[Ffmpeg-devel-irc] ffmpeg.log.20160928

[burek]

[02:09:49 CEST] <SeanC> http://pastebin.com/8qQFYLiM <== Having trouble with ffmpeg Compilation, PKG_CONFIG_PATH isn't acting as expected
[02:12:15 CEST] <JEEB> in that script at least PKG_CONFIG_PATH is on a different line
[02:12:34 CEST] <JEEB> I usually do it PKG_CONFIG_PATH=/a/b/lib/pkgconfig ./configure --params
[02:13:52 CEST] <DHE> alternatively you can put it before the configure command and export it as well
[02:13:55 CEST] <SeanC> Thanks, that seemed to get ./configure to not fail out.
[02:14:01 CEST] <DHE> eg: export PKG_CONFIG_PATH=...
[02:52:24 CEST] <llogan> JEEB: same here. IIRC the guide linked to in the paste had it on the same line but some random user changed it for some random reason.
[03:51:51 CEST] <haimg> hello
[03:52:01 CEST] <haimg> i have a few questions about ffserver
[03:52:05 CEST] <haimg> and a crash report
[03:52:10 CEST] <haimg> for it.
[03:54:12 CEST] <haimg>    frame #0: 0x000000010078f4f2 ffserver_g`avcodec_parameters_from_context [inlined] codec_parameters_reset(par=0x0000000000000000) + 23 at utils.c:4023 [opt]
[03:54:12 CEST] <haimg>    4020	    if (par && par->extradata) {
[03:54:12 CEST] <haimg>    4021	      av_freep(&par->extradata);
[03:54:14 CEST] <haimg>    4022	    }
[03:54:16 CEST] <haimg> -> 4023	    par->extradata = NULL;
[03:54:18 CEST] <haimg>    4024	
[03:54:20 CEST] <haimg>    4025	    memset(par, 0, sizeof(*par));
[03:54:22 CEST] <haimg>    4026	
[03:54:24 CEST] <haimg> (lldb) bt
[03:54:26 CEST] <haimg> * thread #1: tid = 0xb0f580, 0x000000010078f4f2 ffserver_g`avcodec_parameters_from_context [inlined] codec_parameters_reset(par=0x0000000000000000) + 23 at utils.c:4023, queue = 'com.apple.main-thread', stop reason = EXC_BAD_ACCESS (code=1, address=0x10)
[03:54:28 CEST] <haimg>   * frame #0: 0x000000010078f4f2 ffserver_g`avcodec_parameters_from_context [inlined] codec_parameters_reset(par=0x0000000000000000) + 23 at utils.c:4023 [opt]
[03:54:30 CEST] <haimg>     frame #1: 0x000000010078f4db ffserver_g`avcodec_parameters_from_context(par=0x0000000000000000, codec=0x0000000103803800) + 11 at utils.c:4083 [opt]
[03:54:32 CEST] <haimg>     frame #2: 0x0000000100016349 ffserver_g`prepare_sdp_description(stream=<unavailable>, pbuffer=<unavailable>, my_ip=(s_addr = 0)) + 473 at ffserver.c:2999 [opt]
[03:54:34 CEST] <haimg>     frame #3: 0x0000000100014cb9 ffserver_g`handle_connection + 558 at ffserver.c:1688 [opt]
[03:54:36 CEST] <haimg>     frame #4: 0x0000000100014a8b ffserver_g`handle_connection(c=<unavailable>) + 17739 at ffserver.c:1006 [opt]
[03:54:38 CEST] <haimg>     frame #5: 0x000000010000faec ffserver_g`main [inlined] http_server + 47 at ffserver.c:777 [opt]
[03:54:40 CEST] <haimg>     frame #6: 0x000000010000fabd ffserver_g`main(argc=<unavailable>, argv=<unavailable>) + 4205 at ffserver.c:4032 [opt]
[03:54:42 CEST] <haimg>     frame #7: 0x00007fff912855ad libdyld.dylib`start + 1
[03:54:44 CEST] <haimg> oops
[03:55:00 CEST] <haimg> here is: http://pastebin.com/Bav5ZpnP
[03:55:08 CEST] <alex88> hi guys :) I'm currently running ffmpeg from our webapp when an event is triggered to start recording, this way `ffmpeg -i rtsp://@192.168.1.15:554/play1.sdp -acodec copy -vcodec copy file.mp4` the problem is that it starts recording after like 5-6 seconds, is there a way to "control" ffmpeg to pause/start recording?
[04:03:32 CEST] <haimg> here is the ticket for the reported issue: https://trac.ffmpeg.org/ticket/5869#ticket
[04:39:23 CEST] <spoon_> hey guys, i updated my ubuntu server from 14.04lts to 16.04lts and after that ffmpeg cant probe files anymore
[04:40:13 CEST] <spoon_> ive purged the installs and reinstalled them and even updated ffmpeg 3 and its the same error
[04:50:33 CEST] <spoon_> am i am to get help here?
[05:08:29 CEST] <spoon_> can anyone help me?
