[Ffmpeg-devel-irc] ffmpeg.log.20160511

[burek]

[00:30:03 CEST] <vade> im transcoding a video , and I have the same number of packets and frames from source and destination, but my resulting video stream (the only stream) ha a duration thats slightly off. my source is 5 seconds on the dot, my resulting stream is 4.96. Both have 120 frames. original is 24 fps, my resulting is 24.20 which is the difference in duration. is there a trick to marking my output context duration / timebase to force absolute frame rate? sh
[00:30:04 CEST] <vade> I NOT be setting certain codec contex flags on encode, like average frame rate, or stuff like that?
[01:03:11 CEST] <esdwdftty> Big or not big advantage to decoding video (video players) between 64 bit and 32 bit OS? In encoding the difference I have on my CPU is ~ 40% faster (64 Bit). When decoding the difference I do not see significant during playback of a video file the CPU load is about the same average 60% of CPU 720p 30 frame. The encoding I do rarely.
[01:04:26 CEST] <vade> youre probably waiting to decode most of the time for realtime playback I imagine. You can likeply playback orders of magnitude faster than realtime.
[01:04:58 CEST] <vade> i bet a profile is just sleeping / spinning waiting for an appropriate time to get the next frame, thus similar CPU. Just a guess.
[01:13:03 CEST] <yongyung> What is the difference between using (libx264) -b:v 1000k -bufsize 1000k and -maxrate 1000k -bufsize 1000k?
[01:15:18 CEST] <esdwdftty> The translation software. Me here that interests it makes sense to change OS Windows 32 bit on Linux 64 bit only for video decoding it CPU. I have Windows 32 bit can install Linux 64 BIt. Please write without fanaticism for Linux or dislike for Windows. Linux OS difficult for me if something it is to do yourself.
[01:19:20 CEST] <neuro_sys> I'm tinkering with libav*. After av_read_frame(), if I call avcodec_decode_video2 for video stream, av_read_frame never finishes, and main loop never breaks. Do we have to always use filters (as the example I'm following does)?
[01:22:44 CEST] <esdwdftty> I did not see the benefits when watching video files (decoding on CPU) between 64 bit Linux and 32 bit Windows.
[01:23:02 CEST] <esdwdftty> dxva i see
[01:23:07 CEST] <neuro_sys> https://gist.github.com/neuro-sys/4a3897e115b4969b35277c44c85fd8d6
[01:25:54 CEST] <esdwdftty> Higher frequency of the processor too, the difference is visible.
[01:31:56 CEST] <esdwdftty> Humor: maybe can need compare Windows 2000 16 bit  with OS 64 bit then the difference will show itself.
[03:17:19 CEST] <cobadger> baptiste: pm?
[03:59:47 CEST] <baptiste> cobadger, shoot
[04:02:09 CEST] <cobadger> baptiste: opened pm to you
[04:53:11 CEST] <h64> hello. i'm having a weird problem with -map_metadata 0, the creation_date is off by a few hours.
[04:53:18 CEST] <h64> nearly a day
[04:53:45 CEST] <h64> in other words, it's not directly copying the creation date
[04:54:32 CEST] <h64> maybe it's losing timezone information? input is UTV
[04:54:34 CEST] <h64> *UTC
[04:55:43 CEST] <h64> result is ahead by exactly 6 hours
[04:56:36 CEST] <h64> (sorry i thought it was nearly a day because the days werent the same and i am bad at math lol)
[05:05:34 CEST] <h64> ok i got it fixed. i set environment variable to TZ=UTC and no conversion was done.
[05:06:10 CEST] <h64> perhaps theres a timezone conversion bug in ffmpeg?
[07:02:26 CEST] <yongyung> Are you guys aware of any proper ways to upload videos to YT? With that I mean resumable uploads, bandwidth caps, error detection for individual chunks (I know tcp does that but it's not that great as I had to find out when I had to re-upload a video I had uploaded over night).
[07:30:31 CEST] <furq> yongyung: https://github.com/youtube/api-samples/blob/master/python/upload_video.py appears to support resumption
[07:30:38 CEST] <furq> no idea about the other two though
[07:32:56 CEST] Action: davidshen84 
[07:32:59 CEST] <davidshen84> hello
[07:33:29 CEST] <davidshen84> I am new to ffmpeg library. I have some experience using the ffmpeg command line tool
[07:33:38 CEST] <davidshen84> anybody?
[07:34:04 CEST] <davidshen84> I want to create a simple python tool to read the video stream information
[07:35:42 CEST] <davidshen84> e.g. Stream #0:1(eng): Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1280x720, 5942 kb/s, 29.97 fps, 29.97 tbr, 1k tbn
[07:35:50 CEST] <davidshen84> any idea how to do that?
