryfast. [20:01:14 CEST] <furq> does x264 use avx at all [20:01:36 CEST] <furq> or any of the new features since sandy bridge [20:02:18 CEST] <DelphiWorld> back [20:02:21 CEST] <DelphiWorld> so DHE ? [20:02:39 CEST] <furq> 19:00:03 ( furq) a recent 8-core xeon should be pushing 20 streams easily [20:02:43 CEST] <furq> that is the conclusion we came to [20:02:59 CEST] <DHE> source code says yes, it has avx1 and 2 support [20:03:00 CEST] <furq> if the power bill isn't a concern then just buy a fast xeon and use x264 [20:03:08 CEST] <kepstin> i'm pretty sure x264 uses some avx and/or avx2 stuff now [20:03:11 CEST] <furq> and if you can get qsv working then great [20:03:55 CEST] <DelphiWorld> furq: ok, what about a I7 6700 ? [20:04:14 CEST] <furq> that's quad-core [20:05:16 CEST] <kepstin> yeah, just checked, x264 has a pretty significant amount of avx2 assembly versions of prediction and quantization functions. [20:05:22 CEST] <furq> how annoying [20:05:56 CEST] <furq> i might finally have a reason to upgrade then [20:07:32 CEST] <jkqxz> A 6700 should be somewhere close in total performance to an original E5-2690. [20:08:35 CEST] <jkqxz> (Half as many cores which are close to twice as fast. The worse memory and cache might pull it down for this, though.) [20:08:47 CEST] <DHE> furq: the 2698 v4. crazy expensive, but very good. [20:08:57 CEST] <furq> jkqxz: it's 2.9ghz vs 3.4ghz [20:09:27 CEST] <furq> those sandy bridge xeons are pretty cheap now [20:09:43 CEST] <DHE> yeah but cache support is important. I spent a lot of time with `perf` before I came to the conclusion that my server vendor screwed up [20:10:11 CEST] <DHE> okay, that was more RAM than cache, but still [20:10:19 CEST] <furq> it's not going to be twice as fast though [20:10:31 CEST] <furq> IPC hasn't improved that much since sandy bridge [20:10:41 CEST] <jkqxz> AVX2 as well. [20:11:09 CEST] <kepstin> an annoying thing about avx2 is that the bigger xeon chips have to downclock when running lots of avx2 stuff because of power consumption :/ [20:11:12 CEST] <DHE> furq: in my case the vendor had put 1 stick of RAM in. memory access performance was a bottleneck. I got +30% improvement just by adding a second stick. and it wasn't caused by actually running out of RAM [20:11:33 CEST] <jkqxz> And at comparable power (135W) the desktop one will run a lot faster than 3.4GHz. [20:12:18 CEST] <furq> are sandy bridge xeons locked for overclocking [20:12:19 CEST] <JEEB> hmm, so far I've heard that AVX2 in general requires downclock, which is why writing AVX2 optimizations is harder, since even if you save in cycles it doesn't help if the CPU has to downclock [20:12:35 CEST] <furq> i know that westmere xeons aren't but that was before intel realised they could charge extra for it [20:12:46 CEST] <JEEB> so you have to make sure your optimizations are actually faster by some amount or more [20:13:20 CEST] <furq> yeah it'd be nice to see some benchmarks of how much difference avx2 actually makes with x264 [20:14:08 CEST] <JEEB> the optimizations that got into mainline x264 are faster in general I would guess, since x264 actually had nice review around there [20:14:23 CEST] <JEEB> I think x265 was the project where they merged some AVX2 optimizations and then reverted them :D [20:14:40 CEST] <JEEB> because while they were faster, they weren't faster enough [20:14:48 CEST] <JEEB> (to actually make the whole thing faster with the downclock) [20:14:51 CEST] <furq> http://www.newegg.com/Product/Product.aspx?Item=9SIA75M3GE9839 [20:14:55 CEST] <furq> that is really cheap though [20:15:06 CEST] <furq> you could probably get a pair of those and a dual-socket board for roughly the same money as a new unlocked i7 and board [20:15:14 CEST] <furq> assuming you hate your power company [20:15:38 CEST] <Sirisian|Work> Is there a command in ffmpeg to take a jpg or png and reencode it stripping everything unneeded. I have a jpg that renders in the browser, but can't load it into a WebGL program. Hundreds of other jpgs work fine. I need a kind of clean command. (I tried 'convert' and even going from jpg to png then png back to jpg using -strip and it seems to preserve a kind of artifact that breaks this WebGL program). [20:16:43 CEST] <furq> Sirisian|Work: jpegtran -copy none [20:17:47 CEST] <kepstin> regarding avx2 in x264, dark shikari wrote some stuff: https://mailman.videolan.org/pipermail/x264-devel/attachments/20130423/ffd6… (pdf direct link) [20:17:54 CEST] <jkqxz> Dual-socket boards are still pretty expensive. The rush of Sandy Bridge generation servers being retired is pretty much upon us, though, so wait a bit and you will be able to get a silly machine on ebay for that sort of price. (Or if you go now then big Nehalems are pretty much free.) [20:17:57 CEST] <furq> i'm not sure what the cool kids are using for png these days, i use advpng because i usually have it installed [20:18:20 CEST] <furq> and yeah someone in here linked a dual-socket westmere-ep board with 2x6-core xeons for like $350 [20:18:29 CEST] <furq> granted it was on ebay [20:18:41 CEST] <furq> i think it had a bunch of sas disks bundled as well [20:18:45 CEST] <furq> which is good if you want some nice magnets [20:18:49 CEST] <DelphiWorld> DHE: i'll past you my transcoding cmd and tel me if i'm doing something bad [20:19:18 CEST] <jkqxz> ebay will give you a whole server with that and >100GB of RAM in for that price now. [20:19:27 CEST] <furq> i think this was an entire server [20:19:33 CEST] <furq> it had a case and psu but no ram iirc [20:19:46 CEST] <kepstin> interesting, looks like the avx2 optimizations for x264 were developed on prerelease hardware under nda, so they'd be ready when the chips were released. [20:19:49 CEST] <furq> ddr3-1333 is plenty cheap though [20:20:19 CEST] <DelphiWorld> guys [20:20:44 CEST] <DelphiWorld> i'm wondring why no one from the ffmpeg dev didnt take any action into integrating ffmpeg with MumuDVB? [20:21:11 CEST] <JEEB> nobody cared [20:21:15 CEST] <DelphiWorld> the code is pretty old [20:21:29 CEST] <DHE> necessity is the mother of invention. nobody with coding experience needs it, doesn't happen [20:21:43 CEST] <JEEB> ^this [20:22:06 CEST] <DelphiWorld> ;) [20:22:07 CEST] <DHE> speaking of which, I have 2 bugfixes in my name in the git tree because my company needed them. [20:22:19 CEST] <DelphiWorld> DHE: mumudvb or ffmpeg? [20:22:22 CEST] <DHE> ffmpeg [20:22:48 CEST] <DHE> which describes an alternative: pay someone to do it. [20:23:52 CEST] <JEEB> also usually there tend to be other ways of doing things, and the thing you came up with just happens to be one of many possibilities [20:32:06 CEST] <Sirisian|Work> furq, Didn't work. This is the strangest thing. [20:37:55 CEST] <thebombzen> huh. FFmpeg 3.1.1 is now in the Arch repos. someone gonna change the #ffmpeg title? [20:38:14 CEST] <thebombzen> it says "FFmpeg 3.0 is released" which seems a bit out of date, n'est-ce pas? [20:38:32 CEST] <JEEB> that's just how much the people care about this channel's topic ;) [20:38:53 CEST] <DHE> 5 months old now [20:49:24 CEST] <furq> if the topic is being updated then ffmpeg is also in debian testing and stable backports now [20:59:11 CEST] <DelphiWorld> back guys [20:59:21 CEST] <DelphiWorld> see my current FFline: [20:59:32 CEST] <DelphiWorld> ffmpeg -i udp://@239.100.1.8:1234 -acodec libfdk_aac -ac 2 -ab 64k -ar 48000 -s 720x576 -r 29.97 -vcodec libx264 -vb 1000k -x264opts colorprim=bt709:transfer=bt709:colormatrix=bt709:fullrange=off -vprofile baseline -level 3.1 -f flv rtmp://myCDN.net/live/echorouktv [21:00:01 CEST] <DelphiWorld> any commant? [21:00:26 CEST] <CFS-MP3> furq, could you test this stream if you have a minute? http://54.73.134.173//live/Mosaic.stream/playlist.m3u8 [21:00:45 CEST] <CFS-MP3> I'm trying to find out where the hiccup comes from... I thought it was load but it isn't [21:00:55 CEST] <CFS-MP3> I thought it could be network but that doesn't seem to be the case either [21:01:08 CEST] <CFS-MP3> asking you since you assisted a bit yesterday [21:07:18 CEST] <DelphiWorld> CFS-MP3: using wowza? [22:13:04 CEST] <shincodex> um [22:13:06 CEST] <shincodex> how to send [22:13:11 CEST] <shincodex> fpic to ffmpeg configure [22:13:35 CEST] <JEEB> --enable-pic I think [22:13:48 CEST] <JEEB> although usually when you are getting that error it's your dependencies that have to be PIC'd :P [22:15:41 CEST] <shincodex> no [22:15:44 CEST] <shincodex> its cause i take ffmpeg [22:15:48 CEST] <shincodex> shove em all in statics [22:15:55 CEST] <shincodex> then make a C++ code around it [22:16:04 CEST] <shincodex> make it exportable and wala libffmpeg.so [22:16:16 CEST] <shincodex> i just set fpic on it and was like why no ffmpeg work oh... [22:16:18 CEST] <shincodex> no fpic [22:17:11 CEST] <CFS-MP3> Are named piped known to work? All info I find online is contradictory [22:17:18 CEST] <shincodex> i couldnt find it cause i kept looking for fpic [22:17:23 CEST] <shincodex> instead of pic [22:17:51 CEST] <shincodex> boom mess compiles [22:18:33 CEST] <DHE> CFS-MP3: named pipes work, but not all formats support streaming without seek capability [22:18:51 CEST] <shincodex> great [22:18:57 CEST] <shincodex> This program has stopped working [22:18:59 CEST] Action: shincodex suicides [22:19:48 CEST] <CFS-MP3> DHE: What I'm trying to do is create a mosaic of 9 channels in a 3x3 grid [22:20:04 CEST] <DHE> --enable-pic is needed. extra-cflags isn't enough because the ASM needs it as well [22:20:06 CEST] <CFS-MP3> in theory it works, but the playback hiccups a lot [22:20:17 CEST] <DHE> live channels? oh dear... [22:20:30 CEST] <CFS-MP3> DHE yes [22:20:58 CEST] <DHE> just opening 9 inputs and analyzing them might take long enough that your buffers might overflow [22:20:58 CEST] <CFS-MP3> furq suggested to split the work into 3 ffmpeg streams (one for each row) and then combine the output with another one [22:21:06 CEST] <DHE> that might help [22:21:08 CEST] <shincodex> might I ask [22:21:22 CEST] <shincodex> Why use the tool pr? when mingw msys dont include this tool? [22:21:39 CEST] <shincodex> i get yasm not being includeing and fetching it [22:21:42 CEST] <CFS-MP3> DHE so the idea is to use named pipes between the first 3 ones and the final one [22:21:58 CEST] <CFS-MP3> you can check the stream here: http://54.73.134.173//live/Mosaic.stream/playlist.m3u8 [22:22:01 CEST] <CFS-MP3> it more or less works [22:22:12 CEST] <CFS-MP3> except when it hiccups which is quite often [22:23:08 CEST] <DHE> not bad, but yeah I've seen codec glitches [22:24:25 CEST] <CFS-MP3> that's just one ffmpeg [22:24:28 CEST] <CFS-MP3> 25275 captions 20 0 2508828 1.457g 14552 R 150.7 9.4 16:34.16 ffmpeg [22:24:32 CEST] <CFS-MP3> 150.7 [22:24:41 CEST] <CFS-MP3> the CPU is a quadcore i5 [22:25:06 CEST] <DHE> that's a very large process memory footprint [22:25:06 CEST] <CFS-MP3> so I'm hopeful that I can somehow improve the usage... [22:25:15 CEST] <DHE> and that's 2 instances of ffmpeg you've pasted already using the majority of your CPU time [22:25:46 CEST] <CFS-MP3> 2? [22:26:03 CEST] <DHE> well, at 150% CPU I'm extrapolating [22:26:11 CEST] <DHE> i5 maxes out at 400% [22:26:38 CEST] <CFS-MP3> Yes of course... [22:26:50 CEST] <CFS-MP3> 150% comes from one ffmpeg doing 6 channels plus the final encode [22:27:04 CEST] <DHE> hmm... that seems more reasonable then [22:27:06 CEST] <CFS-MP3> With 3 it goes to 80% or so [22:27:18 CEST] <furq> does either method work [22:28:14 CEST] <CFS-MP3> furq my attempts yesterday with pipes didn't work - I tried pipe:// but output went to terminal (?) [22:28:34 CEST] <furq> what happens without it [22:28:45 CEST] <CFS-MP3> without pipe:// it tries to overwrite the pipe with a regular file [22:29:06 CEST] <DHE> a named pipe isn't significantly different from a file. make it with mkfifo and just point to it as if it were a file [22:29:38 CEST] <furq> are you sure it's overwriting it [22:29:58 CEST] <furq> it always prompts if the output filename exists [22:30:03 CEST] <CFS-MP3> DHE that was what I did yesterday but let me retry, I was tired so maybe I made a mistake [22:31:04 CEST] <furq> failing that then -f nut - > /tmp/pipe1 [22:35:18 CEST] <CFS-MP3> File '/tmp/fifo1' already exists. Overwrite ? [y/N] Last message repeated 7 times [22:35:18 CEST] <CFS-MP3> N [22:35:18 CEST] <CFS-MP3> Not overwriting - exiting [22:35:48 CEST] <furq> what happens if you say yes [22:35:50 CEST] <CFS-MP3> Let's see if I say Y :-) [22:36:02 CEST] <furq> i've never needed to do it but you get the same prompt if you write to /dev/null [22:36:18 CEST] <CFS-MP3> trying... it takes a while to ask though since it has to open all input files [22:38:40 CEST] <CFS-MP3> OK, with y it seems to be writing to the fifo, not replacing it with a regular file [22:39:52 CEST] <CFS-MP3> let's add -y somewhere... [22:42:36 CEST] <CFS-MP3> OK... so that seems to work... [22:43:58 CEST] <CFS-MP3> what's the most logical output for the first ffmpeg to send to the fifo? [22:44:20 CEST] <CFS-MP3> I currently have -vcodec libx264 -g 60 -vprofile main -vbsf h264_mp4toannexb -strict experimental -preset ultrafast -threads 0 -deinterlace -f mpegts /tmp/fifo1 [22:44:33 CEST] <CFS-MP3> which is just what I need for the final thing [22:49:41 CEST] <CFS-MP3> With this [22:49:42 CEST] <CFS-MP3> c:v rawvideo /tmp/fifo1 [22:50:03 CEST] <CFS-MP3> I get [NULL @ 0x3463600] Unable to find a suitable output format for 'c:v' [22:50:03 CEST] <CFS-MP3> c:v: Invalid argument [22:50:28 CEST] <furq> you forgot a - [22:51:08 CEST] <furq> and don't use x264 for the intermediate files, you'll need to reencode to use hstack at the end [22:51:55 CEST] <Bray90820_> Can someone tell me why I'm getting this error? [22:51:55 CEST] <Bray90820_> http://pastebin.com/raw/ctw9EahT [22:52:30 CEST] <furq> [mp4 @ 0x7fc51a822c00] Could not find tag for codec wmav2 in stream #1, codec not currently supported in container [22:52:35 CEST] <furq> you can't store wma in mp4 [22:53:05 CEST] <CFS-MP3> furq, for now I just want to feed the fifo with the raw video :-) [22:53:16 CEST] <furq> -c:v rawvideo [22:53:22 CEST] <furq> and yeah that's what you should use anyway [22:53:25 CEST] <CFS-MP3> furq yes, have that [22:53:33 CEST] <Bray90820_> furq: Can I convert the audio to something else? [22:53:44 CEST] <furq> CFS-MP3: you're missing the - before c:v [22:53:46 CEST] <CFS-MP3> how do I pick the labeled stream I want to copy? [22:53:49 CEST] <furq> Bray90820_: -c:a aac [22:53:52 CEST] <CFS-MP3> furq, yes, I added that [22:53:58 CEST] <CFS-MP3> so now I have [22:54:15 CEST] <CFS-MP3> Filter scale has an unconnected output [22:54:22 CEST] <furq> pastebin the full command [22:54:23 CEST] <Bray90820_> furq: -c:a "video.mp4? [22:54:34 CEST] <furq> what [22:55:35 CEST] <Bray90820_> Sorry that was a little botched would I put -c:a acc before the output file? [22:55:38 CEST] <CFS-MP3> furq: http://pastebin.com/8Sd6L9QE [22:55:52 CEST] <furq> Bray90820_: yes [22:55:55 CEST] <Bray90820_> Ok thanks [22:55:59 CEST] <furq> replace -c:a copy with -c:a aac [22:56:40 CEST] <Bray90820_> Sorry is it aac or acc [22:56:44 CEST] <furq> aac [22:56:56 CEST] <Bray90820_> Thanks [22:57:02 CEST] <furq> CFS-MP3: that looks like it should work [22:57:17 CEST] <furq> although you can just do [22:57:22 CEST] <Bray90820_> it seems to be working now thanks [22:57:35 CEST] <furq> [out0][out1][out2] hstack=inputs=3,scale=w=iw/2:h=ih/2 [out] [22:57:37 CEST] <CFS-MP3> Filter scale has an unconnected output [22:58:39 CEST] <CFS-MP3> Yes I have lots of intermediate stuff... for clarity (for myself, since I'm learning). I'll get rid of that a bit later [22:59:58 CEST] <kepstin> it might be worthwhile to use something like '-f nut' rather than e.g. '-f rawvideo' so you don't have to manually specify the video size and codecs on the next ffmpeg in the chain. [23:03:54 CEST] <CFS-MP3> thanks kepstin [23:03:57 CEST] <CFS-MP3> so: [23:03:58 CEST] <CFS-MP3> http://pastebin.com/b40bZ8BJ [23:04:00 CEST] <CFS-MP3> that seems to work [23:04:31 CEST] <CFS-MP3> so part two - take that as input and generate the output... let's see if I can get it to work [23:04:44 CEST] <kepstin> you might have to manually specify -c:v rawvideo [23:04:56 CEST] <kepstin> it'll default to encoding to mpeg4, probably not what you want :) [23:07:01 CEST] <CFS-MP3> yes, I'm horrified at the current quality [23:07:52 CEST] <CFS-MP3> but even though it looks horrible, it's reading from the pipe [23:07:55 CEST] <CFS-MP3> and encoding it [23:07:58 CEST] <CFS-MP3> so that's quite progress [23:10:18 CEST] <CFS-MP3> OK, got it with raw video... now I'm at the point in which I'm getting excited [23:23:12 CEST] <CFS-MP3> OK, so it's working, but I have the same hiccup problem [23:23:20 CEST] <CFS-MP3> is this warning serious? [23:23:21 CEST] <CFS-MP3> [mpegts @ 0x4359ae0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 32) [23:23:21 CEST] <CFS-MP3> [graph 0 input from stream 0:3 @ 0x4480b60] Changing frame properties on the fly is not supported by all filters. [23:23:21 CEST] <CFS-MP3> Last message repeated 2 times [23:23:21 CEST] <CFS-MP3> [mpegts @ 0x43c5360] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8) [23:23:27 CEST] <CFS-MP3> I mean could that explain it? [23:25:40 CEST] <furq> you need to increase it once per input [23:26:52 CEST] <CFS-MP3> furq I just did (to 512) and I still get the warning [23:34:43 CEST] <CFS-MP3> OK so now I have 3 ffmpegs [23:34:45 CEST] <CFS-MP3> 16058 captions 20 0 1181832 448504 13808 R 73.9 2.7 3:21.51 ffmpeg [23:34:45 CEST] <CFS-MP3> 15999 captions 20 0 1190588 617964 13668 R 73.6 3.8 3:38.90 ffmpeg [23:34:45 CEST] <CFS-MP3> 16135 captions 20 0 681052 326332 13580 S 35.1 2.0 1:36.43 ffmpeg [23:34:59 CEST] <CFS-MP3> all of them below 100%, and the server itself is below 300% at all times [23:35:03 CEST] <CFS-MP3> which seems quite good [23:35:13 CEST] <CFS-MP3> but the video still stutters...the bastard [23:36:22 CEST] <Jpacker> Hello, I'm using FFMPEG to Mix Two Audio files but there is a very small delay with the two. I've read that -itsoffset only applies to Video streams, is there any other workaround to get the audio synced? [23:40:10 CEST] <thebombzen> Jpacker: if you use the "-ss" option /before/ the input, it'll only seek that one stream. [23:40:37 CEST] <furq> CFS-MP3: does it stutter with fewer inputs [23:40:39 CEST] <thebombzen> so if you have ffmpeg -i first.wav -ss 0.5 -i second.wav, it'll seek to 0.5 seconds in second.wav, but it'll start first.wav as normal. [23:40:46 CEST] <furq> in particular with one input [23:41:16 CEST] <furq> actually, before you test that [23:41:23 CEST] <furq> does it stutter if you output to a file instead of to hls [23:42:10 CEST] <Jpacker> Thank you for such a fast response! Going to try that now. [23:43:01 CEST] <CFS-MP3> furq no, if I just take each of the streams individually and encode them they play perfectly [23:45:02 CEST] <CFS-MP3> furq you can check this out: http://54.73.134.173/live/ngrp:sub_BBC_One.stream_all/playlist.m3u8 [23:45:12 CEST] <CFS-MP3> that plays perfectly all the time [23:46:04 CEST] <thebombzen> furq: CFS-MP3: is there any reason not to do all that in a filtergraph? [23:46:14 CEST] <thebombzen> like why do you need 3 FFmpegs [23:47:05 CEST] <Jpacker> I'm getting an unrecognized option thebombzen, if I wanted to seek the first wav I would call ffmpeg -ss 0.5 -i first.wav? [23:47:19 CEST] <thebombzen> yea [23:47:40 CEST] <CFS-MP3> thebombzen yes, I couldn't get one single ffmpeg instance to take all the CPU [23:48:06 CEST] <CFS-MP3> so I thought (but I might be wrong) that was causing the stuttering [23:48:15 CEST] <thebombzen> ah [23:48:22 CEST] <thebombzen> stupid question: [23:48:25 CEST] <thebombzen> did you try -threads? [23:48:46 CEST] <CFS-MP3> Yes, with lots of different values and even positions in the command line :-) [23:48:52 CEST] <thebombzen> Jpacker: not sure what you mean by unrecognized option, cause -ss works. see: https://ffmpeg.org/ffmpeg.html#Main-options for details with -ss [23:49:13 CEST] <thebombzen> CFS-MP3: huh. I don't like named pipes just because there's a large pipe-wait. but then again if your cpu is the limiting factor I guess not...? [23:50:31 CEST] <CFS-MP3> thebombzen I can live with the pipe-wait, I mean, I don't care that instead of live is now live +3 seconds [23:50:40 CEST] <CFS-MP3> it's the stuttering that is annoying me [23:51:45 CEST] <Jpacker> I believe it worked thebombzen, had an extra space in there. Thank you :) [23:51:53 CEST] <thebombzen> Jpacker: yw :D [23:52:00 CEST] <thebombzen> CFS-MP3: what do you mean by stuttering? [23:52:07 CEST] <thebombzen> do you mean, like, FFmpeg is blocking on input? [23:52:45 CEST] <thebombzen> cause that's what I mean by pipe-wait. named pipes have very small buffers so if you transcode through a pipe, frequently you'll get I/O blocking [23:53:12 CEST] <thebombzen> I tried the filtergraph -lavfi '[0:v] [1:v] [2:v] hstack=inputs=3 [row1], [3:v] [4:v] [5:v] hstack=inputs=3 [row2], [6:v] [7:v] [8:v] hstack=inputs=3 [row3], [row1] [row2] [row3] vstack=inputs=3 [v]' -map '[v]' and that worked with a test [23:53:22 CEST] <thebombzen> although didn't do any profiling. [23:54:15 CEST] <thebombzen> CFS-MP3: try doing this ffmpeg -i <inputs> <conversion> -f matroska - 2>/dev/null | ffmpeg -f matroska -i - -c copy out.file [23:54:17 CEST] <CFS-MP3> thebombzen stuttering as in playback the video stops for a second or two from time to time and then continues happily for a bit, etc [23:54:34 CEST] <CFS-MP3> the inputs are UDP streams and so is the output [23:54:42 CEST] <thebombzen> then the second ffmpeg process will print a "playback speed" in units of times. i.e. 1.5x speed, etc. [23:55:10 CEST] <thebombzen> if it's lower than 1.0x speed, then try it without doing a pipe. and see if it's lower than 1.0. if htere's a difference, the I/O blocking is likely. [23:55:39 CEST] <CFS-MP3> playback is over 1x at all times (now, but with just one graph doing all 9 inputs it was around 0.6) [23:56:11 CEST] <thebombzen> so when you play the stream it stutters. okay. [23:56:17 CEST] <thebombzen> stupid question: did you try another player like MPV? [23:57:23 CEST] <thebombzen> also it's possible that it's a networking problem. i.e. there's a UDP bottleneck somewhere [23:57:55 CEST] <thebombzen> and that it's not FFmpeg's fault. haven't done much UDP programming other than sending video packets over UDP [23:58:17 CEST] <thebombzen> wait that's relevant haha. but not really the same application [23:59:08 CEST] <CFS-MP3> the thing is that only this stream is having the problem [23:59:31 CEST] <CFS-MP3> all others are working fine, but they come from ffmpeg just transcoding [23:59:39 CEST] <CFS-MP3> can mpv play a network stream? [23:59:50 CEST] <furq> yeah [00:00:00 CEST] --- Thu Jul 21 2016