[05:17:04 CEST] <spoon_> http://pastebin.com/twCWfL2c
[05:17:53 CEST] <c_14> -v quiet isn't very helpful
[05:18:01 CEST] <c_14> Run just the ffprobe on the file please
[05:18:07 CEST] <c_14> ffprobe file.mp4
[05:18:13 CEST] <c_14> Then upload the output from that to a pastebin
[05:21:58 CEST] <spoon_> i have no idea why but after i manually ran probe on the file the whole script started working...
[05:23:04 CEST] <c_14> Heisenbugs \o/
[08:55:06 CEST] <tsoanety> Hi everybody
[08:56:24 CEST] <tsoanety> I search since 2 days the way to stream audio only to youtube using ffmpeg alsa and the audio input but  i didnt find
[08:56:34 CEST] <tsoanety> In despair i contact you
[08:57:15 CEST] <tsoanety> I can record the input to a file but i cant stream the audio file afterwards
[08:57:19 CEST] <tsoanety> So please
[08:57:32 CEST] <tsoanety> Can somebody help me
[08:57:36 CEST] <tsoanety> Thanks
[10:13:17 CEST] <quantumsm> It is possible to use drawtext filter and the text to start scrolling each time from a random y position ?
[10:14:09 CEST] <retard> are you trying to make a niconico thing?
[10:15:27 CEST] <quantumsm> more exactly ?
[10:36:30 CEST] <annoymouse4210> IS there way to get estimate duration while transcoding?
[11:32:59 CEST] <nonex86> when i open some file with h264/avc stream using ffmpeg api how can i understand is it interlaced (paff/mbaff) or progressive?
[11:53:50 CEST] <skkkssks> is it possible to join multiple videos with different bitrate and resolution ?
[11:55:02 CEST] <skkkssks> also I wanna download a stem cell video
[11:56:21 CEST] <skkkssks> how to do it with ffmpeg ? link - http://video.limelight.com/player/fp10loader.swf?deepLink=true&playerForm=3408cce3252244e9b8db28df9f21f924&channelId=dd542dc018c44edc8bdcafe9cd63b1b7&85527519
[12:58:01 CEST] <tobiasBora> Hello,
[12:59:26 CEST] <tobiasBora> I would like to have some advices on video format : I would like to have if possible an open source container format, which provide quality usable in streming (I want to put my video on a owncloud website), and a file not too big (I cannot use 1G for 10mn of video)
[12:59:35 CEST] <tobiasBora> *streaming
[13:00:01 CEST] <tobiasBora> and with these things I would like to have a quality not too bad.
[13:00:07 CEST] <tobiasBora> For the moment I do something like :
[13:00:17 CEST] <tobiasBora> avconv -i myfile.MOV myfile.webm
[13:00:45 CEST] <tobiasBora> without any option I can have a pretty good improvment (size divided by 25)
[13:01:34 CEST] <tobiasBora> and the quality isn't horrible (but sometimes it's a bit bad, and even with the option -q:v 1 I don't have better quality...)
[13:07:57 CEST] <tobiasBora> Does anyone has good "numbers" to have a pretty good output result for the less space expensive usage ?
[13:09:02 CEST] <BtbN> Well, everything in ffmpeg is opensource!
[13:09:22 CEST] <BtbN> And the container has no effect on the quality at all. Some are not streamable though
[13:24:15 CEST] <limbo_> tobiasBora: replace avconv with ffmpeg and that will just work.
[13:24:52 CEST] <limbo_> If it's not "streamable", whatever the equivelent of a moov atom in a webm should be moved to the start.
[13:25:42 CEST] <tobiasBora> limbo_: Isn't avconv supposed to replace ffmpeg ?
[13:25:50 CEST] <BtbN> no?
[13:26:10 CEST] <BtbN> You must be using some ancient version of libav if that message still shows up.
[13:26:19 CEST] <furq> that question mark is misplaced considering how often this comes up
[13:26:29 CEST] <tobiasBora> BtbN: By open source I mean the format. For example the licences around mp3 are quite strange right ? And I think it's the same thing for some video format.
[13:26:47 CEST] <furq> most of the patents for mp3 have expired now
[13:26:53 CEST] <furq> the last ones expire in the us in december 2017
[13:27:33 CEST] <tobiasBora> So what is the difference between ffmpeg and avconv ?
[13:27:45 CEST] <BtbN> One is ffmpeg, the other is libavs equivalent of ffmpeg.
[13:27:55 CEST] <furq> is libav still being developed
[13:28:00 CEST] <furq> it seems pretty dead
[13:28:02 CEST] <BtbN> yes, but nobody uses it anymore.
[13:28:20 CEST] <furq> what about the people on old ubuntu LTS releases
[13:28:26 CEST] <furq> or did you mean by choice
[13:28:45 CEST] <tobiasBora> So ffmepg is still under heavy development, while avconv/libav is nearly dead ?
[13:29:12 CEST] <furq> something like that
[13:29:23 CEST] <furq> debian and ubuntu are back on real ffmpeg now
[13:29:37 CEST] <furq> i don't know of any distro which is still on libav
[13:29:56 CEST] <tobiasBora> ok thank you
[13:30:16 CEST] <furq> as far as i can tell the whole mess pretty much happened because the debian ffmpeg maintainer was one of the libav fork guys
[13:30:45 CEST] <furq> so he took it upon himself to declare that ffmpeg was deprecated
[13:30:58 CEST] <limbo_> tobiasBora: there's no need to license mp3, the issue with it is patents.