[09:40:32 CEST] <Anaphaxeton> goodmorning
[09:41:00 CEST] <Anaphaxeton> there is a video that  i want to transcode to embed subtitles to
[09:41:28 CEST] <Anaphaxeton> it is a hight quality h264 etc etc video
[09:41:59 CEST] <Anaphaxeton> i don't want to downgrade the quality and i am willling to make a sacrifice on size
[09:43:46 CEST] <Anaphaxeton> i thought about encoding it into mpeg2, because it is very simple compared to h264 and i dont want a psychovisual encoding over psychovisually encoded video
[09:43:53 CEST] <Anaphaxeton> what is your opinion?
[09:45:04 CEST] <bmariesan> Hi everyone, I'm experiencing some issues with ffmpeg udp streaming to a remote device. I have either two computers in different networks or one computer and one mobile device. Since they are in different networks I'm utilising a NAT traversal technique called UDP hole punching, meaning that I'm setting an initial connection to a HTTP server from both devices and the server finds out the public IPs and the ports opened in the NAT for 
[09:45:13 CEST] <furq> i don't think psy will really make much of a difference but you could just use x264 and turn psy off if you're really concerned
[09:45:28 CEST] <bmariesan> You can find my ffmpeg command and ffplay commands at http://pastebin.com/QbTcgswX
[09:48:26 CEST] <Anaphaxeton> furq, in principle, what should one do for encoding in a case like mine?
[09:50:18 CEST] <furq> there isn't any magic way to not lose any visual quality short of encoding lossless
[09:50:28 CEST] <furq> personally i'd just use x264 with a low crf
[09:50:51 CEST] <Anaphaxeton> ok! much appreciated :)
[10:02:47 CEST] <Anaphaxeton> i like the subtitles vlc puts out a lot. how could i achieve a similar quality when embedding subs to the video?
[10:29:03 CEST] <adamk__> hey, does anyone know what might cause this error? Invalid option or argument: 'ref=6', parsed as 'ref' = '6'
[10:29:25 CEST] <adamk__> i thought ref was an option
[10:30:16 CEST] <furq> -refs
[10:32:14 CEST] <adamk__> furq: thanks, this seems to work. i was looking at another file which was encoded and it had that in its encoding settings
[10:34:37 CEST] <furq> -ref is the x264 option
[12:22:00 CEST] <neuro_sys> reading from a sequence of pngs versus a h264 video to encode a video in terms of speeds didn't make any difference. I'd expect a difference at the least. Both took 50 seconds.
[12:24:12 CEST] <neuro_sys> oh reading from BMPs now, seems faster
[12:24:49 CEST] <neuro_sys> only slightly
[12:26:16 CEST] <mr_lou> So I've noticed, that when I create videoclips in my editor, I'm always doing the exact same thing: Simply putting each clip right after each other with a simple crossfade effect. There has to be an easier way to do this without using a video-editor? An ffmpeg command to simply take a bunch of mp4 files and mix them together, crossfading the first/last second of each?
[12:27:22 CEST] <neuro_sys> yeah, sounds like it can easily be automated
[12:37:32 CEST] <mr_lou> Yea I think it must. Googling isn't bringing me any examples though.
[12:40:04 CEST] <mr_lou> I can find examples that require me to know the durations of each video.
[12:40:10 CEST] <mr_lou> That's not good enough. :-)
[14:19:40 CEST] <neuro_sys> I need some good reading on PTS to understand it
[14:19:56 CEST] <DHE> neuro_sys: PNGs are compressed with zlib, so while timing being identical is a bit surprising, being slow is expected
[14:26:47 CEST] <neuro_sys> fair enough
[14:29:07 CEST] <n1cksn1ck> hi. i am trying to rotate a video file from landscape to Portrait? But cant seem to find the correct Adv options filter command. Can anyone help me rotate/re-orientate my video file or know the correct Adv options command? Any help will much apprecieate : )
[14:34:44 CEST] <furq> n1cksn1ck: https://ffmpeg.org/ffmpeg-filters.html#rotate
[14:35:03 CEST] <furq> or you can do it without reencoding using mp4 metadata (if it's an mp4)
[14:36:27 CEST] <furq> -c copy -metadata:s:v:0 rotate=90
[14:36:38 CEST] <furq> that depends on player support though
[14:40:45 CEST] <n1cksn1ck> hi furq. thanks for your suggestion. I have been to that page. I didnt try the rotate command. It seemed complicated. I did try the transpose command but got an error.
[14:42:58 CEST] <n1cksn1ck> i will try the command as you have suggested. And let you know how it went.
[14:47:16 CEST] <n1cksn1ck> furq: Wow. It accepted the command. So... so far so good. When the process is finished I will let you know if ive been successful(in rotating the ouput). Thanks heaps furq  : )
[14:48:21 CEST] <furq> that should be pretty much instant
[14:50:45 CEST] <n1cksn1ck> not sure what you mean there furq. Its a 400mb file. I do know that it is approx 6000 frames. And it is only at 1300. I dont have new/modern pc. It's about 10 yrs pld.
[14:51:48 CEST] <furq> if you're just changing the metadata then it shouldn't be encoding anything
[14:54:49 CEST] <n1cksn1ck> furq: um. im not just doing a rotation. im also hoping to resize it. Its a video my aunty took on her ph. But she took it in landscape. She asked if I could turn it the right way up. Some how, she managed to take a video 1920x1080.