1 0

ffmpeg-devel.log.20160719
by burek 20 Jul '16

20 Jul '16

[01:37:14 CEST] <Zeranoe> I'd like to get some feedback for a new Windows builds page I've been working on: http://zeranoe.com/shared/db49jzDNjx1wKPtywhbJX5iJdsLDoTXy/builds/index.html [02:29:35 CEST] <atomnuker> ubitux: what was it that blocked your thread names patch again? [02:30:48 CEST] <atomnuker> (yeah, I'm starting to really wish I could see what the threads are doing) [03:12:55 CEST] <jamrial> atomnuker: i think he ran into compilation issues in some specific scenarios and after four or so iterations he said fuck it :p [03:14:18 CEST] <atomnuker> it sucks that every single bloody pthreads implementation needed to make a completely different function to do this [03:14:24 CEST] <atomnuker> and each has its own damn quirks [03:14:36 CEST] <atomnuker> honestly, how do you screw up something like that? [03:15:22 CEST] <atomnuker> though I think only the OSX one had a very different API [03:15:26 CEST] <jamrial> Zeranoe: wouldn't it be better to offer builds for the latest stable branch in the Recommended section, and the git master builds below that? [03:16:52 CEST] <jamrial> atomnuker: the function apparently is (or was) not standarized, so everyone implemented it their own way. the result was this [03:53:47 CEST] <Zeranoe> jamrial: That would depend on if stable builds are actually considered "better", which I've heard they are not. [03:54:39 CEST] <jamrial> Zeranoe: there's a new release every three months, and point releases every few weeks [03:55:36 CEST] <Zeranoe> Does that make them preferred? [03:55:56 CEST] <jamrial> regressions are introduced to git master quite often, and even if they are fixed fast, if someone downloaded a snapshot from your website that happened to be in the middle of some big change or merge and never bothered to get a new one again will be stuck with that faulty build for who knows how long [03:56:42 CEST] <jamrial> and people /will/ download a binary and stick with it until it doesn't do whatever they want to do at the time and see if they can get a newer one [03:57:11 CEST] <jamrial> so ideally, or at least in my opinion, the recommended one should be a "stable" build [03:57:50 CEST] <Zeranoe> I've been told before that release builds are cherry picked for patches, and aren't tested any further than a regular commit. Perhaps something has changed [03:58:38 CEST] <jamrial> release builds get fixes backported when possible, hence the point releases [03:59:38 CEST] <Zeranoe> I'll keep that in mind. I'd like to hear other peoples thoughts on the matter as well. [03:59:51 CEST] <jamrial> sure, as i said it's my opinion [04:01:37 CEST] <jamrial> you can always mention something like "for the bleeding edge/latest features, get the daily snapshots below instead" alongside a warning that they may be less stable or more buggy than the recommended one [04:04:26 CEST] <Zeranoe> Also an option [07:06:12 CEST] <zyme> Does anyone know a good or fairly easy transcoding video server to output to chromecast? [07:06:45 CEST] <rcombs> [warning: corporate shilling] try Plex? [07:07:33 CEST] <zyme> Thes ones I've run across all use ffmpeg for the transcoding, but don't have instructions on how to setup for chromecasts, even if they list in the specs chromecast compatible; [07:08:22 CEST] <zyme> Plex requires transcoding to have the videos selected ahead of time, and is far from live. [07:08:30 CEST] <rcombs> wat [07:09:01 CEST] <rcombs> it'll live-convert for chromecast (or any other supported device) [07:09:01 CEST] <zyme> At least thats how jt worked last time I used it. [07:09:34 CEST] <rcombs> hi I'm in charge of the transcode engine at Plex [07:13:04 CEST] <zyme> This supposedly is with the plex interface (using the ios remote) or the upnp interface? And is it included with the free version because I use to pay for it but didn't really get anything out of it that I used... [07:17:32 CEST] <rcombs> free; the native clients [07:17:39 CEST] <rcombs> hard to do much of anything over DLNA [07:32:38 CEST] <zyme> And will it Taranscode any video type the PC's codecs support? Or actually I'm really only concerned with playing xvid and hevc. More so Hevc. [07:42:36 CEST] <ubitux> atomnuker: i couldn't make the configure detect a difference between a function with 2 or 3 args iirc [10:35:35 CEST] <rcombs> why does AV_CODEC_ID_AAC_LATM exist [10:36:03 CEST] <rcombs> as opposed to having it as a format for AV_CODEC_ID_AAC like ASC and ADTS, which BSFs and the like can convert between [10:37:36 CEST] <nevcairiel> iirc latm was deemed different enough to warrant a distinct codec id, so the decoder clearly knows what its getting instead of having to guess from the bitstream [10:40:28 CEST] <nevcairiel> adts is just a header infront of a basic aac frame, latm is more complex [10:41:26 CEST] <rcombs> it seems pretty easy to distinguish from ADTS or ASC [10:41:30 CEST] <rcombs> given it has a syncword [10:42:02 CEST] <nevcairiel> i'm sure you can find the discussions on the ML [10:42:12 CEST] <rcombs> it can technically have multiple substreams iiuc but afaict nothing in lav* supports that [10:42:54 CEST] <rcombs> maybe the LOAS demuxer should convert to ADTS [10:44:09 CEST] <nevcairiel> thats just a raw demuxer right now, and its more commonly found in mpegts anyway [10:44:38 CEST] <rcombs> doesn't the mpegts demuxer call into the raw demuxer? [10:44:45 CEST] <nevcairiel> why would it do that [10:44:54 CEST] <rcombs> it uses them for probing, at least [10:44:57 CEST] <nevcairiel> the raw demuxer does nothing [10:45:09 CEST] <nevcairiel> well not directly, thats all avformat utils.c magic [10:45:13 CEST] <rcombs> I guess the parser handles packetization [10:45:42 CEST] <rcombs> my current issues are that (1) you can't remux LATM to <something else> [or vice versa but nobody cares] and (2) it looks like it'd be awkward to fit in with my AAC-parser-on-steroids [10:46:40 CEST] <nevcairiel> there is no reason a bsf couldnt change the codec id [10:47:07 CEST] <nevcairiel> its just part of codecpar, and it can change codecpar [10:47:23 CEST] <rcombs> (in which rcombs pulls a bunch of aacdec_template.c into aac_parser.c to report sampling rate and channel layout and profile, which obnoxiously requires parsing the whole damn bitstream because AAC puts that information all over the place [lol SBR] and doesn't code element lengths) [10:47:40 CEST] <nevcairiel> sounds like something the decoder should just continue doing then [10:47:55 CEST] <rcombs> had to build without the decoder because reasons [12:35:33 CEST] <JEEB> rcombs: yeah that's why mkvmerge used to have a boolean for he-aac :) [12:35:55 CEST] <JEEB> instead of "just parsing it" [12:40:28 CEST] <roxlu> Hi [12:41:05 CEST] <roxlu> Why does the rtp-h264 decoder only uses two STAP-A packets? (here on line 211 https://www.ffmpeg.org/doxygen/2.4/rtpdec__h264_8c_source.html ) ? [12:52:12 CEST] <jkqxz> roxlu: What do you mean? It makes two passes through the NALs in the packet: the first one to determine how big the output data needs to be, the second one to copy to the output. [12:56:12 CEST] <roxlu> jkqxz: ah thanks I see it now [12:57:09 CEST] <zyme> Rcombs: your right it works, Last time I checked it must have not supported the files [13:06:17 CEST] <zyme> I think I almost had UMS working too, I had it down to only two problems, every file type ended in .ts for some reason, even thiugh and I couldn't find anything about that in any of the config files. And lastly it was transcoding to x264 3.1 instead of 4.1. [13:07:33 CEST] <zyme> ~Even though it was really a mkv file, and configured for such ~ [14:59:38 CEST] <KGB> [13FFV1] 15retokromer opened pull request #22: typos (06master...06patch-1) 02https://git.io/vKr8Y [15:18:56 CEST] <ubitux> anyone to help merging 8d36932c? [15:24:21 CEST] <ubitux> so we have ff_h264_decode_slice_header() doing { h264_slice_header_parse(); h264_field_start(); }. what the commit is supposed to do is pick all h->current_slice related chunks from h264_slice_header_parse() and move them in ff_h264_decode_slice_header() around the two h264_slice_header_parse() and h264_field_start() calls [15:28:32 CEST] <ubitux> but this is tricky, because in our case, the init of the ps are entangled in h264_slice_header_parse() [15:28:55 CEST] <ubitux> (see previous merge commit mostly nooped) [15:37:29 CEST] <cone-216> ffmpeg 03Clément BSsch 07master:21c081a24651: lavc/h264: fix spacing before comment [17:14:55 CEST] <roxlu> I'm trying to stream the Sintel video (no audio) using RTP (http://peach.themazzone.com/durian/movies/sintel-1024-surround.mp4) But the generate RTP stream cannot be handled by FFMPEG itself (or I'm using the wrong commands). [17:19:10 CEST] <roxlu> Oh sorry, I was wrong, I was using a wrong SDP file :# [18:44:25 CEST] <cone-216> ffmpeg 03Michael Niedermayer 07master:64caafaeb18a: avformat: Add tee protocol [20:17:20 CEST] <ubitux> anyone wants to help with the current h264 wave? [20:17:32 CEST] <ubitux> because i feel like i'm going to noop it and continue the merging of the rest [20:17:44 CEST] <ubitux> unless someone has a better idea how to deal with it [20:20:29 CEST] <Chloe> ubitux: what happened to the commit which moved h264_init_ps() to h264_field_start()? [20:20:50 CEST] <ubitux> mostly noop'ed [20:20:55 CEST] <ubitux> because it was breaking things [20:22:27 CEST] <ubitux> it looks like h264_slice_header_parse() in ffmpeg needs some work in order to be able to merge the commits from libav [20:22:47 CEST] <ubitux> it's probably handling more corner cases, but also makes things much more complex [20:53:04 CEST] <iive> this doesn't sound like a good thing. [20:56:03 CEST] <ubitux> right [20:56:07 CEST] <ubitux> who wants to help? [21:36:30 CEST] <BBB> jamrial_: does http://pastebin.com/MW6bsxVv fix the issue? [21:37:32 CEST] <jamrial_> BBB: yeah, it did [21:38:04 CEST] <BBB> lemme test 32bit before I re-send it... [21:38:06 CEST] <BBB> tnx for testing [21:38:19 CEST] <jamrial_> i was writing a reply to tell you i thought i found what was wrong :p [21:39:53 CEST] <BBB> anyone want to writ a mix4 vp9 loopfilter impl? [21:39:55 CEST] <jamrial_> it was basically this unless i'm reading it wrong http://pastebin.com/JKhkz6Hd but you evidently figured it out already [21:40:23 CEST] <BBB> yeah thats the same thing [21:40:42 CEST] <BBB> I mean, you can sort of see that the assembly isnt the pretties tthing in the world, in that it doesnt use all the x86inc.asm magic stuff [21:40:50 CEST] <BBB> but it works really well [21:43:02 CEST] <BBB> I guess Ill write the mix4 myself then, should add another few % to the performance on haswell [21:43:17 CEST] <BBB> I wish I knew of reasonable ways to make token decoding faster [22:35:01 CEST] <michaelni> ubitux, posted a patch that moves the h264_init_ps() call point [23:04:14 CEST] <michaelni> ubitux, does this help or was that unrelated to the next merge ? [23:08:15 CEST] <ubitux> michaelni: that looks very helpful yes [23:08:33 CEST] <ubitux> i wouldn't mind a longer explanation in patch 2/2 [23:09:07 CEST] <ubitux> (thanks a lot btw) [23:09:23 CEST] <michaelni> will add a proper explanation [23:33:05 CEST] <Chloe> ubitux: thanks for the comment about the CONST option type, it looks a lot better now [23:34:34 CEST] <ubitux> Chloe: np; so you are josh/Ilya? [23:34:41 CEST] <Chloe> yes [23:35:04 CEST] <ubitux> you wouldn't be Ilya87 in the trac btw? [23:35:16 CEST] <Chloe> No, I'm 'Illya' in trac [23:35:35 CEST] <ubitux> alright; i got confused since i'm working on a ticket where a certain Ilya87 was active [23:37:17 CEST] <Chloe> those are the last libopenmpt patches (for now), if they're alright then I have those and then the two others from manx to merge. (scary) [00:00:00 CEST] --- Wed Jul 20 2016