[13:31:07 CEST] <limbo_> which are no an issue outside of the US.
[13:32:11 CEST] <tobiasBora> limbo_: All right... Same thing for mp4 ?
[13:32:25 CEST] <furq> if you mean h264 and aac then yes
[13:32:31 CEST] <furq> except those patents don't expire any time soon
[13:32:51 CEST] <limbo_> tobiasBora: yes.
[13:33:16 CEST] <limbo_> software patents are mainly an issue in the US. Merely using mp3 does not mean you have to pay royaltees.
[13:33:44 CEST] <limbo_> But people who don't want mp3 decodin software on their computer won't be able to play them.
[13:34:04 CEST] <limbo_> ogg is the popular alternative.
[13:34:10 CEST] <furq> it's pretty much a non-issue for personal use
[13:34:14 CEST] <tobiasBora> Well some people needs to pay right ? For example when you install ubuntu you are asked if you want to install mp3 stuff
[13:34:28 CEST] <limbo_> yes
[13:34:28 CEST] <furq> to such an extent that even debian ships ffmpeg with x264 and lame builtin
[13:34:43 CEST] <limbo_> because canonical doesn't want to get sued.
[13:34:53 CEST] <furq> is that still true
[13:34:56 CEST] <limbo_> linux mint doesn't ask. Just does it.
[13:35:18 CEST] <furq> anything which uses libavcodec on ubuntu will support mp3 regardless of your choice afaik
[13:35:29 CEST] <furq> at least for newer ubuntu releases
[13:35:54 CEST] <tobiasBora> All right... But I have a question : why can I convert into mp4 in better quality than into webm ?
[13:36:07 CEST] <furq> because mp4 has sensible default settings and webm doesn't
[13:36:30 CEST] <furq> you need to set at least -b:v for webm (vp9) to look acceptable
[13:36:39 CEST] <furq> you probably want something like -crf 30 -b:v 0
[13:37:58 CEST] <limbo_> tobiasBora: if you don't specify quality yourself, ffmpeg uses its default. -q:v <number> wll do that for the video. (lower number is higher quality)
[13:38:08 CEST] <furq> does -q do anything with vp9
[13:39:01 CEST] <tobiasBora> furq: Great it's much better ! And do you know why I can convert mp4 in 6x less time than webm ?
[13:39:06 CEST] <limbo_> from what I understand, that's mapped to whateverthe quality settings are for the output stream.
[13:39:15 CEST] <furq> because x264 is much faster than libvpx
[13:39:17 CEST] <limbo_> tobiasBora: hardware accelleration probably.
[13:39:20 CEST] <furq> no
[13:39:25 CEST] <limbo_> ahh. ok.
[13:39:36 CEST] <furq> i don't know that -q is actually mapped to anything with vp9
[13:39:37 CEST] <limbo_> does ffmpeg not use accelleration?
[13:39:39 CEST] <furq> i've not checked though
[13:39:45 CEST] <tobiasBora> limbo_: In both encoding and decoding, or only encoding ?
[13:39:45 CEST] <furq> x264 is a software encoder
[13:40:01 CEST] <furq> you can use a hardware encoder if you want a video which is worse quality
[13:40:33 CEST] <tobiasBora> Ok, thank you for your help !
[13:40:47 CEST] <limbo_> furq: how can you be sure it's worse quality?
[13:41:03 CEST] <limbo_> in all cases
[13:41:06 CEST] <furq> because all of the consumer hardware h264 encoders are much worse than x264
[13:41:22 CEST] <furq> maybe some of the pro ones are better but i don't think ffmpeg supports those
[13:41:23 CEST] <tobiasBora> furq: You can use GPU with ffmpeg ?
[13:41:34 CEST] <furq> if your gpu has a video encoder on it, sure
[13:41:55 CEST] <furq> i think opencl is supported for some video filters, but otherwise you can't use it for encoding
[13:41:56 CEST] <tobiasBora> and how could I ask to ffmpeg to use it ?
[13:42:10 CEST] <furq> https://trac.ffmpeg.org/wiki/HWAccelIntro
[13:45:33 CEST] <tobiasBora> Too bad, opencl isn't usable in ffmpeg in encoding...
[14:26:28 CEST] <lovetruth> hello!!... :)
[14:27:42 CEST] <lovetruth> I've been trying hard to restream some rtmp using ffserver
[14:27:54 CEST] <lovetruth> but can't seem to make it work...
[14:28:37 CEST] <lovetruth> if you enter www.vestic.ro -> something will pop asking if you want to watch live. I want to re-stream that stream, using ffserver ...
[14:28:45 CEST] <lovetruth> can't seem to get ffserver.conf right...
[14:29:58 CEST] <lovetruth> http://pastebin.com/9HuXYGsD <- this is what I've been trying so far...
[14:33:14 CEST] <lovetruth> any help, please?...
[14:35:07 CEST] <nonex86> well, nowdays usual answer here is "don't use ffserver" - "ffserver is unsupported now"/"ffserver suck"
[14:35:37 CEST] <lovetruth> ok
[14:35:44 CEST] <lovetruth> so what do you recommend me for that?...