[14:55:17 CEST] <furq> it would be faster to just rotate it
[14:56:01 CEST] <n1cksn1ck> oh ok. I really dont know  much about video and audio formats etc.
[14:56:18 CEST] <furq> obviously if you were using transpose you'd be reencoding, in which case it'd be faster to resize
[14:56:24 CEST] <furq> but this is just changing some metadata in the container
[14:57:14 CEST] <furq> it doesn't actually rotate the video, it just instructs the player to do so
[15:01:20 CEST] <n1cksn1ck> furq: please forgive my silence/slow response to you. Because i really dont know much about all this video/audio codec/format etc. Im finding it hard how to word my questions or responses to your comment. do you know what i need to type to transpose it 90deg?
[15:01:34 CEST] <adamk___> i found a video which i thought looked really good, is it possible to find which settings and filters the encoder used?
[15:02:01 CEST] <furq> adamk___: mediainfo may show you depending on which encoder was used
[15:02:29 CEST] <n1cksn1ck> furq:  Im up to 3800 frames. just over 2000 frame to go : )
[15:02:50 CEST] <furq> n1cksn1ck: if you're using transpose it should work fine
[15:03:23 CEST] <furq> -metadata:s:v:0 rotate=90 may or may not work but it should be more or less instant
[15:04:00 CEST] <adamk___> furq: it was encoded in x264, i see there are encoding settings. is there a way i can use the same ones?
[15:04:09 CEST] <n1cksn1ck> im not using it(yet). I dont know how : ( can you please give me an example like you did with rotate?
[15:04:28 CEST] <furq> n1cksn1ck: pastebin the command you're using
[15:04:33 CEST] <furq> adamk___: copy and paste
[15:04:58 CEST] <furq> or pastebin them and i'll tell you if it's using one of the standard presets, which it probably is
[15:06:34 CEST] <adamk___> furq: http://pastebin.com/VuUiC6mg
[15:07:41 CEST] <furq> well it's not a standard preset
[15:08:15 CEST] <furq> but it's also using 2-pass which means the guy who encoded it probably isn't to be trusted
[15:08:31 CEST] <furq> either that or he distributed this movie in a bunch of 100MB rar files
[15:09:05 CEST] <n1cksn1ck> furq: I dont know pastebin. but when i tried to use the transpose command. I just added transpose=1.
[15:09:12 CEST] <adamk___> why is he not to be trusted?
[15:09:32 CEST] <furq> 2-pass is pretty much useless unless you're targeting a specific filesize, which is basically worthless because it's 2016
[15:09:56 CEST] <furq> the only reason to do that is if you don't want your perfectly legal release to be nuked off topsites
[15:10:23 CEST] <furq> fwiw i just encode everything with -preset slow -crf 20
[15:10:23 CEST] <adamk___> i see
[15:10:41 CEST] <furq> there's not much to be gained from tweaking settings like that
[15:10:59 CEST] <furq> everything will look great with a low enough crf
[15:12:13 CEST] <furq> n1cksn1ck: -vf transpose=1
[15:12:24 CEST] <furq> that should rotate 90 degrees clockwise
[15:13:33 CEST] <adamk___> furq: i have the bluray source of the same file http://imgur.com/a/3OIuY
[15:14:04 CEST] <adamk___> i have no idea how he did this
[15:14:27 CEST] <furq> is the source the bottom one
[15:14:34 CEST] <adamk___> yes
[15:14:39 CEST] <n1cksn1ck> furq: i think i did try that also(found it at a puppylinux forum) got an error. so tried what i could from ffmpeg Documentation. Also got an error. maybe i typed something wrong. This file is nearly finished : )
[15:14:57 CEST] <furq> i don't think that has anything to do with x264 settings then
[15:15:28 CEST] <adamk___> then what did he do
[15:15:35 CEST] <furq> probably filters
[15:15:43 CEST] <furq> certainly for the colour balance
[15:15:56 CEST] <furq> or maybe he just has a better source
[15:15:57 CEST] <adamk___> and i can't see which ones he used?
[15:16:09 CEST] <furq> not without asking him
[15:16:22 CEST] <n1cksn1ck> furq: the ouput video wasnt rotated : (
[15:16:22 CEST] <adamk___> i see
[15:16:42 CEST] <adamk___> but i can apply filters through ffmpeg, correct?
[15:17:10 CEST] <n1cksn1ck> furq: i will try transpose(how you've suggested).
[15:18:03 CEST] <furq> adamk___: https://ffmpeg.org/ffmpeg-filters.html
[15:20:12 CEST] <adamk___> furq: thanks, i will try to dabble a little bit in that
[15:25:46 CEST] <n1cksn1ck> furq: tried -vf transpose=1 and got an unrecognized option 'vf' in the generated log file. Any suggestions?
[15:26:21 CEST] <furq> pastebin the command
[15:27:38 CEST] <n1cksn1ck> furq: i dont know how.