1 0

ffmpeg.log.20160719
by burek 20 Jul '16

20 Jul '16

[00:01:12 CEST] <whomp> or, how can i convert a video from 29.97 fps to 30 fps? i believe this will solve the problem, as 44100 is divisible by 30. i just want to slow the video down about .1% [00:01:47 CEST] <whomp> i see a command to change the speed, but only by a multiplier, which for a repeating fraction like 29.97 / 30, doesn't seem like it'll work [00:03:34 CEST] <furq> whomp: you can use -shortest to truncate the output file to the shortest input file length [00:04:28 CEST] <whomp> furq, so i would split this up into its audio and video components, then re-merge with shortest? [00:05:29 CEST] <furq> the docs say input stream, so it might work with one file [00:05:35 CEST] <furq> i've only ever used it with separate files [00:06:20 CEST] <whomp> furq, won't i still have the issue of 44100 not being divisible by 29.97? [00:06:30 CEST] <whomp> how would one ever get the audio and video to be the same length? [00:09:26 CEST] <furq> those two things aren't connected at all [00:11:24 CEST] <whomp> when they aren't the same length, ios gets significant performance issues [00:28:38 CEST] <esoul> Hello everybody. I am experimenting with screen capture and want to record rawvideo(like yuv420p or rgb5555l2) . It works with tools like Potplayer. BUt I really want to use ffmpeg. But I am getting the "real-time buffer [screen-capture-recorder] [video input] too full" message pretty quick. I cannot find a solution anywhere. [00:29:08 CEST] <esoul> Is there a dummy solution ? [00:29:28 CEST] <furq> increase -rtbufsize [00:29:46 CEST] <esoul> set to 1500M. But is full really quick. Raw video is huge [00:30:17 CEST] <furq> is your disk keeping up [00:31:26 CEST] <furq> there's no workaround if the buffer is getting data faster than it can evict it [00:33:01 CEST] <esoul> ok [00:33:47 CEST] <esoul> I make some more test [00:36:30 CEST] <esoul> One more thing. Is there a way to flush the buffer? It flushes itself somehow, but this takes soe time. [00:37:29 CEST] <c_14> It'll flush itself as fast as your output processes. [00:42:32 CEST] <esoul> Lets say the bufer is full. I get this message "real-time buffer [screen-capture-recorder] [video input] too full or near too full (76% of size: 1500000000 [rtbufsize parameter])! frame dropped!". Now I press STRG+C to stop the capture because pressing "q" doesn't work as soon as I get this message. After that I cannot start the capture again. I immediately get the "BUffer full" message. [00:42:55 CEST] <esoul> I always wait some time and suddenly I can capture again. [00:56:38 CEST] <vbgunz> no matter what I try, one instance of ffmpeg and 2 inputs, 2 outputs OR 2 ffmpeg instances of 1 input and 1 output, I only randomly ever get the microphone and monitor to rarely sync up to be the same length. otherwish I usually have to truncate the monitor recording. surely theres a better way? [00:57:22 CEST] <vbgunz> basically, I'm trying to record the microphone and monitor, 2 recordings, but they're almost always a different length from one another. [01:07:19 CEST] <sachin65> Hi [01:07:46 CEST] <sachin65> I am trying to remove a certain section in a audio file (mp3) [01:08:24 CEST] <sachin65> like remove from 1.33333 sec to 3.4444 sec [01:12:03 CEST] <sachin65> any resource or help ?? [03:06:30 CEST] <freaksed> hi all, having a rather perplexing problem with ffmpeg just stopping on my new render boxes [03:07:24 CEST] <freaksed> we just got 4 rack mount systems from Boxx, running Windows 10 Pro, but when I run a ffmpeg command on them they just stop midway through and hang there [03:07:52 CEST] <freaksed> the command im running is "ffmpeg -i <input> -r 80 -b 64000k <output>" [03:08:37 CEST] <freaksed> there is no error produced from this, it just runs, and then at some point it stops updating, the time output stops going up [03:09:14 CEST] <freaksed> note we are accessing files from a nas, but this has caused no issues on our other systems [03:13:42 CEST] <freaksed> hi all, having a rather perplexing problem with ffmpeg just stopping on my new render boxes [03:13:42 CEST] <freaksed> we just got 4 rack mount systems from Boxx, running Windows 10 Pro, but when I run a ffmpeg command on them they just stop midway through and hang there [03:13:42 CEST] <freaksed> the command im running is "ffmpeg -i <input> -r 80 -b 64000k <output>" [03:13:42 CEST] <freaksed> there is no error produced from this, it just runs, and then at some point it stops updating, the time output stops going up, i even tried a full trace output but same thing, it just hangs there [03:13:42 CEST] <freaksed> note we are accessing files from a nas, but this has caused no issues on our other systems [03:17:59 CEST] <freaksed> aving a really weird problem with ffmpeg on our new render boxes, running windows 10 pro the command is "ffmpeg -i <input> -r 80 -b 64000k <output>" we can run this fine on our desktops but on our new rack systems the process just hangs, cpu goes to 0% and the log shows no errors, i even tried a full trace output but the log just stops updating [03:47:04 CEST] <fluffrabbit> can someone explain the gradual slowdown that occurs during video encoding? [03:47:41 CEST] <limbo> cpu heating up, maybe. [03:48:05 CEST] <fluffrabbit> have you mitigated the problem? [03:48:44 CEST] <fluffrabbit> I'm using a laptop. ubuntu [03:49:19 CEST] <fluffrabbit> I guess I could get like an ice pack [03:50:03 CEST] <fluffrabbit2> woah got kicked off the channel there [03:51:01 CEST] <fluffrabbit2> test [03:52:09 CEST] <fluffrabbit> my laptop's underside is only lukewarm to the touch [03:52:55 CEST] <fluffrabbit> putting it on a platform with an atrium does not seem to improve encoding speed at all [03:56:02 CEST] <fluffrabbit> this seems to occur with h.264, mpeg-2, and vp9, as well as other codecs I'm sure. with mjpeg it seems to stay fairly consistently fast though and never falls below 11 fps for 1920x1080 [03:57:38 CEST] <fluffrabbit> in other news, CVS caffeine pills are duds [04:02:17 CEST] <fluffrabbit> I get the feeling that there is a long-standing memory leak that causes this. it's not just ffmpeg, it's a problem in libav as well [04:09:37 CEST] <fluffrabbit> interestingly, along with the time to encode each frame, the bitrate is also steadily increasing. [04:13:04 CEST] <fluffrabbit> a solution would be to cap the bitrate, but the wiki recommends the crf option as a surefire way toget good quality, which is what I'm using. afaik crf stands for "CONSTANT rate factor", not "cumulative rate factor" [04:20:22 CEST] <fluffrabbit> it's a great way to improve the viewing experience of videos: the quality gets better towards the end /sarcasm [04:43:01 CEST] <fluffrabbit> conspiracy? [05:48:40 CEST] <kiko> hi [05:49:29 CEST] <kiko> This've been a long time in IRC chat. [05:50:23 CEST] <kiko> Is anyone here? [06:11:40 CEST] <kriskropd> whats the easiest way for an idiot to make a stereo mp3 mono? [06:14:56 CEST] <kriskropd> i should mention - without muting one side of the stereo [06:15:22 CEST] <kriskropd> im trying to merge both stereo sides into a single channel - but my eyes are glazing over trying to figure it out from man [06:16:23 CEST] <kriskropd> nvm - figured it out by googling "mixdown" [06:16:25 CEST] <kriskropd> -ac 1 [06:16:49 CEST] <kriskropd> just the idiot solution i needed :D thanks guys - you are awesome - pat yourselves on the back and have a hug [08:27:33 CEST] <CoJaBo> lol. [11:03:58 CEST] <shams> Hello all [11:04:27 CEST] <shams> i'm having an issue running with ffmpeg [11:05:42 CEST] <shams> i have its binary file in /opt/ffmpeg/bin/ffmpeg but when i execute it with shell_exec() it throws an error permission denie for /opt/ffmpeg/bin/ [11:06:05 CEST] <shams> any here have idea? [11:10:23 CEST] <durandal_1707> missing permissions? [11:11:04 CEST] <shams> i tried to set its permission to chmod 755 /opt/ffmpeg/bin/ [11:11:11 CEST] <shams> but still got the same error [11:12:54 CEST] <iive> shams: make sure /opt is not on volume mounted with noexec. [11:13:13 CEST] <iive> strace might help you to find the exact operation that returns the error. [11:13:25 CEST] <shams> iive: what do you mean? [11:13:40 CEST] <shams> can you guide how can i check it? [11:14:32 CEST] <iive> `mount` might report it, or `cat /proc/mounts` [11:16:29 CEST] <shams> iive: no, not listed under cat /proc/mounts [11:17:18 CEST] <iive> your '/' root might be mounted with noexec [11:17:55 CEST] <__jack__> (which is creepy) [11:18:35 CEST] <shams> what's the solution? [11:23:47 CEST] <iive> shams: do you see `noexec` anywhere in the mounts? [11:24:41 CEST] <shams> iive: yes [11:24:53 CEST] <shams> sysfs /sys sysfs rw,nosuid,nodev,noexec,relatime 0 0 [11:24:56 CEST] <shams> proc /proc proc rw,nosuid,nodev,noexec,relatime 0 0 [11:25:15 CEST] <shams> devpts /dev/pts devpts rw,nosuid,noexec,relatime,gid=5,mode=620,ptmxmode=000 0 0 [11:25:28 CEST] <shams> tmpfs /run tmpfs rw,nosuid,noexec,relatime,size=5359196k,mode=755 0 0 [11:25:39 CEST] <shams> on above 4 places [11:26:14 CEST] <iive> these are all virtual filesystems, and none is root for /opt/ [11:26:20 CEST] <iive> so the problem is different. [11:26:41 CEST] <shams> yep [11:26:58 CEST] <__jack__> chmod 755 /opt/ffmpeg/bin/ -> chmod -R 755 /opt/ffmpeg/bin/ (or chmod 755 /opt/ffmpeg/bin/ffmpeg) [11:27:26 CEST] <shams> already did it [11:30:56 CEST] <shams> Hello Fletcher [11:31:17 CEST] <shams> i hope you can help me with an issue [11:31:46 CEST] <shams> i'm having an issue running with ffmpeg [11:31:52 CEST] <shams> i have its binary file in /opt/ffmpeg/bin/ffmpeg but when i execute it with shell_exec() it throws an error permission denie for /opt/ffmpeg/bin/ [11:32:17 CEST] <iive> shams: does it work if you start it from a shell ? [11:32:30 CEST] <iive> i mean, text console with bash [11:32:42 CEST] <shams> i'm trying it with shell at this time [11:32:49 CEST] <shams> yes, doing it with SSH [11:32:51 CEST] <__jack__> shams: namei -lo /opt/ffmpeg/bin/ [11:34:19 CEST] <shams> @jack: result [11:34:26 CEST] <shams> f: /opt/ffmpeg/bin/ drwxr-xr-x [11:34:32 CEST] <shams> root root / drwxr-xr-x [11:34:40 CEST] <shams> root root opt drwxr-xr-x [11:34:45 CEST] <shams> root root ffmpeg drwxr-xr-x r [11:34:47 CEST] <shams> oot root bin [11:35:07 CEST] <shams> this is the result of namei comm. [11:42:07 CEST] <__jack__> shams: you may try strace: strace -ft /opt/ffmpeg/bin/ffmpeg [11:52:27 CEST] <iive> i suspect SE linux might be involved... no idea how to fight that. [11:53:10 CEST] <iive> maybe `dmesg | tail` would contain something useful [11:57:53 CEST] <treaki__> hi [11:58:24 CEST] <treaki__> once i have stumpled by accident accross a page of fmmpeg with a lot of sample files [11:58:32 CEST] <treaki__> to test with (manny formats) [11:59:01 CEST] <treaki__> now i am looking for a .sac audio file to test with, and cant find this page anymore [11:59:05 CEST] <treaki__> can you help me out? [11:59:20 CEST] <c_14> samples.ffmpeg.org probably [11:59:48 CEST] <treaki__> thanks [11:59:55 CEST] <treaki__> wasnt able to find that by google [12:00:19 CEST] <BtbN> It's the first google hit for ffmpeg samples [12:01:31 CEST] <treaki__> i have fund manny thinks about setting sample rate, maybe i had a typo or my search request was to specific... [12:01:34 CEST] <treaki__> thanks for the help [12:02:36 CEST] <treaki__> but looks like no sac;( [12:02:53 CEST] <treaki__> https://www.google.de/search?q=.sac+site:https://samples.ffmpeg.org/&ie=utf… [12:04:07 CEST] <c_14> treaki__: ./archive/audio/sac3 ? [12:05:38 CEST] <treaki__> ah, thanks [12:39:57 CEST] <haarts> I'm using http://sprunge.us/gagT to convert a MKV to webM. But the resulting file is 20x larger than the original (1:00 of video, 3.2M). How come? [12:40:25 CEST] <haarts> Ultimately I want to convert whatever format I have into WebM. [12:40:35 CEST] <c_14> because you're encoding it losslessly [12:41:03 CEST] <c_14> eh, I don't think you can use -lossless and -b:v... [12:41:19 CEST] <c_14> I think you meant -b:a for the audio stream [12:42:57 CEST] <haarts> aha! That might be the bug which has been plagueing me for so long. I got such shitty quality earilier. [12:47:17 CEST] <haarts> hmm, small file now! That's good. But shitty quality. Do these settings make sense, I want to preserve quality as much as possible? http://sprunge.us/Begc [12:47:46 CEST] <c_14> set a crf [12:47:54 CEST] <c_14> you didn't which is why the quality is low [12:48:28 CEST] <furq> it says -crf 33 [12:48:52 CEST] <c_14> Ah, I'm just blind then [12:48:59 CEST] <furq> it doesn't in the first one [12:49:08 CEST] <__jack__> 33 is high [12:49:11 CEST] <c_14> If you want constant quality, you need -b:v 0 though [12:49:13 CEST] <__jack__> doc said: The CRF value can be from 063. Lower values mean better quality. [12:49:27 CEST] <c_14> Otherwise I think it'll default to constrained quality with a relatively low bitrate [12:49:31 CEST] <furq> it has that too [12:49:42 CEST] <c_14> Well [12:49:47 CEST] <furq> haarts: reduce the crf value until it's good enough [12:49:51 CEST] <c_14> Maybe I should learn to read. [12:50:12 CEST] <haarts> furq: trying 3 now. (binary search :)) [12:50:24 CEST] <furq> that's almost certainly going to be too high but ok [12:50:49 CEST] <furq> surely a binary search would lead you to 17 [12:50:53 CEST] <haarts> file gets big again. 3.2 orig vs 36 recoded. Too high for sure. :) [12:51:07 CEST] <haarts> that would be the next step. [12:51:50 CEST] <haarts> is that the only truly relevant param to tweak? [12:51:59 CEST] <furq> that and -speed [12:52:22 CEST] <furq> i assume -speed will reduce filesize and/or increase quality at the same crf [12:52:58 CEST] <furq> also libvpx's multithreading will reduce quality a little bit, but chances are it's not doing anything anyway [12:53:35 CEST] <__jack__> increase the crf, there are many values between 3 and 33 :p [12:54:24 CEST] <furq> you can also try dropping to 1-pass if it's too slow [12:54:27 CEST] <c_14> haarts: only use -frame-parallel if you're planning on decoding the video with libvpx, it'll actually only hurt quality and decoding speed when decoded by ffmpeg [12:54:42 CEST] <furq> frame-parallel isn't needed for multithreading any more is it [12:54:53 CEST] <furq> i think you just need tile-columns and threads now [12:55:09 CEST] <haarts> 17 looks good. file doubled in size. Would I have to do this tweaking with every different input file type? [12:55:17 CEST] <c_14> It does something to make it easier to decode with libvpx [12:55:21 CEST] <furq> 17 is nominally constant quality, so no [12:55:32 CEST] <furq> pick a value that looks good and it should be the right choice for most inputs [12:55:51 CEST] <furq> s/17/crf/ [12:55:55 CEST] <haarts> excellent [12:56:15 CEST] <furq> or at least it is with x264, i've still not got used to the different mappings with libvpx [12:56:35 CEST] <furq> given that 2-pass crf actually has an effect with libvpx, it's probably not safe to assume much is the same [12:58:07 CEST] <haarts> it is expected that the file size gets significantly bigger? I'm lacking a frame of reference [13:14:34 CEST] <Primal> hello! can anyone please explain how to update ffmpeg? we currently have v.2.4.3 installed. running on apache. access through putty (ssh). [13:15:33 CEST] <__jack__> huh ? what's the relation between apache, ffmpeg and ssh ? [13:16:48 CEST] <Primal> i thought that info would be necessary. [13:17:02 CEST] <Primal> i am assuming i was mistaken :) [13:30:03 CEST] <iuppiter> wow [13:30:20 CEST] <iuppiter> AMD ZEN will have 10bit decoding!! [13:32:05 CEST] <haarts> Primal: what operating system? [13:34:14 CEST] <Primal> for me, windows [13:34:42 CEST] <haarts> That I gathered from having Putty. But the system you SSH into. [13:35:30 CEST] <Primal> linux [13:35:40 CEST] <haarts> what kind of linux? [13:37:06 CEST] <Primal> 2.6.32-042stab108.2 GNU/Linux [13:37:40 CEST] <haarts> close. and what variant? Ubuntu? CentOS? Try 'cat /etc/lsb-release' [13:38:18 CEST] <Primal> haha, ubuntu 14.04 [13:38:31 CEST] <Primal> Ubuntu 14.04.4 LTS [13:39:03 CEST] <furq> http://johnvansickle.com/ffmpeg/ [13:39:12 CEST] <furq> extract that to /usr/local/bin [13:39:52 CEST] <haarts> should he just use apt? [13:40:04 CEST] <furq> there is no ffmpeg in 14.04 or in backports [13:40:11 CEST] <haarts> not sure what the latest and greatest is on Ubuntu. [13:40:14 CEST] <haarts> Wait. Wat? [13:40:21 CEST] <furq> you'd have to use a ppa and those all look crap [13:40:34 CEST] <furq> the most notable one was violating gpl until very recently, if not still [13:40:45 CEST] <haarts> omg [13:40:52 CEST] <haarts> OK, what furq said [13:41:17 CEST] <furq> ffmpeg was gone from ubuntu/debian for a couple of years because of the whole libav situation [13:41:37 CEST] <furq> fortunately no ubuntu or debian user is using a distro that's several years old [13:41:53 CEST] <furq> he said, before bursting into tears [13:41:59 CEST] <haarts> lol [13:42:25 CEST] <haarts> what are you running on your server (if any)? Me Ubuntu 16.04. [13:42:30 CEST] <furq> debian testing [13:43:13 CEST] <haarts> in 16.04 ffmpeg is there, v2.8 [13:43:32 CEST] <furq> it sure is [13:43:53 CEST] <furq> but people on LTS releases usually don't take kindly to you suggesting they upgrade four times [13:43:59 CEST] <furq> not least because it usually doesn't work [13:47:05 CEST] <Primal> so is what you're saying, for a layman like me, that i might as well not bother upgrading? [13:47:36 CEST] <furq> no [13:47:37 CEST] <furq> 12:39:03 ( furq) http://johnvansickle.com/ffmpeg/ [13:47:37 CEST] <furq> 12:39:12 ( furq) extract that to /usr/local/bin [13:47:41 CEST] <furq> that's what i'm saying [13:47:50 CEST] <Primal> haha, ok :) [13:48:22 CEST] <drastik> ffmpeg -i https://streamsite/playlist.m3u8 -c copy -bsf:a aac_adtstoasc -f mp4 out.mp4 [13:48:25 CEST] <furq> you probably also want to remove the ffmpeg you already have installed so it doesn't conflict [13:48:40 CEST] <furq> i guess you installed it from a ppa, in which case `apt-get purge ffmpeg` [13:49:41 CEST] <drastik> hi everyone i am running that command above to download a stream from a website. The problem is that ffmpeg doesnt stop when the ppl on the site stop streaming, but instead it waits for me to type q to quit [13:50:03 CEST] <drastik> anyway idea how to make ffmpeg stop on it own? [13:56:23 CEST] <drastik> anyone please? [14:02:06 CEST] <iuppiter> I've got a video and want to trim the black borders, is this doable? [14:03:57 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#crop [14:04:20 CEST] <drastik> any idea for my problem? [14:04:31 CEST] <furq> drastik: which ffmpeg version [14:04:40 CEST] <drastik> lemme check [14:05:03 CEST] <furq> i just tested and it stops for me when the m3u8 is no longer available [14:05:12 CEST] <furq> if the streaming server you're using doesn't do that then i have no idea [14:05:25 CEST] <drastik> wait [14:05:38 CEST] <furq> you could maybe try -timeout but that didn't seem to have any effect for me [14:05:56 CEST] <drastik> ffmpeg version 3.0.2-4~bpo8+1 [14:06:05 CEST] <furq> probably not that then [14:06:22 CEST] <drastik> it stay in this state for the whole night [14:06:32 CEST] <drastik> i had to manually stop it [14:06:54 CEST] <furq> i suspect the hls muxer isn't smart enough to figure it out [14:07:06 CEST] <drastik> but how to fix it? [14:07:18 CEST] <drastik> what would the timeout do? [14:07:29 CEST] <furq> you can try -timeout 10 before -i [14:07:34 CEST] <furq> but like i said, i don't think it does anything [14:07:41 CEST] <drastik> i c [14:07:48 CEST] <drastik> so any other suggestions? [14:08:00 CEST] <furq> none better than filing a feature request [14:08:08 CEST] <drastik> i c ty [15:29:53 CEST] <zyme> Dumb question, you have a mkv file, DD5.1 H.264 4.1 720P; that always freezes on the exact same scene, not on all players but one you need to use it with and several others. You can skip past this point and resume right after if you watch closely. [15:31:47 CEST] <zyme> Are there any programs around to check the file for encoding errors and easily find/ attempt to fix the file or cut the bad frames out? [15:33:18 CEST] <DHE> doesn't work that way. if there are corrupted video frames that can't be repaired (and very few people use FEC) then you have to drop everything up to the next keyframe [15:46:32 CEST] <zyme> Is there a way to check for corrupt video frames or find out if it's just the decoder? [15:48:31 CEST] <xx_pk_xx> Hey guys. I'm trying to encode bitmap subtitles. However, I'm not sure what data I need as input for AVSubtitle.data. Let's say I have an image of the subtitle. To which pix format should I transform it? I will be greatfull for any advice [16:22:42 CEST] <abumusa> where can I find an example of code that turn a sequence of images into webm file ? [17:27:48 CEST] <xrmx> hello, does anyone have a clue about how to debug Assertion ret >= 0 failed at src/libavcodec/options.c:142 ? http://paste.debian.net/783181/ [17:47:43 CEST] <DHE> xrmx: I'm guessing you specified "1.0" in a field that only accepts integer inputs [18:07:16 CEST] <xrmx> DHE, that's me trying to play a video from a video player, like i have no clue where this "1.0" is :) [18:15:01 CEST] <roxlu> Hi, I just noticed that when i stream a MP4 file (video only) to RTP, that ffmpeg does not send the PPS/SPS as the first packet. Is there a way to tell ffmpeg to first send the PPS/SPS before any other packet? [18:18:19 CEST] <roxlu> When I use the testsrc filter, I do receive a pps/sps as the first rtp-packet [18:28:02 CEST] <DHE> testsrc isn't an encoded video [18:45:49 CEST] Action: DelphiWorld neighboring kepstin somewhere! [18:46:09 CEST] Action: kepstin waves [18:49:32 CEST] <DelphiWorld> kepstin: LOL. got me? :P [19:23:17 CEST] <blb> how does one deal with the VBI lines in IMX MXF files? I know the MXF container has a vertical offset field, but it appears to be ignored [19:27:34 CEST] <xx_pk_xx> Hey guys. I'm trying to encode bitmap subtitles. However, I'm not sure what data I need as input for AVSubtitle.data. Let's say I have an image of the subtitle. To which pix format should I transform it? I will be greatfull for any advice [19:47:31 CEST] <JEEB> blb: oh there's a field for that? [19:47:47 CEST] <JEEB> I think all of the samples I've seen with the non-visual data on top didn't have anything [19:48:26 CEST] <JEEB> container cropping is a thing now so if it's sane it could be supported if it exists [19:50:47 CEST] <JEEB> oh, YOsiz [19:50:49 CEST] <blb> JEEB: using MediaInfo, I see 0x320A for horizontal offset and 0x320B for vertical offset. with this particular file, the offset being 16 [19:51:58 CEST] <blb> 0x320A and 0x320B are ignored by mxf demuxer, afaik [19:52:59 CEST] <JEEB> I might poke at it in the nearby future if I have the time. need to check the samples I have of that painful format [19:53:10 CEST] <blb> maybe i haven't seen how a container can provide cropping info to the upper api [19:58:10 CEST] <JEEB> I will have to look into it a bit more but there's at the very least the difference between codecpar's width/height and coded_width/height [19:58:51 CEST] <blb> JEEB: neat. i appreciate it [19:59:35 CEST] <JEEB> mostly I'm poking the guy behind cropping if there's a definition for the cropping box [19:59:40 CEST] <JEEB> outside of the H.264 decoder that is [20:03:22 CEST] <blb> yes. that's the one part i'm curious about. should the decoder pay attention to that or should the user? [20:03:46 CEST] <JEEB> lemme post a short discussion I just had [20:04:10 CEST] <JEEB> http://up-cat.net/p/7c10f9cd [20:04:23 CEST] <JEEB> unfortunately container cropping is currently limited to the bottom [20:05:51 CEST] <blb> dang :/ [20:06:24 CEST] <JEEB> then there's the whole side data business but then support for that would have to be added to ffmpeg cli as well [20:06:45 CEST] <JEEB> so you add cropping side data and then ffmpeg cli would read the values there [20:06:51 CEST] <JEEB> and crop accordingly [20:06:55 CEST] <JEEB> or another app like player [20:07:22 CEST] <JEEB> ...or you look at how the AVC and HEVC decoders do it [20:07:33 CEST] <JEEB> although at the demuxer level you don't have the decoded picture yet [20:07:34 CEST] <JEEB> meh [20:09:05 CEST] <blb> i'm technically using the ffmpeg api. i can decode the full 720x608 frame. after the decoding, if i knew that i should ignore the top 32 lines, i'd be fine, but that info isn't made available from the demuxer [20:09:33 CEST] <blb> the mxf field technically says 16, but it's per-field [20:09:39 CEST] <JEEB> hah! :D [20:09:55 CEST] <JEEB> but yeah, then side data would be OK for you as well [20:10:28 CEST] <blb> if the decoder did it, that'd be awesome, if not, that's ok. just as long as i could access it [20:10:56 CEST] <JEEB> seems like adding a type of side data would be the least bad way currently [20:11:17 CEST] <JEEB> but the Father Of Side Data says it might be more work than I'd be willing to do for it [20:11:28 CEST] <JEEB> http://up-cat.net/p/6c7dade3 [20:12:11 CEST] <blb> mxf is one of those containers that really likes to override what the codec says [20:12:19 CEST] <JEEB> tell me about it :P [20:12:20 CEST] <blb> i've seen aspect ratio and frame rate overrides [20:12:23 CEST] <blb> argh! [20:12:31 CEST] <JEEB> oh, I was just about to ask about aspect ratio [20:12:43 CEST] <JEEB> because I had a case of some vendor always outputting 4:3 AR'd stuff [20:12:51 CEST] <JEEB> while the content could randomly be 4:3 or 16:9 [20:12:52 CEST] <blb> aspect ratio is correct i believe, the container reads that field [20:13:25 CEST] <JEEB> ok, then it's just a case of even crappier stuff :P because the AR flag in the video stream at least was *always* 4:3 [20:14:11 CEST] <JEEB> and the aspect ratio was meant to be applied *after* the 32 samples were cropped off [20:14:49 CEST] <blb> btw, if you want to see a file like that i'm talking about, here's one https://gstreamer.freedesktop.org/data/media/incoming/mxf/opencubetech.com/… [20:15:09 CEST] <JEEB> cheers [20:15:14 CEST] <blb> media info shows it should be 720x576 with the original height being 608 [20:15:27 CEST] <JEEB> yeah, that sounds like the stuff I had seen as well [20:15:45 CEST] <JEEB> the aspect ratio part intrigues me because I distinctly remember that thing not being in the video or container correctly [20:15:45 CEST] <blb> vlc plays it with the VBI lines showing [20:16:24 CEST] <blb> i do have a file where mxf says it should be 16:9 but the video data said 4:3 [20:17:08 CEST] <JEEB> oh, and let's not forget about audio mappings [20:17:18 CEST] <blb> :S nothx [20:17:47 CEST] <JEEB> yeah, this looks like what I saw [20:17:53 CEST] <JEEB> doesn't seem to have AR data on MXF side [20:17:58 CEST] <JEEB> and 4:3 AR on video [20:19:54 CEST] <JEEB> but yeah, audio seems to be a whole separate mess. so far I've just done selection under "if MXF and either channel count is something or stream count is something" [20:20:50 CEST] <blb> separating the channels into streams... argh [20:21:25 CEST] <JEEB> and then I'm not even sure if there's metadata in there which would tell me how many channels are actually used :P [20:21:41 CEST] <JEEB> currently it's pretty much case-by-case kind of thing for me [20:25:04 CEST] <madprops> hi [20:25:18 CEST] <madprops> im trying to create a single big image from 6 long images [20:25:22 CEST] <madprops> i want to stack them vertically [20:25:34 CEST] <madprops> this is what im trying [20:25:35 CEST] <madprops> ffmpeg -i 1.png -i 2.png -i 3.png -i 4.png -i 5.png -i 6.png -filter_complex vstack output.jpg [20:25:44 CEST] <madprops> but it only seems to merge the first 2 and ignores the rest [20:25:47 CEST] <madprops> how can I do this? [20:26:51 CEST] <DHE> vstack=inputs=6 [20:26:57 CEST] <DHE> instead of just "vstack" [20:34:19 CEST] <madprops> DHE, I tried ffmpeg -i 1.png -i 2.png -i 3.png -i 4.png -i 5.png -i 6.png -filter_complex vstack=inputs=6 output.jpg [20:34:30 CEST] <madprops> I got Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [20:34:58 CEST] <madprops> the last image has a different height than the other ones [20:35:11 CEST] <madprops> don't know if that has something to do [20:39:11 CEST] <furq> it shouldn't [20:39:17 CEST] <furq> All streams must be of same pixel format and of same width. [20:39:42 CEST] <madprops> they are [20:40:08 CEST] <furq> you might need -pix_fmt yuvj420p [20:40:28 CEST] <furq> actually if it worked with two images then probably not [20:50:27 CEST] <kepstin> madprops: can you pastebin the *complete* ffmpeg output please? [20:50:58 CEST] <kepstin> (when ffmpeg prints that 'maybe incorrect parameters' message, it usually has printed earlier what the *actual* error is) [20:51:09 CEST] <madprops> sure [20:51:47 CEST] <madprops> http://gsick.com/paste/110/ [20:52:18 CEST] <furq> [mjpeg @ 0000000000567400] [IMGUTILS @ 000000000022f000] Picture size 1920x156476 is invalid [20:52:21 CEST] <furq> nice [20:52:52 CEST] <furq> the max resolution for jpeg is 65,535 [20:53:48 CEST] <madprops> oh... [21:24:23 CEST] <DHE> they're actually 30,000 pixels tall? didn't realize that.. [21:55:18 CEST] <shincodex> full human body image to scale maybe [00:00:00 CEST] --- Wed Jul 20 2016