[14:36:23 CEST] <lovetruth> (restream something to around 50 clients)
[14:36:56 CEST] <nonex86> never work on this kind, sorry... maybe nginx with rtmp module?
[14:37:19 CEST] <nonex86> at least for streaming
[14:37:55 CEST] <lovetruth> ps, why ffserver suck?...
[14:38:44 CEST] <nonex86> dont know, just repeat someones answer from this channel :)
[14:39:00 CEST] <lovetruth> thanks :)
[14:39:30 CEST] <nonex86> maybe because its not developed anymore
[14:39:39 CEST] <furq> nginx-rtmp will restream rtmp just fine
[14:39:50 CEST] <furq> and yeah ffserver is going to be deprecated in 3.2
[14:40:02 CEST] <furq> it's not been developed for ages and it was never very good when it was
[14:40:26 CEST] <lovetruth> and if I want some transcoding?...
[14:40:47 CEST] <lovetruth> like from mpeg2 to h264? or the audio or some encoding?...
[14:41:12 CEST] <bencoh> you can fork ffmpeg from nginx-rtmp iirc
[14:41:14 CEST] <furq> you can exec ffmpeg from within nginx-rtmp
[14:41:26 CEST] <furq> snap
[14:41:29 CEST] <lovetruth> I've already installed ffmpeg
[14:41:39 CEST] <lovetruth> should I uninstall it and install nginx rtmp?...
[14:41:50 CEST] <furq> there's no need to uninstall ffmpeg
[14:41:59 CEST] <lovetruth> I have (ubuntu studio 14.04 x64)
[14:42:05 CEST] <bencoh> isn't it documented in nginx-rtmp?
[14:42:14 CEST] <lovetruth> haven't read nginx rtmp yet :)
[14:42:17 CEST] <bencoh> ah :)
[14:42:25 CEST] <furq> i mean you probably don't want to use the ffmpeg in ubuntu 14's repos
[14:42:29 CEST] <furq> assuming that's the one you've got
[14:42:33 CEST] <lovetruth> no,
[14:42:40 CEST] <lovetruth> compiled from source code
[14:43:18 CEST] <lovetruth> I have the compiled from source code* version, I meant :)
[14:43:32 CEST] <furq> also rtmp only really supports h264 and aac, so there's no need to reencode if you're only using rtmp
[14:43:45 CEST] <furq> it supports other codecs but they're not worth mentioning
[14:44:06 CEST] <lovetruth> I might be going to receive some mpeg2 stream, but for firsts, I just wanted to restream rtmp
[14:44:09 CEST] <bencoh> mp3? ;>
[14:58:27 CEST] <lovetruth> ok... I have this html source code of some embed code http://paste.ee/p/vpruG
[14:58:40 CEST] <lovetruth> I've installed nginx-rtmp
[14:58:58 CEST] <lovetruth> but now? what?...
[14:59:57 CEST] <lovetruth> I don't have open broadcast server & neither I own the server which streams to me. I just own another server which I want to use to restream that stream
[15:00:20 CEST] <lovetruth> I'm at testing... part -> but no test seems to work...
[15:07:07 CEST] <furq> lovetruth: https://github.com/arut/nginx-rtmp-module/wiki/Directives#exec_pull
[15:18:13 CEST] <spacemadman> hey guys
[15:18:38 CEST] <spacemadman> quick question.. I have a batch script and I have ffmpeg installed.. any idea how to get it to execute?
[15:18:44 CEST] <spacemadman> I am failing miserably at this
[15:32:45 CEST] <DHE> sounds like ffmpeg isn't in your PATH or current directory
[15:35:22 CEST] <skkkssks> how to download swf compressed file stream using ffmpeg ?
[15:43:12 CEST] <skkkssks> anyone ?
[15:43:24 CEST] <skkkssks> how to download swf compressed file stream using ffmpeg ?
[15:48:16 CEST] <limbo_> download?
[15:48:23 CEST] <limbo_> what're you trying to do?
[15:50:08 CEST] <alain-francois> bonjour mesdames messieurs
[15:50:33 CEST] <alain-francois> je ne connais absolument rien à ffmpeg , j'ai donc besoin d'aide
[15:50:54 CEST] <alain-francois> je suis entrain  d'enregistrer une source analogique
[15:51:08 CEST] <limbo_> I don't speak french, can you ask again in english?
[15:51:10 CEST] <alain-francois> avec la commande suivante :
[15:51:20 CEST] <alain-francois> excuse me
[15:51:37 CEST] <alain-francois> i'm recording analog source
[15:52:08 CEST] <alain-francois> with the following command , i have the video but not the audio
[15:53:47 CEST] <limbo_> what is the command?
[15:53:53 CEST] <alain-francois> ffmpeg -f v4l2 -framerate 25 -video_size 640x480 -i /dev/video1 output.mkv
[15:54:00 CEST] <alain-francois> here it is
[15:54:43 CEST] <alain-francois> i wonder : what is the command  ?
[15:54:48 CEST] <limbo_> I don't think there is audio from those kinds of video devices.