[15:29:02 CEST] <furq> http://pastebin.com/
[15:31:13 CEST] <n1cksn1ck> furq: do you mean all the stuff in the Adv options Box?
[15:31:47 CEST] <n1cksn1ck> and does it matter if im using ffconvert v1.2?
[15:38:06 CEST] <n1cksn1ck> hey furq: im need a lil break. thanks heaps for your time and suggestions. I'm going to make some food. Grab a cuppa ; ) If i dont talk with you again. I hope you have a great day.
[15:38:13 CEST] <ibrasiskis> Hello does anyone knows if the hap codec in ffmpeg is hardware accelerated?
[15:42:22 CEST] <ibrasiskis> That is uses opengl to decompress frame textures?
[15:47:02 CEST] <jkqxz> ibrasiskis:  No, it's purely software.
[16:03:18 CEST] <ibrasiskis> It sucks a little, because that is the point of the codec. It would be adventure to port the codec to linux myself, but probably that is over my capabilities as OpenGL/multimedia programmer. :)
[16:07:53 CEST] <IanWizard> ibrasiskis: what codec?
[16:11:26 CEST] <ibrasiskis> Hap codec, I asked if it uses gpu to decompress frames on ffmpeg, but I was told that it is purely software based.
[16:17:44 CEST] <ibrasiskis> Hap is Intraframe (keyframe only) codec specialiazed for visualisations and projections, encodes alpha channel and is normally used to compose scenes from multiple scenes.
[17:01:19 CEST] <yelmond> hello - is there a way to convert image files (say, .nef raws) on-the-fly using another program as ffmpeg processes them into a video? Trying to avoid having to convert ALL the images first and then running ffmpeg.
[17:03:26 CEST] <c_14> Assuming the program to convert the images can do so in sequence and output them on stdout, yes
[17:03:32 CEST] <c_14> use ffmpeg's image2pipe demuxer
[17:05:37 CEST] <yelmond> thanks c_14! I'll look into image2pipe.
[17:22:52 CEST] <mundus2018> Can someone help me convert a bunch of .m2ts files to a lower bitrate/quality
[17:23:17 CEST] <mundus2018> right now like an hour and a half video is about 6GB
[17:23:36 CEST] <mundus2018> Im shooting for 700mb-1gb
[17:27:14 CEST] <n1cksn1ck> thanks again furq:
[17:30:47 CEST] <pgorley> Is ffmpeg_videotoolbox.c an example on how to use vt or code I need to include to use it?
[17:32:59 CEST] <c_14> mundus2018: https://trac.ffmpeg.org/wiki/Encode/H.264
[17:33:22 CEST] <c_14> pgorley: pretty sure it's an example
[17:33:29 CEST] <jkqxz> Mainly an example.  The ffmpeg_*.c hwaccel files are implementation-specific bits needed to use that hwaccel in the ffmpeg utility (combined with the other bits in ffmpeg.c).
[17:33:52 CEST] <pgorley> Alright, thanks!
[18:06:50 CEST] <Sashmo> can anyone tell me whats the best way to overlay a animated logo on top of a video? or better yet an swf or flv with alpha on top of the video, ideally looking at adding dynamic data on top of my live stream
[18:08:27 CEST] <Angus> Trying to create a custom video decoding plugin here. What does hwaccel_device do?
[18:30:00 CEST] <neuro_sys> Is there a magick trick to speed up ffmpeg (increasing threads perhaps)?
[18:30:22 CEST] <neuro_sys> I have 5 input streams one complex filter with 5 to 10 chains and one output stream
[18:43:22 CEST] <BtbN> Add more cores, reduce quality.
[18:43:22 CEST] <nMaib0> how can I convert an audio file from 24 fps to 23.976?
[18:43:32 CEST] <nMaib0> dts to opus
[18:43:42 CEST] <BtbN> Audio does not have video-fps.
[18:44:45 CEST] <nMaib0> yeah tell that to the video I just converted from 24 to 23.976 and now the audio is out of sync, I need to speed it up to match the resultant video
[18:45:10 CEST] <nMaib0> I could fix it by telling matroska to play the video at 24 but that's a crappy fix
[18:45:16 CEST] <BtbN> so you didn't convert the video, you just speeded it up.
[18:46:34 CEST] <nMaib0> no I converted it to 23.976
[18:47:12 CEST] <nMaib0> maybe I converted it wrong...
[18:47:53 CEST] <Mavrik> If your audio is out of sync then you just sped up the video without actually doing the pulldown :/
[18:47:54 CEST] <nMaib0> I read that most BD 24fps files are in reality 23.976 so I used assumefps on avisynth, like some user advicced on doom9
[18:48:39 CEST] <Mavrik> Well it seems it wasn't :P
[18:48:43 CEST] <nMaib0> yeah, but I find that to be a crappy fix, matroska can do that easily and I already tested and it works.