1 0

ffmpeg-devel.log.20160718
by burek 19 Jul '16

19 Jul '16

[01:34:37 CEST] <cone-829> ffmpeg 03Aman Gupta 07master:9ca967145827: avcodec/ccaption_dec: implement positioning for closed captions [01:51:30 CEST] <cone-829> ffmpeg 03Aman Gupta 07master:a49ed9e32d5f: avcodec/ccaption_dec: default rollup to row 10 [02:33:36 CEST] <TheRyuu> https://github.com/TheRyuu/FFmpeg/commit/7b8be2f63f2d4df1d15f51a3639331a260… <---does this look about right for adding an option like this? [02:42:44 CEST] <Chloe> TheRyuu: maybe --enable-win-cfg instead? (shorter), and disable control_flow_guard if it's not windows [02:43:31 CEST] <Chloe> TheRyuu: also remember to do add_cflags [02:43:56 CEST] <TheRyuu> why not check_cflags, it's only avaiable on vs2015 [02:44:43 CEST] <Chloe> ohwait yeah, that's fine [02:45:36 CEST] <TheRyuu> where and why would it have to be explicltly disabled if I'm not on Windows? it only gets touched inside of a block of code that only gets triggered with msvc [02:47:30 CEST] <TheRyuu> unless I'm misunderstanding how this works, as long as you never pass the option then it shouldn't have any side effects [02:47:48 CEST] <Chloe> I was just thinking that there's no point checking for the flag if it's not on windows for performance (two less checks), but if you're going to explicitly ask for it then it I guess doesn't matter, it wont affect systems without the option. [02:48:16 CEST] <TheRyuu> I mean the checks only get triggered on Windows anyway [02:48:26 CEST] <TheRyuu> since it's under the big msvc if [02:48:46 CEST] <Chloe> oh right [12:19:26 CEST] <JEEB> > ffv1 in mov [12:19:27 CEST] <JEEB> my face [12:19:33 CEST] <JEEB> didn't know it was specified [12:20:29 CEST] <nevcairiel> JEEB: http://ffmpeg.org/~michael/ffv1.html#in-isoiec-14496-12-mp4-file-format [12:20:58 CEST] <JEEB> wow [20:05:10 CEST] <omerjerk> hey [20:05:51 CEST] <omerjerk> I need to use this function in my code - https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/asvdec.c#L72 [20:06:02 CEST] <omerjerk> I basically need to read bits in the reversed order. [20:06:10 CEST] <omerjerk> Michael suggested to use this function. [20:06:26 CEST] <omerjerk> how should I use this ? [20:16:15 CEST] Action: michaelni wants to reply but sees "omerjerk has quit (Quit: Page closed)" [20:34:24 CEST] <Chloe> Where do I add documentation for the options of an avformat? [22:37:14 CEST] <cone-544> ffmpeg 03Josh de Kock 07master:daf421d2ab6f: MAINTAINERS: add myself for libopenmpt [23:27:14 CEST] <cone-544> ffmpeg 03Zhao Zhili 07master:450cf400fc4e: avutil/mem: fix wrong comment [00:00:00 CEST] --- Tue Jul 19 2016