[15:55:01 CEST] <alain-francois> my audio is in hw:4.0 alsa
[15:55:27 CEST] <limbo_> right. But you're not specifying that as an input to ffmpeg
[15:56:00 CEST] <alain-francois> excuse me , i'm too speed
[15:56:40 CEST] <limbo_> you need to give ffmpeg an audio input.
[15:57:18 CEST] <alain-francois> it seems cool , for me , that ffmpeg input autorizes video AND audio . otherwise it means nothing
[15:57:53 CEST] <skkkssks> limbo_: I want a sterm-cell video, I wanna save it locally. link - http://video.limelight.com/player/fp10loader.swf?deepLink=true&playerForm=3408cce3252244e9b8db28df9f21f924&channelId=dd542dc018c44edc8bdcafe9cd63b1b7&85527519
[16:00:28 CEST] <kepstin> alain-francois: ffmpeg does not magically know which audio device and which video device go together. You have to give ffmpeg 2 separate inputs, one for video, one for audio.
[16:01:00 CEST] <alain-francois> ok . and now , how to do it  ?
[16:01:21 CEST] <kepstin> skkkssks: use the 'youtube-dl' tool (it supports limelight, despite the name)
[16:01:23 CEST] <limbo_> skkkssks: what page is it embedded in?
[16:01:27 CEST] <skkkssks> limbo_: when downloading it with ffmpeg command - ffmpeg -i "http://video.limelight.com/player/fp10loader.swf?deepLink=true&playerForm=3408cce3252244e9b8db28df9f21f924&channelId=dd542dc018c44edc8bdcafe9cd63b1b7&85527519" -vcodec copy -acodec copy out.mp4
[16:01:46 CEST] <skkkssks> it says swf compressed file detected
[16:01:48 CEST] <limbo_> That URL doesn't work for me. It just loads forever.
[16:02:07 CEST] <limbo_> that swf file desn't contain a video, it just loads and plays one.
[16:02:26 CEST] <skkkssks> limbo_: http://thelongevitystudy.com/160725A.php
[16:03:05 CEST] <skkkssks> embeded page ^^^
[16:03:07 CEST] <kepstin> alain-francois: add "-f alsa -i hw:4.0" as a second input.
[16:03:53 CEST] <kepstin> er, actually I think it's "-f alsa -i hw:4,0" - comma, not period
[16:04:00 CEST] <limbo_> skkkssks: doesn't work for me either.
[16:04:16 CEST] <alain-francois> i'll try it , thanks
[16:04:35 CEST] <skkkssks> limbo_: how to do it with ffmpeg or rtmp ?
[16:05:31 CEST] <limbo_> I can't even get the video to play/load.
[16:05:50 CEST] <limbo_> Look at the network monito in your browser, if yo're using chrome.
[16:06:03 CEST] <limbo_> You should see a URL that i's downloadin the file from.
[16:06:12 CEST] <skkkssks> thats surprising limbo_ maybe you dont have flash installed or activated
[16:06:39 CEST] <limbo_> I do.
[16:06:43 CEST] <skkkssks> limbo_: yeah I got this link from there only, otherwise there is nothing in chrome or firefox network monitor
[16:07:33 CEST] <skkkssks> if you want I can screenshot and upload - chrome and firefox network monitor ?
[16:08:04 CEST] <alain-francois> root at debian:/home/alain# ffmpeg -f v4l2 -framerate 25 -video_size 640x480 -i /dev/video1 -f alsa -i hw:4.0 output.mkv
[16:08:33 CEST] <alain-francois> gave me
[16:08:44 CEST] <alain-francois> ALSA lib pcm_hw.c:1667:(_snd_pcm_hw_open) Invalid value for card
[16:08:44 CEST] <alain-francois> [alsa @ 0x1075a20] cannot open audio device hw:4.0 (No such device)
[16:08:44 CEST] <alain-francois> hw:4.0: Input/output error
[16:08:56 CEST] <kepstin> alain-francois: please read my second message :)
[16:09:06 CEST] <limbo_> skkkssks: go on. The firefox one probably won't catch it. Make sure the video is partway played when you make the screenshot though.
[16:09:37 CEST] <skkkssks> limbo_: are you on firefox ? share screen ?
[16:09:51 CEST] <kepstin> skkkssks: limbo_ the video plays for me, it looks like a flash player pulling from either rtmp or hls, it's unclear
[16:10:09 CEST] <kepstin> so you'd have to find the urls of the underlying stream to use it with ffmpeg
[16:10:23 CEST] <kepstin> youtube-dl's limelight extractor doesn't work on that page for me, but it might be a start :/
[16:10:36 CEST] <skkkssks> kepstin: I played video on chrome and IDM is capturing it as RTMPE stream but not downloading
[16:11:22 CEST] <alain-francois> wht's the difference kepstin ?
[16:11:28 CEST] <skkkssks> how to parse that link so as to get the direct link of video and type of stream including other info ?
[16:11:41 CEST] <skkkssks> kepstin: ^^^^ limbo_
[16:11:54 CEST] <kepstin> alain-francois: to repeat myself, "it's "-f alsa -i hw:4,0" - comma, not period"
[16:12:03 CEST] <limbo_> skkkssks: give me a screenshot of of the network requests while playing the video.