[18:48:52 CEST] <nMaib0> yeah haha
[18:49:26 CEST] <Mavrik> So either you use atempo to speed up audio a bit
[18:49:43 CEST] <Mavrik> Or do the conversion with actually changing fps
[18:50:38 CEST] <nMaib0> yeah I think I'll leave the 24fps
[18:50:46 CEST] <nMaib0> and reconvert again
[18:53:24 CEST] <casf> i'm streaming AVFrames through an AVFilterGraph and would to change filter parameters mid-stream.  is there a recommended way to do this or an example somewhere?
[18:56:53 CEST] <durandal_1707> casf: only by recreating graph
[18:57:48 CEST] <casf> hrmmm
[18:58:08 CEST] <casf> i'm worried about gaps, but i haven't even tried it yet
[18:58:17 CEST] <casf> or like...dropped frames from the buffer sink
[18:58:42 CEST] <casf> i guess i could just reuse the buffer sink?  freeing the filter graph just destroys the links and leaves the filter context objects unmodified.
[19:02:46 CEST] <casf> durandal_1707: have you implemented anything similar that rebuilds a filter graph mid-stream?
[19:07:01 CEST] <BtbN> some filter support runtime reconfig
[19:07:23 CEST] <casf> BtbN: are there any docs for that?
[19:07:31 CEST] <BtbN> I'd guess so.
[19:11:28 CEST] <vade> hello. Im trying to flush my resample context via swr_convert_frame . the doc header state NULL input is how to do this, but I get a crash
[19:11:31 CEST] <vade> is this expected?
[19:12:04 CEST] <vade> btw Mavrik fixed my issue the other day - was not setting duration on packets to the muxer so last frame had 0 duration, thats why frame counts matched byt duration / fps didnt.
[19:21:12 CEST] <durandal_1707> casf: I had
[19:21:30 CEST] <casf> durandal_1707: found this https://github.com/PromyLOPh/pianobar/blob/master/src/player.c -- good example of dynamic volume
[19:21:45 CEST] <casf> that coupled with BtbN's recommendation that _some_ filters support runtime reconfig is super helpful
[19:21:48 CEST] <casf> thanks to you both. :D
[19:22:45 CEST] <casf> i'm guessing i'll eventually need to deal with the graph reconstruction, but for now this is a good starting point for me - just getting my feet wet with ffmpeg libs anyway
[19:23:28 CEST] <casf> and sorry, lines 81-133 have the example since that is a pretty big file
[19:26:20 CEST] <vade> Yea, im not sure how im supposed to vend samples from my resample context via swr_convert_frame - i cant provide a NULL input without crashing, and if I make an empty frame with appropriate format, channel and sample rate, i keep spewing out empty packets
[19:26:29 CEST] <vade> rather, empty frames.
[19:29:35 CEST] <tobor> So I am sitting here at work on a windows_10 computer and know nada bout this os. See there is a static build for ffmpeg on windows. Need to import video from a DV video camera to a via firewire. Want to drop this directly to our 18TB ZFS NAS. Can this be done using a windows_10 os?
[19:32:49 CEST] <tobor> my output ding ffmpeg -f dshow -list_devices true -i dummy
[19:32:52 CEST] <tobor> http://paste.debian.net/683428/
[19:37:39 CEST] <tobor> wow gonna be nice in a few weeks when GNU/Linux will be incorporated into windows_10. The idea of doing something like compiling with gcc in a windows shell or something like  - watch -n1 'grep Mhz /proc/cpuinfo |sort -k4rn' in windows seems awesome.
[19:39:09 CEST] <tobor> Not sure how that is working out with the GPL though. Seems Can Canonical worked something out with MS.
[19:40:28 CEST] <tobor> lol can just apt-get directly in a windows bash shell and use Canonical's (ubuntu's) repos.
[19:41:27 CEST] <kiroma> Hello
[19:41:36 CEST] <tobor> lol can just apt-get directly in a windows bash shell and use Canonical's (ubuntu's) repos in a few weeks.
[19:42:00 CEST] <tobor> sup kiroma
[19:42:34 CEST] <tobor> and yes I did say bash.
[19:42:39 CEST] <kiroma> I am a beggining linux user, where should I put header files to configure ffmpeg properly?
[19:44:01 CEST] <tobor> kiroma, I wrote this a few years back. http://pastebin.com/NHXrsBxm
[19:44:15 CEST] <tobor> prolly does not work today
[19:44:24 CEST] <tobor> what distro you on?
[19:45:08 CEST] <kiroma> linux mint 17.3 and ffmpeg 3.0.2
[19:45:20 CEST] <kiroma> I wanted to configure it so that I could use nvenc
[19:47:11 CEST] <tobor> in the past I would use  checkinstall --pkgname=x264 --pkgversion="3:$(./version.sh | awk -F'[" ]' '/POINT/{print $4"+git"$5}')" --backup=no --deldoc=yes --fstrans=no --default
[19:47:42 CEST] <ChocolateArmpits> tobor: wait does that mean there will be a way to build ffmpeg without using mingw/msys ?
[19:47:58 CEST] <tobor> yeah!