1 0

ffmpeg.log.20160718
by burek 19 Jul '16

19 Jul '16

[00:00:08 CEST] <JEEB> if you due to any legal requirements need a more limited binary, it's up to you to configure it [00:00:17 CEST] <JEEB> the configuration is available, though [00:00:32 CEST] <JEEB> (you can enable/disable components as required) [00:01:53 CEST] <furq> there is also an important distinction between codecs which someone issues paid licenses for, and codecs which might incorporate patented technology [00:02:00 CEST] <furq> because the latter seems to be pretty much all codecs [00:02:25 CEST] <egnun> Ok. o_O [00:02:56 CEST] <furq> it's extremely unlikely you'll get sued for only using theora, or that the plaintiff has a chance of winning a case, but not impossible [00:03:38 CEST] <JEEB> and for on2/google's formats there's mostly just google's nice whispers (they do not say they will defend you against any litigation, though, of course) [00:03:47 CEST] <furq> http://en.swpat.org/wiki/Ogg_Theora [00:03:51 CEST] <furq> Patent licensing group MPEG LA have made vague, unsubstantiated claims that all video formats infringe their patents: [00:03:54 CEST] <furq> i like that sentence [00:04:14 CEST] <furq> mpeg-la sure are a cheery bunch [00:04:35 CEST] <JEEB> well, there's even more cheerful bunches out there [00:04:43 CEST] <JEEB> like those actively killing HEVC [00:05:35 CEST] <furq> i've not been keeping up with the hevc licensing saga [00:05:45 CEST] <JEEB> MPEG-LA made a patent pool with simple licensing and of course some business people decided that they wanted a bigger pie. so now we have (at least) two patent pools and I think technicolor (?) separately [00:06:25 CEST] <JEEB> essentially this makes HEVC be in a very funky position even if the two other things end up crawling back to MPEG-LA [00:06:35 CEST] <furq> is hevc used anywhere yet [00:06:46 CEST] <JEEB> in some broadcast and VOD things [00:07:22 CEST] <__jack__> (torrents use it :3) [00:07:28 CEST] <JEEB> a Japanese channel has been airing it from 2014, and netflix made the (totally marketing) decision to use it for 2160p (which are only available for specific DRM'd plastic boxes) [00:07:49 CEST] <JEEB> and by now there are test/production channels with it around the world (a dozen or so I guess?) [00:07:54 CEST] <furq> i meant other than torrents encoded with x265 veryfast at far-too-small sizes [00:08:20 CEST] <furq> because someone read on the internet that x265 is 100% better than x264 and then immediately stopped reading any information forever [00:08:53 CEST] <JEEB> so there is usage of it being around, but it totally seems like the licensing status is moving people looking for alternatives [00:09:22 CEST] <furq> it would be funny if AV1 actually delivered something usable next year and hevc ended up dead in the water [00:09:24 CEST] <JEEB> hopelly the AOM thing leads to a more proper spec+ref impl-based development model for the google things [00:09:30 CEST] <furq> or AOM even [00:10:14 CEST] <JEEB> because when people were yelling at how closed HEVC was I was seeing vp9 development going on pretty much behind closed doors and HEVC was being openly developed on the JCT-VC mailing list [00:10:48 CEST] <furq> does vp9 have a spec yet [00:10:57 CEST] <JEEB> I think they finally made a bitstream guide kind of thing [00:11:16 CEST] <JEEB> due to going SW-only at first there's some bugs that ended up being The Way [00:27:08 CEST] <bklang> Hi all. Im trying to find a way to blend 5 frames in a row together. The idea is to reduce motion in a time-lapse video. Ive found the tblend filter, but that only seems to blend 2 frames, which doesnt appear to be enough for me. Are there any ways to do this with the CLI, or do I need to write something with libav? [01:25:27 CEST] <thebombzen> the biggest problem I have with x265 is that it's too slow [01:25:48 CEST] <thebombzen> like I'm willing to use x264 for everything still because I like my videos to encode without waiting until the day after next tuesday [01:26:35 CEST] <thebombzen> the downside of modern video codecs is that they all encode too slowly. which I guess they didn't really care about because who wants to use HEVC other than big multinational corporations who can pay for renderfarms, n'est-ce pas? [01:27:04 CEST] <thebombzen> I mean, an ordinary user with limited UL bandwidth couldn't /possibly/ care about encoding time [01:27:27 CEST] <thebombzen> sorry I'm just salty that HEVC takes forever. [01:29:44 CEST] <JEEB> also do note that in '06 x264 was slow [01:29:51 CEST] <JEEB> CPUs were a lot slower etc [01:30:29 CEST] <furq> well then it's a good job cpus have been getting so much faster recently [01:31:31 CEST] <JEEB> it's not only CPUs but also the optimizations on the encoding side, which is what comes with time. also new SIMD instruction sets bring out new possibilities for optimization [01:33:41 CEST] <JEEB> in any case, x265 as a project has its own issues (no real community that would contribute, format being under uncertain terms etc) but also it's kind of weird to compare an encoder three years after the inception of a format against an encoder that has been developed for a format that was initially finished in '03 (some profiles happened a year or two later, such as high profile which added mainly 8x8 dct) [01:40:27 CEST] <thebombzen> I guess. H.264 is 13 years old. I guess that HEVC encoders will be as good in 10 years as H.264 encoders are nowadays [01:40:40 CEST] <thebombzen> but it feels awkward to use 2003 technology, y'know? [02:17:27 CEST] <egnun> Hello, it's me again. [02:18:00 CEST] <egnun> I wanted to convert a video from .webm to .ogv [02:18:14 CEST] <egnun> It said that the Theora encoder is missing. [02:18:37 CEST] <limbo> install a version of ffmpeg with that encoder. [02:18:47 CEST] <egnun> When I execute [02:18:47 CEST] <egnun> $ ffmpeg -codecs [02:18:47 CEST] <egnun> it says [02:18:47 CEST] <egnun> D.V.L. theora Theora [02:18:49 CEST] <limbo> or, whatever set of gstreamer plugins you need. [02:19:16 CEST] <egnun> Well, I compiled ffmpeg myself. [02:19:32 CEST] <egnun> But I am wondering why it didn't compile the theora encoder. [02:20:21 CEST] <egnun> I used following configure parameters: [02:20:21 CEST] <egnun> --enable-gpl --enable-version3 --disable-nonfree [02:20:43 CEST] <egnun> Could "--enable-version3" have caused the problem? [02:21:49 CEST] <darkapex> try with --enable-libtheora [02:21:52 CEST] <darkapex> https://trac.ffmpeg.org/wiki/TheoraVorbisEncodingGuide [02:22:20 CEST] <darkapex> the wiki is quite good btw [02:24:20 CEST] <furq> egnun: https://www.ffmpeg.org/general.html#Video-Codecs [02:24:27 CEST] <furq> anything with an E requires an external library [02:26:34 CEST] <egnun> furq: Well, there seems to be an installation of a package "libtheora". [02:27:28 CEST] <darkapex> egnun: yes it's an external library, you'll need to install it from your package manager if it hasn't been already (but as the wiki said the native ffmpeg encoder is experimental and not advised to use for production) [02:28:17 CEST] <egnun> So, because the native ffmpeg encoder is unstable, it uses the external one? [02:29:07 CEST] <darkapex> I don't know which one it uses by default, but if you want to force the native one specify '-codec:a vorbis -strict experimental' [02:29:31 CEST] <darkapex> i.e. if installing libvorbis is a problem, retry the command with this ^ and let us know if it still gives an error [02:29:31 CEST] <furq> there is no native theora encoder [02:29:51 CEST] <darkapex> oh shet why did i start talking about vorbis my bad [02:30:05 CEST] <darkapex> you'll need to install libtheora [02:30:07 CEST] <furq> well you'll want libvorbis as well if you want to encode ogv [02:31:09 CEST] <furq> you probably also want libvpx and libopus if you want the most useful license-free codecs [03:01:05 CEST] <egnun> How can I tell pkg-config to use one specific library? [03:01:25 CEST] <furq> huh [03:02:17 CEST] <egnun> Ah, nvm [03:02:21 CEST] <egnun> nevermind [03:03:07 CEST] <egnun> I have a "libopusfile" installed, but there is no "libopus" in "/usr/lib64/pkgconfig" [03:03:40 CEST] <egnun> There is no *opus*.pc in that directory [03:04:13 CEST] <egnun> I just could compile ffmpeg without opus [03:04:38 CEST] <furq> you want libopus, not libopusfile [03:04:58 CEST] <egnun> Well, without opus encoding. [03:08:13 CEST] <egnun> I have a libopus [03:08:25 CEST] <egnun> But no file that is called "libopus.pc" [03:08:33 CEST] <furq> do you have the devel package installed [03:08:53 CEST] <furq> presumably libopus-devel but i have no idea what fedora calls it [03:09:30 CEST] <egnun> Well, you'd think that there is one. [03:09:33 CEST] <egnun> But there is none.^^ [03:09:43 CEST] <egnun> I've already looked for it. ^^ [03:09:49 CEST] <furq> nice [03:10:34 CEST] <egnun> Ahh1 [03:10:36 CEST] <egnun> Ahhh!! [03:10:44 CEST] <egnun> It's not called "libopus" but just "opus" [03:12:06 CEST] <furq> how nice of them [03:12:24 CEST] <egnun> ^^ [03:12:29 CEST] <egnun> I should file a report. [03:12:33 CEST] <egnun> But not now! [03:12:41 CEST] <egnun> Now I am going to install ffmpeg! [03:12:41 CEST] <furq> i like how fedora's package info site takes you to three different domains before giving you a file list [03:12:55 CEST] <egnun> Which site? [03:13:23 CEST] <furq> https://admin.fedoraproject.org/pkgdb/package/rpms/opus/ [03:14:15 CEST] <furq> oh never mind apps.fedoraproject.org has a better search [03:14:22 CEST] <furq> it's nice of them to not bother getting that to #1 on google [03:15:36 CEST] <egnun> ^^ [03:15:44 CEST] <egnun> Who uses google anyway? [03:15:53 CEST] <egnun> DuckDuckGo > * [03:16:16 CEST] <furq> why not just use bing directly [03:16:31 CEST] <egnun> & [03:16:34 CEST] <egnun> Yahoo. [03:17:17 CEST] <furq> yahoo is also just bing [03:17:51 CEST] <egnun> Ya, hoo cares? [03:18:21 CEST] <furq> google know everything about me anyway [03:18:29 CEST] <furq> i might as well take advantage of that fact and get better search results [03:19:09 CEST] <egnun> Well, I get quite good results via DDG too. [03:19:38 CEST] <egnun> And if it doesn't give me the results I am looking for, I can still use google [03:19:41 CEST] <egnun> even via DDG. [03:20:25 CEST] <furq> that all seems plausible [03:20:34 CEST] <furq> but what good is irc if not for making fun of people's minor life choices [03:21:37 CEST] <egnun> I don't hate you [03:22:24 CEST] <furq> i feel like we've made a real breakthrough here [03:23:28 CEST] <egnun> :D [04:19:49 CEST] <egnun> ffmpeg seems to be working so far! \o/ [04:19:56 CEST] <egnun> Thank you all for your help! [04:34:45 CEST] <ycon_> Hi all, I'd like to convert a .mp4 video to AVI, mpeg,mpeg4 or mkv [04:35:51 CEST] <ycon_> I mean I want to convert from .m4v. [04:35:52 CEST] <ycon_> But when i did ffmpeg -i video1.m4v to output.avi the quality was un-readable [04:40:47 CEST] <egnun> So the converting process worked without problems? [04:42:15 CEST] <egnun> Do you need to convert it to .mp4? [04:42:15 CEST] <egnun> You should try to convert it to .ogv instead. [06:32:26 CEST] <seanrdev> So I have some videos that get somewhat choppy during (action) playback. Now it's not a huge deal however I have a skylake 4 core processor and an nvidia 965m gtx card. I have used multiple players and I'm wondering what needs to be adjusted to stop choppyness during intense scenes in video playback. [06:35:27 CEST] <seanrdev> oh also the video files are stored on a ssd vnand drive with read speeds at 22,587 MB/s [08:19:15 CEST] <Betablocker> moin moin [08:19:32 CEST] <Betablocker> is there any REST api for ffmpeg ? [09:24:48 CEST] <egnun> Betablocker: You can REST asure, that I have no clue. [09:59:32 CEST] <Fjorgynn> jaså [10:28:24 CEST] <roxlu> hey, when I use --sdp_file stream.sdp, the generated SDP file adds the string "SDP" as the first line. Why is that? [11:57:16 CEST] <Galadriel> Hi, i tried to compile ffmpeg but there are a lot of (declared at libavcodec/avcodec.h:4763) [-Wdeprecated-declarations] warnings. Any idea? [11:57:35 CEST] <Galadriel> this is the newest version [12:03:08 CEST] <Galadriel> anyone [12:03:09 CEST] <Galadriel> ? [12:12:11 CEST] <markvandenborre> I have two video streams that I'm putting together into a pretty picture-in-picture stream [12:12:51 CEST] <markvandenborre> I would like to add the exact (ntpdate) time into that stream, while causing as little sync or delay problems as possible [12:19:08 CEST] <JEEB> Galadriel: internal deprecated stuff usage is not somethign you should care about [12:19:16 CEST] <JEEB> unless you want to develop FFmpeg itself [12:21:12 CEST] <Galadriel> @JEEB: thanks for your answer .. when i encode a video i also have in the log [12:21:14 CEST] <Galadriel> [webm @ 0x2770880] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. [12:33:18 CEST] <Galadriel> JEEB: is this ok ? [13:13:19 CEST] <moala> hey, I'm trying to decode AC3 into an LtRt format: https://sourceforge.net/projects/audiotestfiles/files/Audacity%20example%20… [13:13:49 CEST] <moala> ffmpeg -i 5.1_de.ac3 -acodec libmp3lame -ab 320k -ac 2 5.1_de.mp3 [13:14:36 CEST] <moala> but what I obtain is the rear channels mixed to the front channels, not the matrixed ("ProLogicII") version [13:14:53 CEST] <moala> any idea about how to obtain that? [13:18:04 CEST] <BtbN> -c:a libmp3lame -b:a 320k -af aresample=matrix_encoding=dplii -ac 2 [13:24:10 CEST] <moala> BtbN, wow, thanks a lot! [13:24:42 CEST] <BtbN> might even drop the ac, the aresample filter should output 2 channels [13:25:40 CEST] <moala> yep, works [17:11:18 CEST] <simulacr> curl, wget etc supports proxy env without protocol prefix, but ffmpeg just _ignore_ them (even without error/warning!), wtf? [17:28:10 CEST] <simulacr> ah 2259, nevermind [19:14:15 CEST] <jfhbrook> I have kind of an odd question: I'm working on a mixed media server that does resizes and conversions of stills and video on-the-fly [19:14:29 CEST] <jfhbrook> which is great because we don't know what sizes of videos, etc., that people will want a'priori [19:14:45 CEST] <jfhbrook> but there are things we *do* know, like we know what format conversions our system supports [19:14:59 CEST] <jfhbrook> and we have problems with a thundering herd when content goes live [19:15:25 CEST] <jfhbrook> so my question, I guess, is: what kinds of pre-processing can I do to improve performance of the final resize? [19:15:49 CEST] <jfhbrook> like, if I knew aspect ratios and formats, would it be useful to pre-convert to the right ratio and format, and then do resizing on-demand? [19:15:53 CEST] <jfhbrook> or would that be a waste? [19:16:09 CEST] <jfhbrook> and are there resizes I can do that would make it easier to do a further resize later? [19:56:37 CEST] <soulshock> you can do a filter-chain to resize only once for each target resolution and create many bitrates, i.e. -i input.mp4 -filter_complex "scale=X:Y, split=3[out1][out2][out3] -map [out1] -b:v 1000k out1.mp4 -map [out2] -b:v 500k out2.mp4 -map [out3] -b:v 250k out3.mp4 [20:55:30 CEST] <jfhbrook> soulshock: the problem is I don't know the resizes a'priori [20:56:20 CEST] <jfhbrook> soulshock: so I can, say, convert from mp4 to webm, and I could choose some pre-resizes that are convenient for getting other sizes off, but I can't actually pre-render any of the actual resizes because they're requested on-the-fly via path params [20:57:06 CEST] <jfhbrook> so like, /media/abc123/w_100,h_50/foo.webm [20:57:14 CEST] <kepstin> jfhbrook: might be best to just look at your stats and just pre-render any cases that are really common [20:57:47 CEST] <klaxa> offering users too much customization might also backfire [20:58:11 CEST] <klaxa> imagine someone requesting /media/abc123/w_99999999999,h_99999999999/foo.webm [20:58:29 CEST] <kepstin> if you want to reduce the work you do scaling, you should make use of the fact that all video players usually have scalers builtin, often gpu accellerated :) [20:58:38 CEST] <jfhbrook> klaxa, that's the reality of the world, for various reasons [20:59:10 CEST] <jfhbrook> ie, this is an existing API contract put in place because designs refuse to commit to preset dimensions [20:59:33 CEST] <jfhbrook> and we're trying to extend the service to be able to handle mp4 and webm in addition to jpg, png and (potentially animated) gif [21:01:02 CEST] <klaxa> in that case how about re-encoding every video to half-, quarter- and eight-size and use the next bigger one to generate your requested output? [21:01:17 CEST] <kepstin> one thing I can think of that might provide a useful speedup is to transcode the source file to some format that's low cpu to decode, to reduce cpu usage in final encode. [21:01:22 CEST] <klaxa> that obviously only speed up requests lower than half your max resolution [21:01:25 CEST] <jfhbrook> klaxa, yeah, wondering if that's a viable strategy [21:01:51 CEST] <jfhbrook> also wondering if it's cheaper to resize while sticking to a common format, vs resizing and changing format at the same tim e [21:01:54 CEST] <jfhbrook> time*( [21:02:01 CEST] <jfhbrook> ie, is resizing mp4 to mp4 faster than mp4 to webm? [21:02:12 CEST] <klaxa> highly depends on codec settings [21:02:13 CEST] <jfhbrook> given the same crop/resize transforms? [21:02:14 CEST] <klaxa> and codecs used [21:02:16 CEST] <jfhbrook> hm [21:02:32 CEST] <jfhbrook> so let's say the kind of mp4s and webms that average users would upload [21:02:34 CEST] <klaxa> you will probably want x264 and vp8 [21:02:36 CEST] <kepstin> jfhbrook: the resizing is constant based on video size, no dependencies on format at all [21:03:02 CEST] <kepstin> jfhbrook: (well, png + gif are special, since they're changing to rgb/palette as well) [21:03:15 CEST] <klaxa> also keep in mind that every re-encode loses information [21:03:48 CEST] <jfhbrook> hmmmm [21:04:23 CEST] <klaxa> i think generally speaking x264 outperforms vp8 [21:04:30 CEST] <kepstin> honestly, I'd just encode to some common presets, and serve