[16:12:37 CEST] <skkkssks> kk
[16:12:43 CEST] <alain-francois> what is my error ? i copied it strictly
[16:13:09 CEST] <kepstin> alain-francois: you wrote "hw:4.0" I said "hw:4,0"
[16:13:30 CEST] <kepstin> alain-francois: make your font bigger if you can't see that :)
[16:13:46 CEST] <furq> maybe he was converting it into french
[16:14:28 CEST] <alain-francois> thanks kepstin , i'm =triying it .
[16:15:03 CEST] <alain-francois> excuse me for a so stupid error
[16:20:11 CEST] <alain-francois> ok kepstin , you'dre right . your command is ok . fine thanks
[16:24:38 CEST] <spacemadman> hey guys, can ffmpeg make a MXF InterOP MPEG for Cinemas?
[16:27:40 CEST] <kepstin> spacemadman: probably, if you can find someone who knows the spec well enough to pick the correct parameters.
[16:38:39 CEST] <skkkssks> limbo_: kepstin 1st image - livecell homepage with video loading http://www.tiikoni.com/tis/view/?id=781ab2c
[16:39:15 CEST] <skkkssks> limbo_: kepstin 2nd image with network monitor in firefox - http://www.tiikoni.com/tis/view/?id=28e6c9e
[16:39:39 CEST] <skkkssks> limbo_: kepstin 3rd image with network monitor in chrome\
[16:39:48 CEST] <skkkssks> limbo_: kepstin 3rd image with network monitor in chrome -
[16:39:59 CEST] <skkkssks> limbo_: kepstin 3rd image with network monitor in chrome - http://www.tiikoni.com/tis/view/?id=f67d398
[16:39:59 CEST] <kepstin> that's not chrome, that's firefox
[16:40:03 CEST] <kepstin> oh
[16:40:06 CEST] <limbo_> I told you, firefox won't capture requests flash makes.
[16:40:31 CEST] <kepstin> hmm, chrome doesn't seem to have it either
[16:40:48 CEST] <kepstin> so it's not http requests then, i don't think chrome shows rtmp in the network monitor
[16:41:16 CEST] <limbo_> It should show all net requests flash makes.
[16:41:32 CEST] <skkkssks> if you look in chrome screenshot- there is IDm download dialogue, its the only source which tells me that its an RATMPE stream
[16:43:08 CEST] <spacemadman> kepstin: Thats a tough one.. there's a tool called OpenDCP that I found, but it seems to only accept m2v's and wav's as seperate streams to make the MXF Mpeg
[16:44:28 CEST] <spacemadman> if i need to convert a 25fps mov to a seperate m2v and wav at 24fps, how would i achieve that?
[16:44:47 CEST] <DHE> the -map parameter
[16:45:07 CEST] <skkkssks> theres rtmp (rtmpdump) which is kinda universal tool and embedded in all other tools universally so as to get rtmp streams, but I dont know how to use it
[16:45:11 CEST] <spacemadman> DHE: can you please help elaborate?
[16:46:07 CEST] <skkkssks> my 1st concern is how to analyse a stream link so as to know what kind of stream it is and 2nd get its direct link
[16:46:42 CEST] <skkkssks> kepstin: limbo_ ^^^
[16:48:20 CEST] <kepstin> skkkssks: you might want to join #youtube-dl irc channel, they have a bunch of folks who do this sort of thing regularly.
[16:48:45 CEST] <kepstin> since their main thing is to write a tool to download web videos from various sites :)
[16:49:42 CEST] <spacemadman> kepstin: any feedback for me? Is it possible to convert a mov which is at 25fps to a m2v video file and a seperate wav file at 24fps and keep their lengths the same?
[16:50:04 CEST] <kepstin> er, wav files don't have fps...
[16:50:54 CEST] <kepstin> or do you mean you have to convert the video from 25 to 24fps? ffmpeg will do that if you use the '-r' output option or 'fps' video filter (both do the same thing), but it'll make the video a bit juddery since it does it by dropping frames.
[16:52:09 CEST] <kepstin> a lot of 24-25fps conversions are done by simply slowing down or speeding up the video/audio (usually with pitch correction on the audio, nowadays), you might want to consider that.
[16:52:27 CEST] <SouLShocK> spacemadman https://github.com/bcoudurier/FFmbc is it one of these formats you're looking for?
[16:53:11 CEST] <durandal_1707> SouLShocK: are you ffmbc promoter?
[16:53:14 CEST] <SouLShocK> no
[16:53:18 CEST] <spacemadman> kepstin: sorry, i didnt mean convert the audio to 24fps, but meant to keep the durations the same.
[16:53:22 CEST] <SouLShocK> I just happen to use it
[16:54:27 CEST] <spacemadman> SouLShocK: thanks a bunch for that link
[16:55:13 CEST] <BtbN> "This branch is 441 commits ahead, 50144 commits behind FFmpeg:master." highly optimized for professional usage tho!
[16:55:16 CEST] <SouLShocK> haha
[16:55:20 CEST] <BtbN> I'd recommend against using that.