[19:48:03 CEST] <tobor> freaking awesome!
[19:48:04 CEST] <ChocolateArmpits> oh snap
[19:48:22 CEST] <ChocolateArmpits> But it's W10 only ?
[19:48:32 CEST] <tobor> for poor a fsck like me that cant use GNU/Linux at work
[19:49:13 CEST] <tobor> just type in " bash " in shell and poof you have a REAL linux shell.
[19:49:47 CEST] <c_14> kiroma: --extra-cflags='-I/path/to/headers'
[19:50:08 CEST] <c_14> Or just throw the header in /usr/local/include (that's usually in the default include path)
[19:50:20 CEST] <kiroma> Oh, allright. Thanks!
[19:51:19 CEST] <tobor> lol take a look at this. GNU/Linux by default in windows_10 https://www.youtube.com/watch?v=nKtNFZRWO0U
[19:51:40 CEST] <tobor> using gcc or aptitude and what now. Nuts
[19:53:54 CEST] <tobor> The DD's over at Debian to MS to f**k off but Canonical grabbed their toes and told MS to let her rip!
[19:54:51 CEST] <tobor> ChocolateArmpits, not sure if it is only w10
[19:55:40 CEST] <kepstin> it is only win 10, afaik. It's basically like wine except the other way around. They added a compat layer that lets you run native linux apps on the windows nt kernel.
[19:56:36 CEST] <kepstin> and they ship it with an (apparently unmodified) ubuntu userspace, but in theory you can run anything with it.
[19:56:41 CEST] <tobor> but how the heck and say " watch -n1 'grep MHz /proc/cpuinfo | sort -k4rn' work? I mean /proc??????
[19:57:45 CEST] <kepstin> '/proc' is part of the linux kernel api, they're probably simulating it for app compatibility reasons.
[19:57:46 CEST] <tobor> or setting up IPT tables and making a real firewall. That is kernel stuff
[19:58:05 CEST] <kepstin> it almost certainly doesn't support iptables.
[19:58:17 CEST] <tobor> from what I hear it will???? How
[19:58:24 CEST] <kepstin> (you can of course run a full virtual machine with a linux kernel if you want to do that)
[19:58:42 CEST] <tobor> blah only use MS stuff at work.
[20:00:13 CEST] <tobor> hmmm wonder what say uname -a would show?
[20:03:04 CEST] <tobor> what ever. Right now I am just mad that does not exist yet on this windows_10 machine. I don't know windows and I am trying to import video via firewire to a ZFS nas using ffmpeg
[20:03:21 CEST] <tobor> dshow?
[20:04:29 CEST] <tobor> kepstin can I pass though this firewire card with something other them kvm? I use a lot of vfio and qemu at home
[20:11:30 CEST] <durandal_1707> casf: dynaudnorm filter?
[20:15:34 CEST] <tobor> right this is not working - ffmpeg -f iec61883 -i auto -hdvbuffer 100000 testout.mpg . right so I take it libiec61883 is not how it is done in windows.
[20:16:43 CEST] <tobor> what in heck is used here? Not the stack juju
[20:18:12 CEST] <casf> durandal_1707: not yet - just tinkering with volume right now
[21:26:19 CEST] <neuro_sys> I have this command generated programmatically, but it takes 50 seconds. Any chance I could reduce it down?
[21:26:22 CEST] <neuro_sys> https://gist.github.com/neuro-sys/1d857664c5a32cabae4a2b1d0a826a60
[21:27:20 CEST] <neuro_sys> the parts of the video where there's no overlaying (filtering) involved maybe is causing useless demux/remux/decode/encode phases. I'm not sure how it really works internally in that regard.
[21:29:51 CEST] <ChocolateArmpits> neuro_sys: if that's your command then there is encoding happening after the filter
[21:31:03 CEST] <neuro_sys> ChocolateArmpits: Yes it's the command.
[21:31:11 CEST] <neuro_sys> could you elaborate?
[21:31:54 CEST] <ChocolateArmpits> if you don't specify encoding options ffmpeg will then use defaults. Depending on the video format that can result in slow encoding process
[21:31:59 CEST] <neuro_sys> what I'm trying to achieve is to overlay 5 alpha image streams on top of the first video stream
[21:32:20 CEST] <ChocolateArmpits> Your command isn't just overlaying the video but also encoding the result
[21:32:24 CEST] <neuro_sys> but overlays happen only in certain parts of the video; hence enable=between(n,a,b)
[21:32:44 CEST] <neuro_sys> isn't encoding the result naturally required?
[21:33:20 CEST] <ChocolateArmpits> well yeah but you can pick different formats or speed up the encoding by setting proper parameters
[21:33:56 CEST] <ChocolateArmpits> as it is it's encoding to h264 and probably using medium preset
[21:34:07 CEST] <ChocolateArmpits> if the video is hd medium will be quite slow
[21:34:10 CEST] <neuro_sys> It's supposed to be viewed in a browser, and I'm not sure which would be the optimal then
[21:34:45 CEST] <ChocolateArmpits> what's the resolution and framerate of the output ?