up a "close" size to what was requested, and let the user's player scale it if needed. [21:04:33 CEST] <klaxa> in encoding [21:05:17 CEST] <jfhbrook> yeah kepstin that's an interesting one, because at one point in the past we *were* doing pre-cuts, but then we'd get a list of 80 cuts that someone absolutely needed and.......yeah. [21:05:26 CEST] <kepstin> but I guess you might have some business requirements that preclude that. [21:05:27 CEST] <jfhbrook> and that was *before* we were talking about changing format on the fly [21:05:35 CEST] <jfhbrook> yeah kepstin [21:08:14 CEST] <jfhbrook> hmmmmmm [21:08:31 CEST] <jfhbrook> so it sounds like maybe for v1 trying to pre-cache crops isn't the best idea [21:08:40 CEST] <furq> you could maybe try playing with swscale's options or use zscale for scaling [21:08:54 CEST] <jfhbrook> unless we come up with a way for users to specify what they want to pre-render on upload [21:08:55 CEST] <furq> swscale has a "fast_bilinear" scaling algorithm which could help [21:08:58 CEST] <jfhbrook> which might be doable [21:09:01 CEST] <furq> although i expect it'll degrade the quality [21:09:12 CEST] <jfhbrook> I see [21:09:15 CEST] <kepstin> I suspect that the biggest part of your overhead is in the video encoding itself, particularly with vp8 [21:09:18 CEST] <furq> but if you're doing webm the problem is presumably the encoding [21:09:27 CEST] <furq> yeah what he said [21:09:35 CEST] <kepstin> so a faster scaler will provide slight speedups, but it's really not the biggest target for optimization [21:09:52 CEST] <jfhbrook> kepstin: oh interesting! So in this case pre-converting and then resizing later would be a win? [21:09:57 CEST] <kepstin> in order, it's probably encoder, then decoder, then scaler [21:10:09 CEST] <furq> no? [21:10:10 CEST] <kepstin> no, because if you scale then you need to do the encode again [21:10:21 CEST] <furq> you need to decode and reencode to scale [21:10:44 CEST] <furq> it's not like jpeg where you can do it losslessly [21:11:16 CEST] <jfhbrook> oh, I see [21:11:29 CEST] <furq> i'm not really an expert in doing this at such a large scale, but i can't see an obvious solution other than throwing more hardware at it [21:11:44 CEST] <jfhbrook> so in this case the suggestion is to pick codecs that are cheak to encode/decode [21:11:48 CEST] <jfhbrook> cheap* [21:11:59 CEST] <furq> well for the web you're pretty much stuck with h.264 and vp8/9 [21:12:06 CEST] <jfhbrook> I see [21:12:10 CEST] <furq> i wouldn't use vpx if you don't have to though [21:12:13 CEST] <furq> libvpx is very slow [21:12:28 CEST] <jfhbrook> yeah, well, downstream consumers XD [21:12:34 CEST] <jfhbrook> "Requirements" [21:12:39 CEST] <kepstin> jfhbrook: if you have lots of io bandwidth/storage, it might make sense to transcode the source files to some format that has minimal or no loss, and is fast to decode [21:12:42 CEST] <jfhbrook> still, I'll note this in my, well, notes [21:12:46 CEST] <kepstin> then encode from that [21:12:53 CEST] <jfhbrook> kepstin: oh? what kinda format would be ideal for that? [21:13:05 CEST] <furq> rawvideo would be ideal but you're talking a huge amount of storage space [21:13:12 CEST] <kepstin> depends on the balance of bandwidth to cpu usage you want [21:13:22 CEST] <furq> there is some lossless yuv format which is optimised for fast decoding, but i forget which one it is now [21:13:25 CEST] <furq> maybe utvideo? [21:13:30 CEST] <kepstin> if you have lots of storage/bandwidth, try something like huffyuv, utvideo [21:14:12 CEST] <kepstin> you could also try just encoding to a fairly high quality h264 via x264 with -tune fastdecode [21:14:19 CEST] <furq> for reference, one hour of 720p yuv420p rawvideo is ~125GB [21:14:31 CEST] <furq> so you'd need a fair old amount of storage space to get away with that [21:14:34 CEST] <Mavrik> furq, huffyuv? [21:14:57 CEST] <furq> i thought huffyuv was fast to decode by virtue of being a bit crap [21:15:15 CEST] <kepstin> if you only do it for recently uploaded videos that you expect a lot of requests for, and throw out the temp file for older videos, a raw or fast lossless format might make sense [21:15:19 CEST] <DHE> it's lossless though, isn't it? [21:15:25 CEST] <furq> sure [21:15:51 CEST] <jfhbrook> hmmmm [21:16:01 CEST] <kepstin> furq: well, it's super simple, lossless, very fast. It's basically just a format for use when you would otherwise use raw video, but don't have enough io bandwidth :) [21:16:03 CEST] <Mavrik> furq, yes, it is [21:16:14 CEST] <Mavrik> But it's still order of magnitude reduction of size :) [21:16:33 CEST] <furq> yeah but i'm pretty sure there's a more efficient lossless yuv codec optimised for fast decode [21:16:33 CEST] <jfhbrook> so the trade-off there, in my mind, is less disk space and more bandwidth, since we're using s3 [21:16:53 CEST] <kepstin> ouch, s3? bandwidth is kinda expensive there. [21:17:18 CEST] <furq> i hope you're not doing the whole thing on aws [21:17:39 CEST] <kepstin> if you are doing it all on aws, and want to try this sort of 'temp raw file' thing, it might make sense to put the temp files on an instance store or storage volume instead of s3 [21:18:15 CEST] <jfhbrook> furq: yeah, s3 and ec2 [21:18:32 CEST] <furq> actually i forgot amazon have that transcoding service now [21:18:35 CEST] <furq> is that any good [21:18:41 CEST] <jfhbrook> I think we cache files on-disk cause chances are good there will be multiple renders of the same file [21:18:54 CEST] <furq> i sort of assumed it would be ridiculously expensive because it's aws [21:18:56 CEST] <jfhbrook> I read into it a bit furq and I'm far from an expert (surprise!) but it sounds *okay* [21:19:08 CEST] <jfhbrook> yeah, I kinda don't pay attention to cost (not MY wallet oho!) [21:19:27 CEST] <kepstin> I assume the amazon transcoding stuff is pretty much just they have a bunch of ec2 servers that do transcodes for you, hooked up into their queue and messaging stuff. [21:19:38 CEST] <jfhbrook> yeah, it's all SNS/SQS [21:19:44 CEST] <furq> i assume it's all hardware encoding [21:19:52 CEST] <jfhbrook> zencoder is similar except there's an http interface for interacting with the jobs [21:20:20 CEST] <kepstin> you can do your own hardware encoding on aws via the gpu instances (nvenc), but iirc. it's actually cheaper to do cpu encoding [21:20:24 CEST] <furq> oh [21:20:29 CEST] <furq> yeah apparently they're using x264 [21:20:31 CEST] <jfhbrook> yeah I was reading that somewhere [21:20:36 CEST] <furq> that doesn't seem worthwhile then [21:21:29 CEST] <jfhbrook> yeah, I mean, in my case I feel like we already have a bunch of media servers doing imagemagick stuff, making the same servers do ffmpeg stuff sounds sane, minus imagemagick's penchant for taking up *all* your system ram [21:22:16 CEST] <furq> i take it you're doing more than just resizing with imagemagick [21:22:20 CEST] <TD-Linux> I do a lot of encoding on ec2 and it works fine. encoding through EFS also works fine if you don't want s3 [21:22:23 CEST] <jfhbrook> resizes and crops, furq [21:22:39 CEST] <TD-Linux> that said these are very slow encoding jobs [21:22:44 CEST] <kepstin> resizes and crops shouldn't be that bad, unless you're talking really big images... [21:23:15 CEST] <jfhbrook> that's true kepstin but we sometimes have multi-meg animated gifs we have to deal with [21:23:23 CEST] <jfhbrook> like the kind that *should* be videos [21:23:43 CEST] <furq> i thought the cool kids used gifsicle for that [21:23:48 CEST] <jfhbrook> I mean [21:23:49 CEST] Action: kepstin definitely put some image size, ram, and *run time* limits on his imagemagick processors, and if people exceed them, well, too bad :/ [21:23:51 CEST] <jfhbrook> I'm not that cool XD [21:24:31 CEST] <kepstin> aside from that one time I set the run time limit too small, and it started affecting some images from customers we actually care about ;) [21:25:06 CEST] <jfhbrook> heh [21:25:40 CEST] <jfhbrook> yeah I mean, I'm just thinking we eyeball it based on usage, like trade off lack of memory/disk space for imagemagick vs available memory/disk for ffmpeg [21:26:24 CEST] <kepstin> keep in mind that x264 can use a *lot* of ram in the slower presets, since it keeps around a lot of raw frames as reference images. [21:26:24 CEST] <jfhbrook> speaking of, nothing I should be worried about as far as using stdio for this vs the fs, right? [21:26:31 CEST] <jfhbrook> oh, good to know [21:26:36 CEST] <jfhbrook> we might want to make the boxes bigger [21:26:55 CEST] <kepstin> but if you're using faster presets (which I assume you are), that's not as much of a problem [21:27:05 CEST] <kepstin> still, you should measure it [21:27:13 CEST] <jfhbrook> haven't quite gotten that far yet kepstin XD still in the discovery phase [21:27:33 CEST] <jfhbrook> so like, rn I'm detailing what bonus/challenges ffmpeg offers over amazon's transcoder or zencoder [21:27:43 CEST] <kepstin> we've had people come in here before complaining that video encoding was causing an OOM on their little boxes :) [21:27:46 CEST] <jfhbrook> trying to get notes on avconv too, just found a few random blog posts though---sounds like bad blood? [21:28:11 CEST] <kepstin> avconv is just the ffmpeg tool, renamed, and with the options arbitrarily changes in incompatible ways [21:28:36 CEST] <jfhbrook> yeah, that seems to be my memory from the last time I trued using avconv (because debian) [21:29:33 CEST] <kepstin> ffmpeg and avconv command line tools are very much tied to the ffmpeg vs. libav versions of the libraries. [21:29:39 CEST] <furq> debian is back in ffmpeg now [21:29:44 CEST] <jfhbrook> nice furq [21:29:48 CEST] <soulshock> ffmpeg -i input.mov -pass 1 -pix_fmt yuv420p -profile:v high -c:v libx264 -f mp4 NUL sets profile to "main" for some reason: [libx264 @ 00000000037c2e60] profile Main, level 4.1 [21:29:51 CEST] <furq> debian is back on ffmpeg, rather [21:29:53 CEST] <jfhbrook> from a clueless consumer standpoint that's definitely good news [21:29:54 CEST] <furq> or ffmpeg is back in debian [21:30:04 CEST] <soulshock> is it impossible to use profile high in 1st pass? [21:30:32 CEST] <kepstin> soulshock: you're working under a misunderstanding of x264's profile handling [21:30:44 CEST] <soulshock> that's highly probable [21:30:56 CEST] <kepstin> soulshock: x264 fills the profile field in the result based on the features actually used in the encoded video stream [21:31:06 CEST] <kepstin> soulshock: the -profile option only sets a "maximum limit" [21:31:27 CEST] <kepstin> soulshock: when doing a 1st pass encode, you aren't supposed to use the resulting video file, so x264 disables some stuff to make it faster [21:31:44 CEST] <kepstin> soulshock: as a result, it might not be the same profile as the 2nd pass [21:32:26 CEST] <soulshock> aha ok. [21:32:39 CEST] <soulshock> and the -preset option, does that also not apply? [21:33:13 CEST] <kepstin> you should usually use the same -preset option on first and second pass [21:33:29 CEST] <kepstin> since it can change some settings that affect the bitrate prediction, iirc. [21:33:56 CEST] <soulshock> ok. that was my understanding. cool [21:34:57 CEST] <kepstin> soulshock: also take a look at the description of the "--slow-firstpass" option to x264, to see what x264 disables in the first pass by default. [21:36:48 CEST] <soulshock> ok [21:37:25 CEST] <jfhbrook> thanks y'all, and I'm sure I'll be in here at a later date all "omg how u video lol" once this project gets greenlit [21:37:42 CEST] <jfhbrook> either that or my coworker that totally knows c++ will insist on writing native bindings and I won't be on it ¯\_(Ä)_/¯ [21:38:50 CEST] <kepstin> soulshock: keep in mind that using a faster preset in x264 will also result in turning off some features, e.g. I don't think you'll ever get 'high' output with -preset ultrafast. [21:40:45 CEST] <furq> ultrafast is always baseline isn't it [21:40:50 CEST] <soulshock> kepstin yeah. I'm experimenting with -preset veryslow because I want to build files I can use for psnr comparison [21:41:24 CEST] <soulshock> i.e. copy the netflix idea of per-title-encodingf [21:42:56 CEST] <kepstin> hmm, ultrafast does appear to be a superset of the options x264 sets for baseline, yeah [21:43:52 CEST] <__jack__> speaking of "netflix per title encoding", what's the technical stuff behind that commercial speach ? [21:46:51 CEST] <__jack__> is there anything more than just preset & crf, instead of constant bitrate, I mean [21:47:19 CEST] <DHE> constant quality with -qp, variable quality with -crf, and variable bitrate with -b:v [21:47:46 CEST] <kepstin> -qp isn't constant quality, it's constant *quantizer* [21:47:56 CEST] <DHE> which is usually close enough [21:48:24 CEST] <DHE> it basically means the bitrate skyrockets when the picture becomes busy [21:48:56 CEST] <kepstin> crf is nominally constant perceptual quality, with some optimizations to lower the quantizer where the psy optimizations think extra detail isn't needed. [21:49:09 CEST] <furq> __jack__: i'm pretty sure it's only relevant if you can only use cbr [21:50:30 CEST] <abumusa> I want to build a screen recorder using Native Client of Google Chrome, I am having hard time to build the encoder because the compiling process takes lots of time, I am looking for ready code I use [21:50:34 CEST] <kepstin> online streaming is a tricky challenge, since you have to actually limit bitrate to fit the best quality in a particular internet connection bw. [21:51:07 CEST] <abumusa> http://stackoverflow.com/questions/38444827/screen-capture-with-ffmpeg-and-… [21:51:42 CEST] <BtbN> I hope a website won't ever be able to capture my screen. [21:52:03 CEST] <kepstin> BtbN: too late for that, chrome already has builtin screen capture support for webrtc stuff [21:52:10 CEST] <kepstin> iirc it's only exposed to extensions right now [21:52:19 CEST] <BtbN> So not going to use Chrome, nothing to change here. [21:52:24 CEST] <abumusa> exactly, I am building an extension [21:52:43 CEST] <abumusa> I want to make this thing open source, I hate the fact I have to buy a screen recorder each time I want record a screencast [21:52:49 CEST] <kepstin> firefox does as well :) [21:53:29 CEST] <kepstin> abumusa: just use obs [21:53:45 CEST] <abumusa> Compiling the code is taking to much time from me, because it cross compiles to three different platform, the documentation stinks, I've been compiling the code for the past week [21:54:30 CEST] <abumusa> kepstin: I need the software to be a chrome extension it is part of the requirements I can not change or use 3rd party software [21:54:56 CEST] <abumusa> I am building a usertesting.com like website, so I need to push UX tests using chrome extensions [21:55:18 CEST] <abumusa> so I need to use FFMPEG to compile screen capture which Chrome offer to a webm file [21:55:33 CEST] <abumusa> there are four extensions doing this, so I think it is not that hard to program [21:55:50 CEST] <abumusa> but I do not have the experience to build it and could not find a freelancer to do it for me [21:56:15 CEST] <abumusa> if this project works for me I will open source the code of it [21:56:40 CEST] <abumusa> If anyone of you interested in building this with me, I am more than happy to show what I have done so far [21:58:14 CEST] <kepstin> but yeah, libvpx's "constant quality" mode (exposed via the "-crf" option in ffmpeg, even though it's nothing like x264's crf mode) is actually kind of interesting, since it was designed explicitly for online video streaming. [21:58:57 CEST] <kepstin> it's used in combination with a bitrate and vbv settings, so you can' say "encode the video for this bitrate, but use less bits if not needed in order to maintain at least this quality" [22:00:04 CEST] <kepstin> I'm not sure, but I think you might be able to use x264's crf mode along with vbv settings to achieve something similar? [22:11:07 CEST] <abumusa> In case you know some who wants to work on this, please let me know [22:41:55 CEST] <DelphiWorld> hey dudes [22:42:01 CEST] <DelphiWorld> i have a small issue if someone could help? [22:42:25 CEST] <DelphiWorld> ffmpeg -i http://1.1.1.1:4343/1.ts -acodec copy -vcodec copy -f flv rtmp://localhost/live/1.ts [22:42:44 CEST] <DelphiWorld> this work perfectly except when the http source goes down, so the ffmpeg process shutdown [22:42:55 CEST] <DelphiWorld> is there any timeout for the input so ffmpeg keep retrying? [22:46:25 CEST] <kepstin> well, http is a tcp connection, if it doesconnects, it disconnects, and you have to restart from scratch... [22:46:33 CEST] <kepstin> just run the command in a shell script loop? [22:46:53 CEST] <DelphiWorld> kepstin: sad... pretty big issue always need to restart manualy [22:47:02 CEST] <DelphiWorld> if the source was multicast this issue wont happend [22:47:03 CEST] <furq> https://www.ffmpeg.org/ffmpeg-protocols.html#http [22:47:16 CEST] <furq> maybe reconnect_at_eof and/or reconnect_streamed will do what you want [22:47:33 CEST] Action: DelphiWorld reading furq suggestion [22:49:37 CEST] <DelphiWorld> furq: where to put that reconnect_at_eof ? [22:49:50 CEST] <furq> -reconnect_at_eof 1 -i http://... [22:50:08 CEST] <DelphiWorld> thank furq, doing it now [22:52:29 CEST] <DelphiWorld> ok, furq. ffmpeg -reconnect_at_eof 1 http://furq.me/chan.ts [23:13:57 CEST] <DelphiWorld> it's up till now [23:17:57 CEST] <streulma> how can I encode a http mjpeg stream to mp4 format x264 h264 codec ? [23:17:57 CEST] <streulma> or better an rtsp h264 stream to mp4 file [23:18:09 CEST] <streulma> mp4 stream sorry [23:53:24 CEST] <whomp> how can i set the duration of the audio and video to be the same, and to be the lesser of the two's current durations? i'm dealing with video files where one of the two is slightly longer than the other, and it causes playback issues on ios [23:57:38 CEST] <whomp> in general, how am i supposed to sync my videos that play at 29.97 fps with my audio that is at 44100hz? [00:00:00 CEST] --- Tue Jul 19 2016