[16:55:20 CEST] <kepstin> spacemadman: but yeah, ffmpeg should have no issue making mxf with the required codecs, you just have to make sure you're using the correct codec settings
[16:55:33 CEST] <SouLShocK> yeah I'd try ffmpeg first. it does have mxf support
[16:57:05 CEST] <spacemadman> need to figure how. The results that I've achieved so far is giving me video with a longer duration than the audio in OpenDCP
[16:58:16 CEST] <skkkssks> any rtmp channel ?
[16:58:18 CEST] <spacemadman> I've used this command to convert the video to m2v | ffmpeg -i "input" -r 24 -vcodec mpeg2video -pix_fmt yuv420p -s 1920x1080 -aspect "16:9" -b 40000k -intra -an video-out.m2v
[16:58:48 CEST] <spacemadman> And this one for audio | ffmpeg -y -i "input" -filter:a "atempo=0.96" -acodec pcm_s24le -ar 48000 audio.wav
[16:59:10 CEST] <kepstin> spacemadman: that video command leaves the video the same length, but the audio command shortens the audio
[16:59:32 CEST] <kepstin> er, lengthens the audio
[17:00:10 CEST] <kepstin> if you want to slow down the video to match the audio, you need to put '-r 24' *before* the input file, so it rewrites the timestamps on the frame. (or there's things you can do with setpts filters alternatively)
[17:31:45 CEST] <NapoleonWils0n> hi all
[17:32:16 CEST] <NapoleonWils0n> is there a way to suppress dts invalid timestamp errors without using log level 9 which hides everything
[17:32:52 CEST] <NapoleonWils0n> i can redirect output to dev null, but is there a way to disable invalid timestamp errors
[18:33:21 CEST] <ac_slater> hey all. Using the libraries.. if I wanted to have a read loop for video, like : `while(av_read_frame(...)) { foo(); usleep(??); }`. How would I calculate the amount of time between frames?
[18:33:36 CEST] <ac_slater> I know I need to use PTS/DTS but maybe I need the use the clock too?
[18:33:51 CEST] <agrathwohl> My current build of FFmpeg (3.1.3) does not appear to support the `-aq-mode 4` option for libvpx-vp9. Is there any way to pass custom options that are available in vpxenc to FFmpeg command line? Or is there any plan to implement the 4th aq-mode option? (in vpxenc --help the option is listed as `equator360`)
[18:34:23 CEST] <durandal_1707> what that do?
[18:34:58 CEST] <agrathwohl> It's a quantization mode specific to equirectangular monoscopic 360 video @durandal_1707
[18:36:15 CEST] <kepstin> ac_slater: depends on the what your video source is. If you're reading from a live stream, webcam, screen capture, etc - you want to read as often as possible (no waiting), since the read will block until the next frame is ready
[18:36:52 CEST] <ac_slater> kepstin: a file sadly
[18:37:34 CEST] <ac_slater> kepstin: I was just piping my file to localhost via udp before and having my application read from it ;). But I can't do that abynire
[18:37:36 CEST] <ac_slater> anymore *
[18:37:39 CEST] <kepstin> ac_slater: hmm, so then you want to emulate what the ffmpeg tool does when you use the '-re' option.
[18:37:58 CEST] <ac_slater> kepstin: yea essential. And I've looked at that path in `ffmpeg` and `ffplay` before.
[18:38:11 CEST] <ac_slater> It' hard to follow
[18:40:58 CEST] <ac_slater> kepstin: maybe I just do it based on fps.
[18:41:22 CEST] <ac_slater> kepstin: but what about formats that have weird PTS/DTS combos (like h264 with b frames)
[18:42:06 CEST] <kepstin> ac_slater: the basic loop should be decode a frame, wait until it's the correct time to show it (based on pts), repeat.
[18:42:26 CEST] <ac_slater> how do I deduce real time from PTS ?
[18:42:29 CEST] <ac_slater> kepstin:  ^
[18:42:49 CEST] <kepstin> ac_slater: multiply by the time base to get seconds. There should be some AVRational helper functions for that
[18:42:57 CEST] <ac_slater> ooo
[18:46:23 CEST] <kepstin> (to avoid floating-point accuracy issues, you should probably subtract the next frame pts from current frame pts, then use that to determine how long you need to wait between frames)
[18:46:41 CEST] <ac_slater> very interesting
[18:46:45 CEST] <ac_slater> I'll try that
[18:46:48 CEST] <ac_slater> thanks mate
[18:47:46 CEST] <BtbN> isn't av_read_frame a blocking call?
[18:47:51 CEST] <BtbN> So no need to sleep at all
[18:48:02 CEST] <skokkk> anyone able to help with my question?
[18:49:33 CEST] <ac_slater> BtbN: it's only blocking if the avio* read call is
[18:49:43 CEST] <ac_slater> in the case of files, fread() isn't blocking
[18:50:02 CEST] <BtbN> well, because it has the data readily available.
[18:50:17 CEST] <ac_slater> BtbN: If I use `-re` on the command line ffmpeg tool, then it sleeps
[18:51:55 CEST] <BtbN> the re logic in ffmpeg.c seems quite complex
[18:52:19 CEST] <skokkk> hi?
[18:54:28 CEST] <ac_slater> skokkk: did you ask your questi
[18:54:31 CEST] <ac_slater> question  *
[18:57:04 CEST] <ac_slater> kepstin: would AVPacket::duration help me?