[21:35:06 CEST] <neuro_sys> I was thinking if I should just overlay each part of the video seperately using -ss and -t, and then concatenate them to avoid useless decoding/encoding or whatnot if that makes any sense
[21:35:29 CEST] <neuro_sys> because in 60 seconds of video (25 fps: 640x350), there're only 15-20 seconds of overlaying parts
[21:35:41 CEST] <ChocolateArmpits> that's impossible
[21:35:56 CEST] <neuro_sys> I understand
[21:36:09 CEST] <neuro_sys> so it would not have any speed gain, or just not possible?
[21:36:18 CEST] <ChocolateArmpits> of course it's possible
[21:37:36 CEST] <ChocolateArmpits> anyways try this after the map command and before the output "-vcodec libx264 -g 50 -preset veryfast -b:v 900k -vprofile main -movflags faststart -chunk_duration 1000k"
[21:37:56 CEST] <neuro_sys> okay, I'm trying now
[21:38:26 CEST] <ChocolateArmpits> faststart will move the index to the beginning of the file, chunk_duration I think set the chunks of the index in sizes of 1 second here
[21:42:09 CEST] <neuro_sys> wow!
[21:42:22 CEST] <ChocolateArmpits> any faster ?
[21:42:34 CEST] <neuro_sys> yeah, it was in 25 seconds now!
[21:42:39 CEST] <neuro_sys> rather than 50 seconds
[21:42:56 CEST] <neuro_sys> I have no idea how that works, but I'll study them
[21:43:11 CEST] <ChocolateArmpits> do you have specific bitrate requirements ?
[21:43:40 CEST] <neuro_sys> no I'm not told any requirement
[21:43:47 CEST] <c0rnw19> hello world
[21:44:08 CEST] <neuro_sys> I think 1mbp/s but I'll check
[21:44:44 CEST] <ChocolateArmpits> well you can try -vcodec libx264 -g 25 -preset superfast -b:v 1000k -vprofile baseline
[21:44:48 CEST] <ChocolateArmpits> this will be faster
[21:44:58 CEST] <ChocolateArmpits> but the quality will end up worse
[21:45:01 CEST] <ChocolateArmpits> than the previouis
[21:45:14 CEST] <ChocolateArmpits> try for yourself
[21:45:16 CEST] <furq> does -g really make a difference to encoding speed
[21:45:17 CEST] <neuro_sys> hmm, which part reduces the quality in this one? the previous command had -b:v 900k, and this has higher
[21:45:32 CEST] <ChocolateArmpits> furq: well it has to look up less frames
[21:45:41 CEST] <neuro_sys> and what aobut -chucnk_duration?
[21:46:28 CEST] <ChocolateArmpits> neuro_sys: you can add movflags and chunk_duration but your video isn't long so those can be omitted, however they are very important for long videos
[21:48:18 CEST] <ChocolateArmpits> neuro_sys: the specific commands for performance/quality control are "-g"  controls group of picture size, "preset" sets the quality versus speed of the encoder, "-b:v" sets video bitrate, "vprofile" sets the target profile for the encoder, simpler profiles utilize fewer tricks so they are faster
[21:49:02 CEST] <ChocolateArmpits> the important profiles h264 has are baseline, main and high
[21:49:35 CEST] <ChocolateArmpits> you can see the preset explanation here http://dev.beandog.org/x264_preset_reference.html
[21:49:35 CEST] <furq> if you use baseline then you might as well just use preset ultrafast
[21:49:54 CEST] <ChocolateArmpits> well ultrafast should not advised under most circumstances
[21:50:02 CEST] <ChocolateArmpits> it turns off way too much stuff
[21:50:04 CEST] <furq> neither should baseline
[21:50:07 CEST] <ChocolateArmpits> look at the table
[21:50:28 CEST] <ChocolateArmpits> hey mobiles need that baseline
[21:50:43 CEST] <ChocolateArmpits> and if you're running on a toaster you can make use of it too
[21:51:11 CEST] <furq> they both turn off cabac and partitions, which are a big deal
[21:51:46 CEST] <ChocolateArmpits> the guy needs speed, what can I else suggest
[21:52:08 CEST] <ChocolateArmpits> suggesting ultrafast I won't however
[21:52:49 CEST] <furq> i wouldn't suggest baseline either unless you need this video to run on an iphone from 2007
[21:53:24 CEST] <furq> neuro_sys: changing the preset will make the biggest difference by far
[21:54:15 CEST] <furq> you also want to use -crf instead of -b:v
[21:54:37 CEST] <furq> if you have specific bitrate needs then either use the vbv (bad quality) or use 2-pass (slow)
[22:06:05 CEST] <kbarry> I'm trying to better understand some output in ffmplay -loglevel 56: ch:2 chl:stereo fmt:fltp r:44100Hz -> ch:1 chl:mono fmt:s16 r:44100Hz
[22:09:14 CEST] <neuro_sys> as of which version of ffmpeg there's -antialias option available?