1 0

ffmpeg-devel.log.20160717
by burek 18 Jul '16

18 Jul '16

[04:48:43 CEST] <omerjerk> Hi, [04:49:03 CEST] <omerjerk> is there a way of reading bits in the reverse order using something in get_bits.h ? [13:02:55 CEST] <firewated> Hi, I've been trying to get HLS subtitles reading working in ffmpeg and although it's working, when the stream is probed for format detection it detects as WEBVTT with a score of 1 because I strip the WEBVTT header that appears at the beginning of every segment. The WEBVTT headers need to be removed because otherwise they will appear throughout the stream [13:02:55 CEST] <firewated> and be treated as subtitles. Any ideas on how to make it more robust? [13:51:23 CEST] <Compn> the probe or removing of webvtt headers ? [13:51:33 CEST] <Compn> either case i dunno, firewated. stick around for answers :) [13:54:58 CEST] <rcombs> u wot m8 [13:55:06 CEST] <rcombs> HLS subtitles are meant to have the WEBVTT header [14:11:33 CEST] <firewated> Yeah but if I leave them in the stream they'll be treated as subs and appear in the output, once for every segment in the HLS playlist, they need to be removed [14:12:51 CEST] <nevcairiel> sounds like thats the bug you should fix instead [14:13:36 CEST] <firewated> Maybe I'm not explaining it well but I don't think that's a bug [14:13:49 CEST] <firewated> I'm going to try and leave the first segment alone and strip the headers from the rest of the segments and see how that works [14:14:04 CEST] <nevcairiel> isnt hls also supposed to support random access [14:14:19 CEST] <nevcairiel> ie. repeat headers in every segment for that purpose [14:14:23 CEST] <JEEB> yes [14:14:35 CEST] <JEEB> same for PPS/SPS etc in video [14:15:16 CEST] <JEEB> so basically the parser should handle it that each subtitle segment has the headers and that thos aren't subtitles [14:15:22 CEST] <rcombs> yeah, if your player is displaying the "WEBVTT" text as subtitles then you have a bug [14:15:25 CEST] <rcombs> and you should fix that [14:15:46 CEST] <rcombs> each segment should have that header [14:18:37 CEST] <firewated> To be clear, ffmpeg doesn't currently read HLS subtitles, it's something I'm trying to add, I'm testing it by using ffmpeg to read a HLS playlist and output a mp4 file with subtitles [14:19:00 CEST] <JEEB> yes, which is why we noted that it's most probably your parser handling the stuff incorrectly :) [14:24:38 CEST] <firewated> Yeah, I'm going to try stripping the header from all the segments except the first, just wanted to get a feel for the IRC and what not, first time here :) [14:25:50 CEST] <rcombs> that is not the correct solution [14:26:05 CEST] <firewated> What do you suggest? [14:33:33 CEST] <rcombs> fix whatever's treating "WEBVTT" as part of the subtitle text to not do that [14:34:04 CEST] <rcombs> sounds like something's just concatenating segments as text files instead of demuxing them as independent files and concatenating their packets [14:34:11 CEST] <rcombs> that's incorrect behavior for HLS subs [15:05:04 CEST] <firewated> That sounds better, I'll look into it, thanks [15:08:02 CEST] <omerjerk> hey how do i calculate crc of a test file to put it inside tests/ref/fate/ [15:08:33 CEST] <JEEB> the test will be outputting the values it got [15:08:48 CEST] <JEEB> and telling you there's nothing there in the file you're comparing against [15:08:57 CEST] <JEEB> that kind of does give you the values, no? ;) [15:09:07 CEST] <nevcairiel> you can use make fate-my-test GEN=1 to generate the files [15:09:16 CEST] <rcombs> oh huh [15:09:17 CEST] <rcombs> TIL [15:09:30 CEST] <omerjerk> I didn't get it. :| [15:09:43 CEST] <omerjerk> I've to add a new file [15:11:06 CEST] <omerjerk> I did this - https://github.com/omerjerk/FFmpeg/commit/e172e333807b4b3b2558a1ffa735ade79… [15:11:44 CEST] <omerjerk> so I also need to add the crc of this new file. [15:11:48 CEST] <omerjerk> I have it locally as well [15:12:10 CEST] <JEEB> shouldn't we use something better than CRC for stuff like that nowadays? [15:12:25 CEST] <JEEB> I think there was at least md5 (frame/full stream) available [15:12:37 CEST] <omerjerk> I have no idea. [15:12:48 CEST] <omerjerk> trac.ffmpeg doesn't mention anything [15:14:38 CEST] <JEEB> https://trac.ffmpeg.org/wiki/framemd5%20Intro%20and%20HowTo [15:15:10 CEST] <JEEB> you can git grep the fate tests for usage examples [15:18:59 CEST] <omerjerk> so I got the framemd5 output. Now it just needs to go inside tests/ref/fate/mpeg4-als-conformance-07 ? [15:19:57 CEST] <nevcairiel> we told you, just run "make fate-whateveryourtestiscalled GEN=1" and it generates the file [15:20:24 CEST] <omerjerk> okayy [15:21:35 CEST] <omerjerk> just to be sure, what should "whateveryourtestiscalled" be in my case? [15:21:50 CEST] <nevcairiel> how would we know, its your test [15:22:48 CEST] <JEEB> omerjerk: what fate outputs when it runs your test ;) [15:22:53 CEST] <JEEB> also how you defined it :) [15:24:47 CEST] <omerjerk> how do I run my test? that's the problem. [15:24:56 CEST] <omerjerk> :/ [15:25:14 CEST] <omerjerk> tried with make fate-mpeg4-als-conformance-07, doesn't work [15:26:05 CEST] <JEEB> there's not too many als tests so I wonder if something a la make fate-als would work? [15:27:29 CEST] <omerjerk> okay. it's doing something. with make-als [15:29:16 CEST] <omerjerk> it did not run for mpeg4-als-conformance-07 [15:29:34 CEST] <omerjerk> let me see what's the problem [15:29:55 CEST] <omerjerk> where do we define which files to download? [15:30:11 CEST] <omerjerk> in make fate-rsync [15:30:13 CEST] <JEEB> omerjerk: probably because you defined it after the foreach? [15:30:31 CEST] <JEEB> looking at https://github.com/omerjerk/FFmpeg/commit/e172e333807b4b3b2558a1ffa735ade79… [15:30:38 CEST] <omerjerk> I don't think that should be the problem. let me just see again. [15:30:42 CEST] <JEEB> oh wait, no [15:30:47 CEST] <JEEB> yeah that probably isn't an issue [15:30:55 CEST] <JEEB> it just loops through the first ALS samples [15:35:46 CEST] <omerjerk> so the problem is that there is no als_07_2ch192k32bF.mp4 inside fate-suit/lossless-audio [15:36:29 CEST] <JEEB> when your test is fully done and you're sure it's actually correct [15:36:39 CEST] <JEEB> you will ask it to be uploaded to the FATE servers [15:36:54 CEST] <JEEB> then after 24h or so the thing can be merged code-wise [15:38:26 CEST] <omerjerk> oh okay. [15:40:20 CEST] <omerjerk> btw I'm at where I started now. [15:40:36 CEST] <omerjerk> how do I generate the crc/md5 for the patch? [15:40:38 CEST] <omerjerk> :| [15:41:34 CEST] <JEEB> you put the file there? [15:41:40 CEST] <JEEB> locally [15:41:43 CEST] <JEEB> and run the test [15:45:33 CEST] <omerjerk> tried that. still doesn't run for 07 [15:46:48 CEST] <omerjerk> ah. found the problem. had an error in the als.mak file [15:46:54 CEST] <omerjerk> thanks for all the help. :) [15:48:47 CEST] <omerjerk> all fine now. got the crc. [15:49:48 CEST] <Chloe> In the HEVC decoder the function name "transform_add" is fairly unclear, it'd make more sense to have it as "add_residual" or similar; what do people think of this? [17:50:16 CEST] <cone-829> ffmpeg 03Jörn Heusipp 07master:f1eb6ddcb34f: libavformat/libopenmpt: Fix memory leak in error path in read_header_openmpt(). [18:07:26 CEST] <bencoh> .53 [18:07:29 CEST] <bencoh> woops [18:12:36 CEST] <cone-829> ffmpeg 03James Almer 07master:d688f39dc4bc: cosmetics: fix some misspelled words [19:14:16 CEST] <FishPencil> So OpenH264 is compatible with the LGPL 2.1? [19:15:11 CEST] <nevcairiel> its BSD licensed, so yes [19:15:18 CEST] <FishPencil> nice [19:15:30 CEST] <nevcairiel> but it supports only a very small subset of h264 streams [19:30:11 CEST] <FishPencil> What a pain to compile [20:15:39 CEST] <cone-829> ffmpeg 03Michael Niedermayer 07master:af13967b6451: MAINTAINERS: Add Aman Gupta for ccaption_dec [20:46:57 CEST] <ubitux> michaelni: re:2b14204 http://sprunge.us/DCQg [20:47:02 CEST] <ubitux> i guess "consistency" but well [20:55:57 CEST] <jamrial> ubitux: any decent compiler will make that optimization. the thing is that there used to be a few that apparently didn't [20:56:21 CEST] <jamrial> at this point it's mostly a cosmetic thing for consistency with the rest of the codebase, i'd say [21:27:19 CEST] <cone-829> ffmpeg 03Aman Gupta 07master:309322ef65ab: avcodec/ccaption_dec: change write_char() to void as return value is unused [21:29:56 CEST] <michaelni> ubitux, i didnt expect it to make a difference with gcc -O3 but i tend to always replace % by & if i can because the auto replace only works for unsigned or proofen to be positive values [21:30:41 CEST] <michaelni> and keeping a mix of & and % depening on gcc being able to auto replace % is more work and easier to get wrong [21:30:45 CEST] <michaelni> than replacing all [23:39:08 CEST] <cone-829> ffmpeg 03James Almer 07master:64d16fd7f5f3: tools/patcheck: add a couple more common typos [00:00:00 CEST] --- Mon Jul 18 2016

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ffmpeg.log.20160717
by burek 18 Jul '16

18 Jul '16

[00:24:21 CEST] <Visual-Vincent> I would like to speak to a moderator about a FFmpeg license question. [00:27:32 CEST] <__jack__> you may ask [00:27:48 CEST] <__jack__> dunno if someone here will be able to help you [00:29:03 CEST] <Visual-Vincent> @__jack__ : Thank you, but I would rather speak to a moderator or someone who has knowledge in FFmpeg's license. [00:29:55 CEST] <__jack__> you won't ask in public, is that it ? [00:30:31 CEST] <__jack__> you can ask michael(a)niedermayer.cc by email, still dunno if he can help you [00:30:59 CEST] <Visual-Vincent> Don't know if I really care about talking in public/private, though I just felt like talking alone with someone. [00:31:02 CEST] <Visual-Vincent> Thank you. [06:28:24 CEST] <soulshock> curious. ffmpeg -i test.mov -filter_complex "yadif=0:-1:0,scale=1280:720,format=yuv420p" -c:v libx264 -profile:v high -level 4.1 -refs 4 -pass 1 -f mp4 -fastfirstpass 1 -c:a libfdk_aac /dev/null produces "[libx264 @ 0x266dd60] profile Main, level 4.1" and when I run -pass 2 it uses profile High [06:47:23 CEST] <blb> soulshock: you said high profile though? [06:48:04 CEST] <soulshock> yeah I used -profile:v high in the 1st pass [06:48:24 CEST] <soulshock> I would expect it to use high profile in both passes [06:53:28 CEST] <soulshock> unless, of course, profile doesn't matter when doing 1st pass [14:01:56 CEST] <AL13N_lappy> i have some aspect ratio trouble [14:02:59 CEST] <AL13N_lappy> there's a smaller device that plays movies, so i tried to convert it to the native resolution and tried to make it use as much screen as possible, but i still got black borders in some places [14:03:42 CEST] <AL13N_lappy> device is resolution: 480x234 but display size is 15.4cm x 8.6cm (7" screen) [14:04:37 CEST] <AL13N_lappy> a movie that has 296x234 resolutions, has no black borders; while a movie that has 339x234 has horizonal black borders above and below [14:05:04 CEST] <AL13N_lappy> i tried to look with ffmpeg -i into those files, but i noted also SAR and DAR mentions [14:05:40 CEST] <AL13N_lappy> how can i convert a movie into the native resolution without having black borders (even if i crop the movie) [14:05:44 CEST] <AL13N_lappy> there's some strange math here [14:07:32 CEST] <BtbN> scale to the appropiate width/height, and then crop the one that's larger to fit the screen. [14:08:37 CEST] <furq> AL13N: https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar [14:32:24 CEST] <AL13N> BtbN: furq: it's just that on resolutions i get, but it seems there's more to resolutions than just pixel widths... i'm missing something [14:32:37 CEST] <AL13N> as to why a 394x234 movie on a device with 480x234 gives black borders above and below(instead of left and right); and a 296x234 movie has no black borders whatsoever [14:33:46 CEST] <AL13N> furq: this sar and dar, are they stored inside the movie metadata? [14:34:04 CEST] <AL13N> is this something that the device uses to display the movie? [14:37:16 CEST] <AL13N_lappy> this movie: 296x234 [SAR 104:73 DAR 1184:657] is shown fullscreen on the device (which is reported to have 480x234 pixels) [14:38:12 CEST] <klaxa> interesting ratios [14:38:30 CEST] <AL13N_lappy> but, i don't understand why it's shown fullscreen... the math doesn't seem to come close [14:38:34 CEST] <AL13N_lappy> so i'm missing something [14:38:48 CEST] <AL13N_lappy> klaxa: yes, indeed [14:40:04 CEST] <AL13N_lappy> at the very least the SAR doesn't match the actual pixels ratio 296x234 [14:42:12 CEST] <AL13N> so, how does a movie actually gets displayed? [14:42:57 CEST] <AL13N> does it take the pixels and applies a SAR on it first? and then depending on the display, it rescales and uses the DAR on that once more? [14:44:21 CEST] <AL13N> or does SAR mean that the movie needs to be interpreted as 104:73 (even though it's resolution is 196x234)? [14:45:26 CEST] <AL13N> or was the original movie just broken with wrong aspect ratios? [15:04:34 CEST] <DHE> SAR (sample aspect ratio) is the aspect ratio of the pixels, not the whole image [15:04:52 CEST] <DHE> so 1:1 on a 1920x1080 encoding will produce 1920x1080 output [15:09:35 CEST] <anddam> -map_metadata at http://ffmpeg.org/ffmpeg.html#Advanced-options has reference to chapters, but I wasn't able to find a description of what "chapters" are in ffmpeg in man page, wiki or documentation [15:10:07 CEST] <anddam> namely i have a MPEG4 container with an AAC audio stream and i'd like to arbitrarily add chapters to use it in itunes as audiobook [15:11:05 CEST] <anddam> also, out of curiosity, why does ffprobe output in stderr rather than stdout? every time I pipe the output with a pager I notice this and I've always been confused by it [15:14:29 CEST] <__jack__> anddam: ffprobe output to stdout [15:14:45 CEST] <__jack__> stderr is the random generic output, stdout it what you asked for [15:15:03 CEST] <__jack__> try ffprobe -show_stream -print_format json truc.mkv 2>/dev/null [15:15:08 CEST] <anddam> "random"? [15:15:29 CEST] <anddam> > ffprobe 01\ The\ Hobbit.m4b ^/dev/null | wc -l [15:15:31 CEST] <anddam> 0 [15:15:49 CEST] <anddam> ^ is fd 2 in fish shell [15:16:03 CEST] <__jack__> yep, you didn't ask for any information, so you get none [15:16:14 CEST] <__jack__> (random as non-scriptable creapy format, human-only) [15:16:26 CEST] <anddam> oh I see, the generic part now, I didn't know I had to query for something [15:16:40 CEST] <anddam> I've always used the default output [15:16:46 CEST] <anddam> s/see, /see / [15:18:34 CEST] <anddam> at least it makes sense [15:18:38 CEST] <anddam> good [15:19:10 CEST] <anddam> __jack__: do you possibly have any hint about the chapters thing as well? [15:43:19 CEST] <Sonmi> hi. im trying to get deinterlaced 60fps A/V capture from a dazzle DVC100. right now i can only get 30p and no audio, anybody know about this? thanks [15:43:42 CEST] <AL13N> DHE: so if i have a 296x234 pixels and it's SAR 1:1 and DAR 1:1 then a display of 480x234 will show 296x234 ? [15:44:19 CEST] <AL13N> DHE: so if i have a 296x234 pixels and it's SAR 2:1 and DAR 1:1 then a display of 480x234 will show 148x234 ? [15:44:32 CEST] <AL13N> or 592x234 ? [15:45:36 CEST] <AL13N> DHE: so if i have a 296x234 pixels and it's SAR 1:1 and DAR 2:1 then a display of 480x234 will show 148x234 ? [15:46:05 CEST] <AL13N> i assume having SAR=2:1 and DAR 2:1 they would cancel each other out then? [15:47:16 CEST] <AL13N> what about the device that plays it... it has 480x234 output pixels, but the display size is 15.4cm x 8.6cm [15:47:32 CEST] <AL13N> would it use the actual cm to scale? or the resolution? [15:59:05 CEST] <__jack__> anddam: ffprobe -show_chapters [16:02:36 CEST] <AL13N_lappy> DHE: furq: BtbN: i'm calculating; but i'm not getting it... :-( [16:03:05 CEST] <AL13N_lappy> the device must've done something odd [16:12:25 CEST] <anddam> __jack__: but that won't help me adding chapters to a MPEG4 [16:13:15 CEST] <__jack__> huh ok, was still on the ffprobe stuff [16:13:26 CEST] <__jack__> I dunno how to create chapter with ffmpeg [16:49:48 CEST] <klaxa> anddam: i used chapters for matroska files, not sure if it works with mp4, but i can dig around my old code [16:50:44 CEST] <klaxa> anddam: https://www.ffmpeg.org/ffmpeg-formats.html#Metadata-1 this section also describes the syntax for chapters [17:02:23 CEST] <anddam> klaxa: thanks [20:20:13 CEST] <jonathan_x> Is there a way to create an easy audio fadeout at the point where I want to cut off a video (maybe video fadeout as well). [22:52:03 CEST] <maziar> why when i use this command i get this error : http://pastebin.com/xADyk776 [22:52:44 CEST] <furq> [libx264 @ 0x23717e0] width not divisible by 2 (1365x720) [22:52:55 CEST] <furq> -vf scale=-2:720 [22:54:05 CEST] <maziar> furq let me test it [22:57:17 CEST] <Karolis> hello, can someone explain me why native frame rate grabber (-re) with concat filter isn't working correct? http://pastebin.com/B9a7umVZ [23:48:27 CEST] <egnun> Hello #ffmpeg [23:49:06 CEST] <__jack__> hi! [23:49:10 CEST] <egnun> I want to compile ffmpeg, but I *only* want to include free codes. (so I *don't* want to compile it with proprietary codecs) [23:49:20 CEST] <egnun> What options do I need to use? [23:49:54 CEST] <egnun> Is this ok? [23:49:54 CEST] <egnun> ./configure --prefix=/opt/ffmpeg --enable-gpl --enable-version3 --disable-nonfree [23:50:20 CEST] <egnun> (Oh, and I downloaded the files directly from the git-repo) [23:51:58 CEST] <__jack__> sound correct [23:52:08 CEST] <JEEB> you're saying it in a weird way and things incompatible with (L)GPL are disabled by default. this does not of course disable the implementation of reverse engineered proprietary formats that are implemented in libavcodec [23:52:38 CEST] <egnun> __jack__: Ok. [23:53:11 CEST] <egnun> JEEB: I am asking, because the "--enable-nonfree" option confused me. [23:53:52 CEST] <egnun> It said under "Standard options:" [23:53:52 CEST] <egnun> "--enable-nonfree allow use of nonfree code, the resulting libs and binaries will be unredistributable [no]" [23:53:53 CEST] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=configure;h=4956afc52cdbfe6e… [23:54:09 CEST] <JEEB> these are the only things, and they specifically require ENABLE-nonfree [23:54:42 CEST] <furq> there isn't an option to disable everything patent-encumbered [23:54:47 CEST] <__jack__> nope [23:54:52 CEST] <__jack__> patent are not worldwide [23:54:57 CEST] <__jack__> not always, at least [23:54:58 CEST] <furq> nonfree just refers to licensing [23:55:15 CEST] <egnun> furq: Ah, ok. [23:55:20 CEST] <JEEB> for example fdk-aac is open source but not license compatible and thus requires nonfree. same for openssl [23:56:09 CEST] <furq> i guess you want --disable-encoders --disable-decoders and then selectively enable the ones you want [23:56:20 CEST] <egnun> furq: So, that's why mp3 and mp4 will also be compiled? [23:56:28 CEST] <egnun> +e.g. [23:56:29 CEST] <furq> right [23:56:33 CEST] <egnun> Aye. [23:56:40 CEST] <egnun> Damnit. I guess. ^^ [23:56:47 CEST] <furq> yeah it's a bit of a nuclear solution [23:58:07 CEST] <egnun> Is there a difference between "--disable-encoders" and "--disable-decoders" patent wise? [23:58:30 CEST] <JEEB> dunno, depends on how things are licensed I guess [23:58:33 CEST] <furq> you're probably less likely to get sued for decoding something [23:58:34 CEST] <__jack__> egnun: I guess that mp3 is free, on a licencing point of view [23:58:44 CEST] <furq> but in theory, no [23:58:54 CEST] <__jack__> (decoder: mp3, probably internal; encoder, libmp3lame, shared object) [23:59:11 CEST] <furq> lame is either gpl or lgpl [23:59:33 CEST] <JEEB> FFmpeg only touches on the open source SW licensing side, nothing else [23:59:49 CEST] <egnun> Ok, thank you, guys! [23:59:53 CEST] <JEEB> also FFmpeg officially provides only source code [00:00:00 CEST] --- Mon Jul 18 2016

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ffmpeg-devel.log.20160716
by burek 17 Jul '16

17 Jul '16

[00:39:42 CEST] <Chloe> michaelni: I briefly asked about the procedure for deprecating demuxers a little while back. For libmodplug, would something like http://pastebin.com/raw/yvA6H8Y0 be adequate? [00:45:32 CEST] <jamrial> Chloe: if libopenmpt replaces libmodplug for every use case then just remove the latter and make sure to mention it in Changelog [00:47:04 CEST] <jamrial> there's no deprecation process for modules and external libraries like this. just send a patch to the ml removing it, like with did for libvo-aac, libaacplus and libdcadec earlier this year [00:52:18 CEST] <Chloe> it doesn't have full feature parity, some of the options (like bass cutoff) just aren't in libopenmpt anymore, and the libopenmpt demuxer doesn't do any video stuff, which the libmodplug demuxer seems to do. (I'm going to see if I can add the video stuff to libopenmpt though) [00:53:46 CEST] <ubitux> you can ignore the video stuff [00:54:04 CEST] <ubitux> it's just an experimental thing, a random toy [00:54:45 CEST] <ubitux> Chloe: if you add a video mode, you should target sth like http://www.cubic.org/player/main.png instead [00:55:13 CEST] <ubitux> the stupid video mode in libmodplug isn't worth porting [00:55:22 CEST] <ubitux> in libmodplug wrapper* [00:56:16 CEST] <Chloe> sure, well I think the other options in libmodplug can be replaced by filters so it's not like the features would be lost entirely [01:30:35 CEST] <michaelni> ff-cubic-player ... :) [01:46:29 CEST] <Chloe> michaelni: right, I'll speak to the libopenmpt guys about getting it in some distros [01:48:44 CEST] <cone-505> ffmpeg 03Jan Sebechlebsky 07master:1732d89403b7: avformat/utils: Add ff_stream_encode_params_copy() [01:48:44 CEST] <cone-505> ffmpeg 03Jan Sebechlebsky 07master:3d7b5ad217e2: avformat/tee: Use ff_stream_encode_params_copy() [01:48:44 CEST] <cone-505> ffmpeg 03Jan Sebechlebsky 07master:e144b86fa4c7: avformat/tee: Support flushing by writing NULL pkt [01:48:44 CEST] <cone-505> ffmpeg 03Jan Sebechlebsky 07master:6a3515a5a49e: avformat/utils: Add ff_format_output_open() function [01:48:44 CEST] <cone-505> ffmpeg 03Jan Sebechlebsky 07master:df9cc7f0a90c: avformat/tee: Use ff_format_output_open() function [04:07:33 CEST] <kode54> ugh [04:07:43 CEST] <kode54> I noticed several things when updating my software for 3.1.1 [04:07:56 CEST] <kode54> I was still using streams[n]->codec [04:08:35 CEST] <kode54> and when updating to use the codecpar member, I neglected to use the function to copy the codec setup from parameters to codec context before opening it [04:09:30 CEST] <kode54> now the only thing I need to do is update to ditch avcodec_decode_audio4 [04:09:36 CEST] <kode54> but I notice that ffplay still uses that [16:17:36 CEST] <cone-803> ffmpeg 03Mark Reid 07master:ed0dc14ebba1: libavcodec/dnxhd: add support more dnxhr header prefixes [18:22:56 CEST] <durandal_1707> poor players [20:36:03 CEST] <cone-085> ffmpeg 03James Almer 07master:298f556af436: configure: add missing loongson dependencies [20:36:04 CEST] <cone-085> ffmpeg 03James Almer 07master:8f0434f85ced: fate: fix apng tests dependencies [21:10:48 CEST] <Chloe> libav changed the name of the transform_add/transform_skip HEVC functions when integrating ffmpeg's asm-related patches, I was wondering if ffmpeg should do that too, or is it insignificant? the transform_add functions are in a file called hevc_res_add.asm which I assume is for 'residual add' (which is what the new name for transform_add functions are in libav) [21:11:53 CEST] <atomnuker> no, it's stupid and it slows down decoding [21:12:16 CEST] <atomnuker> (not that you should care about hevc anyway) [21:13:08 CEST] <nevcairiel> atomnuker: renaming functions slows down decoding? [21:13:28 CEST] <nevcairiel> that must be some crazy magic [21:13:57 CEST] <atomnuker> oh, that's the other patch I saw [21:14:09 CEST] <atomnuker> the one that split the residuals [21:19:54 CEST] <Chloe> which patch is that? [21:52:47 CEST] <Compn> no one replied to my prores question :\ [22:11:30 CEST] <cone-085> ffmpeg 03Michael Niedermayer 07master:2b14204339bd: avcodec/dnxhddata: replace % by & [00:00:00 CEST] --- Sun Jul 17 2016

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ffmpeg.log.20160716
by burek 17 Jul '16

17 Jul '16

[00:06:14 CEST] <maziar> __jack__ you command is 100% wrong its change time ! http://185.13.231.218/player/480.html [00:07:12 CEST] <__jack__> it changes time .. ? [00:07:18 CEST] <__jack__> ok, tchao [00:07:40 CEST] <furq> what a nice young man [00:15:50 CEST] <thebombzen> something I've noticed is that over the last few years, there's been a large push toward programming for people who wouldn't otherwise program [00:16:00 CEST] <thebombzen> and not just programming, also scrpting, CLI usage, etc. [00:16:26 CEST] <furq> hey guys can you help me with my computercraft script [00:16:39 CEST] <thebombzen> the upside is that CLI/etc. has gotten more support. the downside is that the general quality of questions one might find on the internet has definitely devolved [00:16:47 CEST] <thebombzen> lol [00:17:25 CEST] <thebombzen> I've defintiely noticed a difference in the quality of StackOverflow questions as an example, /especially/ in my native lang (which is Java) [00:17:56 CEST] <furq> i'm pretty confident that lua has suffered the worst [00:18:07 CEST] <thebombzen> lua? is that the computercraft language? [00:18:21 CEST] <thebombzen> Java's been pretty bad b/c of Minecraft tho [00:18:22 CEST] <furq> it's the good language which is also used for unfortunate things like computercraft [00:18:30 CEST] <thebombzen> although part of that is also that younger people are doing Java and other computer things more. [00:18:58 CEST] <thebombzen> I mean seven years ago I learned Java but I was only 13 and I asked really bad questions as a 13yo new Java programmer [00:19:16 CEST] <furq> i mean it's probably a noble thing to try to get teenagers writing code [00:19:32 CEST] <furq> just as long as you don't also encourage them to ask "how do i write linux in lua" on irc [00:19:47 CEST] <thebombzen> now that I"m 20 and have 7 years of Java experience I know how to ask questions better, and it's easy to get arroagant and say that everyoen should knowbetter [00:19:50 CEST] <thebombzen> oops gtg [00:20:21 CEST] <furq> it is very easy to be arrogant [00:20:23 CEST] <furq> that's the beauty of it [02:50:42 CEST] <johnnny22> I have a TS file, when probed looks like this: http://pastebin.com/H11DKUPs It has a Program 1 and a "No Program" . Re-encoding it provides the 2nd output below on that link, where all the information is merged into Program 1. Now, this TS original file is part of a HLS manifest and when I try to encode that manifest with ffmpeg, I get this output: http://pastebin.com/5yWFJ4QQ But if I [02:50:42 CEST] <johnnny22> swap the 1st segment with the re-encoded segment, all goes fine and the whole manifest seems to get encoded. Is there a command line argument that can be used to ask ffmpeg to use the "No Program" as it's video&audio source ? [02:51:53 CEST] <johnnny22> or some other way of fixing this :o [03:40:24 CEST] <DHE> is there a trick to get ffmpeg running in the background with `&` on linux? -nostdin -loglevel error doesn't do it [03:48:23 CEST] <drv> ffmpeg ... < /dev/null & [03:50:13 CEST] <drv> although -nostdin looks like it's supposed to do the same thing... not sure why that doesn't work [03:58:48 CEST] <DHE> nohup ended up working as well [04:03:48 CEST] <thebombzen_> wooh freenode blocks [04:03:56 CEST] <thebombzen_> freenode blocks SASL connections from my phone* [04:04:13 CEST] <thebombzen_> but I got it worked by tunneling it via SSH to my home comp [04:07:53 CEST] <thebombzen_> DHE: in general you can use "nohup <command> &" to run something in the background [04:08:13 CEST] <thebombzen_> although you could also use daemon(1) [04:08:27 CEST] <DHE> no, I need to collect it when it finishes [04:08:44 CEST] <DHE> ffmpeg is actually just a step in a shell script and I am backgrounding& the whole script [04:08:58 CEST] <DHE> and no I can't daemonize the script either [04:09:14 CEST] <thebombzen_> collect it? [04:09:28 CEST] <DHE> wait() [04:09:47 CEST] <thebombzen_> forgive my ignorance dunno what wait() does [04:10:04 CEST] <DHE> waits for a process to exit, and properly cleans it up while returning the exit code [04:10:18 CEST] <DHE> otherwise you get zombies [04:10:30 CEST] <thebombzen_> is it a shell builtin? cause that seems really useful [04:10:41 CEST] <DHE> when professors talk about unix processes in class, they close the door [04:10:56 CEST] <DHE> yes, it is a shell builtin. makes the shell wait for all background jobs to finish [04:11:01 CEST] <DHE> but I'm using it in the context of the system call [04:11:06 CEST] <thebombzen_> I"ve been doing the ever-awkward while <running>; sleep 0.5; done [04:11:16 CEST] <thebombzen_> errr. do sleep 0.5. but yea [04:11:33 CEST] <thebombzen_> >when professors talk about unix processes in class, they close the door [04:11:36 CEST] <thebombzen_> I don't get that reference [04:11:58 CEST] <DHE> "see, it's necessary for a parent process to reap their dead children to prevent the system from filling up with zombies" [04:12:08 CEST] <DHE> imagine hearing that coming from a classroom full of nerds [04:16:26 CEST] <thebombzen_> haha [04:16:55 CEST] <thebombzen_> when I talk about linux everyone tunes me out cause I typically use terms that don't have everyday meanings [04:17:41 CEST] <thebombzen_> I mentioned to someone the other day that I had to install the resume hook, create a swap partition, regenerate the initramfs and reconfigure the bootloader adding the resume option to the kernel line [04:18:00 CEST] <thebombzen_> in order to get my computer to hibernate. and he understood basically zero of it. [07:02:37 CEST] <Bray90820> This is the command I'm using to convert my videos files what I would like to do is chose audio track 2 instead of audio track 1 [07:02:37 CEST] <Bray90820> FFmpeg -i "video.mkv -c:a "video.mp4" [07:04:47 CEST] <Bray90820> Sorry this is my code [07:04:52 CEST] <Bray90820> FFmpeg -i "video.mkv -c:a copy "video.mp4" [07:39:25 CEST] <kepstin> Bray90820: you can use the '-map' option to select specific tracks to use. In this case you might want to try '-map 0:v -map 0:a:1' to select the video track and second audio track from the source file. [07:40:02 CEST] <kepstin> (note that if you use any -map options, you're required to specify all the streams you want, it disables the automatic selection, which picks the first video and first audio stream) [07:49:28 CEST] <Bray90820> kepstin: I have tried all sorts of different stuff with the map option that didn't work do you have any specific code for me to try? [07:49:38 CEST] <Bray90820> I wanna use that code and select audio track 2 [07:49:52 CEST] <kepstin> I gave you specific code to try in my first reply. [07:51:11 CEST] <thebombzen> Bray90820: he just said -map 0:v -map 0:a:1 [07:51:19 CEST] <thebombzen> come on people is this today's theme or something [07:52:01 CEST] <Bray90820> So is it like this [07:52:02 CEST] <Bray90820> ffmpeg -i "video.mkv" -map 0:a:1 "video.mp4" [07:52:17 CEST] <kepstin> that is not what I said to do [07:52:25 CEST] <thebombzen> he said what options to use [07:52:31 CEST] <thebombzen> and I literally copied and pasted them [07:52:34 CEST] <kepstin> and if you read my second message, you'll understand why that won't do what you want [07:53:10 CEST] <Bray90820> I am pretty confused [07:53:23 CEST] <Bray90820> I just don't know what to type into the terminal to get the video with audio track 2 [07:53:31 CEST] <thebombzen> >[01:39:25] <kepstin> Bray90820: you can use the '-map' option to select specific tracks to use. In this case you might want to try '-map 0:v -map 0:a:1' to select the video track and second audio track from the source file. [07:53:39 CEST] <kepstin> then take a deep breath, scroll up, and read my messages again slowly with a clear mind. [07:55:00 CEST] <Bray90820> I saw that but in your code there is no video file inputted [07:55:21 CEST] <thebombzen> Bray90820: if you're going to use a command line you need to be able to know what "options are" [07:55:23 CEST] <kepstin> I gave you, in quotes, some options to add to your existing command. [07:55:37 CEST] <thebombzen> those are called "options" [07:55:50 CEST] <thebombzen> or "arguments" [07:55:55 CEST] <Bray90820> I know what an option is but where do I input the video file [07:56:14 CEST] <kepstin> the map options go after all the input files and before the output file. [07:56:32 CEST] <thebombzen> -map is an output options, which means that they go after the input filename and before the output filename [07:58:10 CEST] <Bray90820> thebombzen: Thank you that worked [07:58:13 CEST] <kepstin> in this case "-map 0:v" means "grab the video track(s) from the first input file", and "-map 0:a:1" means "grab only the second audio track (0-indexed) from the first input file". You need to use both if you want video and audio. [07:58:33 CEST] <thebombzen> well I'm glad I helped but you should thank kepstin cause he actually gave you the answer [07:58:39 CEST] <thebombzen> and I just said it again [07:59:44 CEST] <Bray90820> kepstin: Thank you that worked [07:59:45 CEST] <thebombzen> I should go to bed lol it's 2am [07:59:50 CEST] <Bray90820> Alright [07:59:52 CEST] <Bray90820> Goodnight [10:29:49 CEST] <symtab> hi [10:30:03 CEST] <symtab> i'm using the following command http://pastebin.com/9DGszJzT to write a sliding text on a video [10:30:27 CEST] <symtab> can anyone explain me how i can control after how many seconds the text will appear and how i can increase the speed of the scrolling text [10:30:28 CEST] <symtab> ? [13:09:28 CEST] <ObsidianX> hey folks, i'm trying to publish a stream to crtmpserver via ffmpeg from an h264 named pipe (mkfifo) and it looks as though ffmpeg is transmitting the data but it only shows up on the stream once i CTRL+C ffmpeg [13:10:24 CEST] <ObsidianX> i'm running on a raspberry pi and am streaming the camera module using this command: http://pastebin.com/4uJXPLZ6 [13:10:58 CEST] <ObsidianX> is there something i'm doing wrong from ffmpeg's perspective or is something wrong with crtmpserver, perhaps? [13:11:37 CEST] <furq> fwiw ffmpeg supports encoding with omx now [13:11:43 CEST] <furq> so you might be able to cut raspivid out of the loop [13:12:04 CEST] <furq> i also never had any luck with crtmpserver, nginx-rtmp seems like a much better choice [13:15:27 CEST] <ObsidianX> cool, i'll look into that :) thanks [16:03:01 CEST] <soulshock> is it possible to run psnr during/after encoding in the same cmdline? [16:03:20 CEST] <soulshock> run=calculate [17:11:27 CEST] <kepstin> soulshock: not in general, but iirc the x264 encoder actually prints out PSNR stats after encoding as part of its log output [17:12:15 CEST] <JEEB> per-encoder stats are not comparable as the implementation can be different [17:12:25 CEST] <kepstin> you could probably do it at the same time as encoding by playing around with pipes and having a second ffmpeg command that decodes the video and runs the psnr filter. [17:12:36 CEST] <JEEB> you encode the results and then use a single way to calculate the values comparing original vs a single encode [22:02:09 CEST] <t4nk183> Na ihr kotnascher [22:05:56 CEST] <t4nk183> Hi iGeni [22:06:47 CEST] <t4nk183> I have a question iGeni [22:08:15 CEST] <furq> what a nice young man [23:11:34 CEST] <zotherstupidguy> i am having two mp3 files of the same duration, i want to mix them into one file as the end result also be of the same duration, how to acheive that? [23:16:15 CEST] <c_14> zotherstupidguy: https://trac.ffmpeg.org/wiki/AudioChannelManipulation [23:32:01 CEST] <zotherstupidguy> c_14 thanks, avconv is to replace ffmpeg at the commands, correct? [00:00:00 CEST] --- Sun Jul 17 2016

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