[22:09:41 CEST] <kbarry> I'd like to better understand the fmt: portion of the output,
[22:10:13 CEST] <kbarry> I cannot find a lot of information, and might just lack the right words to google. Anyone mind pointing me in the right direction to search?
[22:15:37 CEST] <casf> kbarry: https://ffmpeg.org/ffplay.html#toc-Main-options -f is under there
[22:15:59 CEST] <kbarry> casf:  Thanks, let me go do some reading
[22:16:25 CEST] <vade> under what circumstances would the start time of a stream as reported by ffprobe not be zero ?
[22:18:16 CEST] <ChocolateArmpits> vade: from my experiences, timestamp information can be taken from a clock so the start time will be the start time since the operation has started
[22:18:31 CEST] <ChocolateArmpits> also it can that an audio stream starts earlier than a video stream so the start time will be negative
[22:19:13 CEST] <ChocolateArmpits> my experience
[22:19:59 CEST] <vade> yea, start isnt negative, its a super small value, 0.021333 - and my video and audio packets appear to have PTS and DTS iniciating zero relative to the muxer when I write them
[22:20:03 CEST] <casf> kbarry: this might also help, but it's a lot more reading: https://ffmpeg.org/ffmpeg-formats.html
[22:20:30 CEST] <vade> im also setting start_time to zero for my streams
[22:22:27 CEST] <kbarry> casf:  Thanks. There are two terms I am running into, "fltp" and "s16", I think i understand these for bemore or less, how the bits are arranges, but I can't find any informations that might shed light on the "why you would use one over the other, or what the affects of using s16 over X "
[22:22:50 CEST] <kbarry> I'm only dealing with audio, and have also found that to be a roadblock.
[22:29:39 CEST] <durandal_1707> kbarry: that depends on how codec works
[22:30:10 CEST] <kbarry> Mind expounding a little?
[22:30:37 CEST] <kbarry> you mean things like , X codec supports s16, but not fltp ?
[22:30:46 CEST] <furq> most audio codecs only support one or the other
[22:32:55 CEST] <furq> it's unlikely to be something you'll ever have to consider
[22:34:57 CEST] <vade> oh interesting
[22:35:04 CEST] <vade> looks like ffprobe clips the negagive
[22:35:27 CEST] <vade> if I do ffprobe -show_strams -select_streams a -infile-
[22:35:33 CEST] <vade> I get start_time - 0.021333
[22:35:45 CEST] <kbarry> fruq, I am only asking about it because I am "somehow" producing a stream where the format is changing mid-stream (the stream is being produced ouside of ffmpeg), the stream is going from  mono to stero, from stero to mono, from fltp to s16, etc.
[22:35:51 CEST] <vade> but pure ffpobe shows me positive start time 0.02133
[22:36:24 CEST] <furq> afaik if ffprobe shows fltp -> s16 then that's just a result of it decoding a lossy format
[22:36:38 CEST] <furq> i don't know why it would be changing channel layout, though
[22:36:57 CEST] <furq> s/ffprobe/ffplay/
[22:37:26 CEST] <kbarry> The short version is "Some other tool;" is taking file.mp3, and foo.mp3, and urlstream, and  broadcasting them as an RTMP stream, one after anoter,
[22:37:32 CEST] <kbarry> lets say each changes every 30 seconds.
[22:38:22 CEST] <kbarry> file.mps is mono, fltp, foo.mp3 is stero, s16. as is the url.
[22:39:07 CEST] <kbarry> When a player (like ffplay) tunes into the rtmp stream, everything is cool, until the content, on the back end, changes, and the specs of the audio change,
[22:39:22 CEST] <kbarry> it started mono, fltp, and not the stream is stero s16
[22:39:30 CEST] <kbarry> not = now
[22:47:16 CEST] <neuro_sys> is there a rough performance comparison chart for video encoding formats
[22:51:18 CEST] <furq> not that i know of, but you won't get much faster than x264
[22:52:22 CEST] <darsie> is x264 better than xvid?
[22:52:28 CEST] <furq> yes
[22:52:31 CEST] <darsie> thx
[23:07:28 CEST] <vade> so heres a question. Im using libswresample to resample audio typically from 44.1 to 48 khz - my limited understanding is that libswresamples swr_convert_frame will keep an internal queue for samples since - well, it has to
[23:08:01 CEST] <vade> however, im getting more samples than frame size from my encoder -
[23:09:27 CEST] <vade> my swr_convert_frame expectedly works if im going 48 -> 48 Khz - however, actually trying to call swr_convert_frame doesnt throw and error and I get a valid frame out from what I can tell
[23:09:57 CEST] <vade> do I need to use an AudioFrameQueue post resample just due to frame size nuances?
[23:19:35 CEST] <vade> yea, so my input frame pre resamples has a frame size of 1024
[23:20:19 CEST] <vade> i resample, its is 1098
[23:20:32 CEST] <vade> my codec context is frame_size is 1024
[00:00:00 CEST] --- Thu May 12 2016