[18:57:48 CEST] <kepstin> ac_slater: no, that's only for things which have durations (mostly audio packets, which contain multiple samples)
[18:58:09 CEST] <ac_slater> makes sense
[18:58:11 CEST] <ac_slater> thanks
[18:59:46 CEST] <skokkk> hi. I have this command: "ffmpeg -ss %o -t %d -i %s -s %wx%h -v 0 -b:v %bk -maxrate %bk -bufsize 256k -flags -global_header -map 0:0 -map 0:%k -ac 2 -f mpegts -c:v libx264 -preset superfast -c:a aac -b:a 96k -strict -2 -threads 0 -copyts -" from a web browser media player, what can I do to improve the quality and speed of this? It's live encoding and looks like $#!T, even if it's a 1080p movie
[18:59:54 CEST] <skokkk> ^
[19:00:14 CEST] <ac_slater> that's too much for us to help you with honestly. Check the ffmpeg H.264 wiki
[19:00:24 CEST] <skokkk> how to improve qaulity with that? quality is about 240p after that
[19:00:28 CEST] <c_14> increase the bitrate
[19:00:30 CEST] <ac_slater> you're not doing any profile or preset settings
[19:00:46 CEST] <c_14> ac_slater: he has -preset superfast
[19:00:55 CEST] <ac_slater> I see it now
[19:00:57 CEST] <c_14> skokkk: Either increase the bitrate or make the preset slower
[19:01:56 CEST] <ac_slater> kepstin: sorry to keep coming back to this. Which AVRational helper would help be multiply PTS by timebase? q2d?
[19:02:23 CEST] <skokkk> c_14: what would you recommend for a lower cpu usage but higher quality video (if possible, if not, which would you recommend I increase/decrease, or both?)
[19:02:59 CEST] <c_14> increase bitrate
[19:03:00 CEST] <ac_slater> skokkk: https://trac.ffmpeg.org/wiki/Encode/H.264
[19:03:12 CEST] <c_14> lowering the preset will use more cpu than increasing the bitrate will
[19:03:13 CEST] <kepstin> ac_slater: you probably want to do something like convert the pts+timebase to an integer no. of e.g. milliseconds or nanoseconds (depending what clock/sleep functions you're suing). I think you want av_rescale_q() for that.
[19:03:36 CEST] <ac_slater> that's exactly what I want kepstin
[19:05:15 CEST] <skokkk> ac_slater: what is recommended buffer size for 90 minute video?
[19:05:46 CEST] <ac_slater> depends on what's going over the wire in the length of the buffer
[19:06:07 CEST] <kepstin> skokkk: buffer size has no relation to video length
[19:06:11 CEST] <ac_slater> kepstin: The args to av_rescale_q are confusing :(
[19:06:32 CEST] <skokkk> ac_slater: kepstin hmm ok, what would be a recommended buffer size based on XXXX?
[19:06:49 CEST] <kepstin> ac_slater: I think you want something like milliseconds = av_rescale_q(pts, time_base, {1, 1000}) but not sure, haven't worked with this recently :)
[19:07:43 CEST] <ac_slater> skokkk: here is what you need to think about: You're streaming some number of 188byte (MPEGTS) packets that have SOME h.264 data in each packet. You're forcing your video bitrate to something that is causing ffmpeg to transcode (re-encode) your video. This takes time.
[19:08:04 CEST] <ac_slater> skokkk: do this. Remove the audio from your command line, with `-an`. Just focus on the video
[19:08:15 CEST] <ac_slater> remove bufsize
[19:08:30 CEST] <ac_slater> and reduce it down so it JUST PLAYS THE ACTUAL FILE AS IS
[19:09:02 CEST] <ac_slater> (and it seems that you always re-encode since you're input isnt always h.264, that's fine)
[19:09:37 CEST] <ac_slater> you're also changing the resolution as well...this all takes considerable amount of CPU time (and will add to the latency of the viewing)
[19:09:41 CEST] <skokkk> ac_slater: so change -ac to -an?
[19:09:51 CEST] <ac_slater> -an is "remove audio"
[19:10:46 CEST] <skokkk> could you please re-write the command, it's confusing to me
[19:11:23 CEST] <ac_slater> nope
[19:11:33 CEST] <ac_slater> play with it outside of your "web" context
[19:11:45 CEST] <ac_slater> the wiki page I linked you to should read from top-to-bottom in terms of complexity
[19:11:49 CEST] <ac_slater> start at the top
[19:12:06 CEST] <ac_slater> sorry :( I'm at work and can't do this right now
[19:14:59 CEST] <ac_slater> kepstin: sadly I think it's going to be more complicated than you're av_rescale_q item
[19:15:05 CEST] <ac_slater> s/item/idea
[19:17:49 CEST] <skokkk> ac_slater: what part is the resolution
[19:18:18 CEST] <ac_slater> skokkk: -s
[19:18:26 CEST] <ac_slater> skokkk: read the manual man, cmon
[19:29:02 CEST] <ac_slater> kepstin: http://paste.debian.net/843262/ ... the output of the rescale seems to be monotonically increasing. Shouldn't it stabilize?
[19:31:38 CEST] <kepstin> ac_slater: that looks correct, it's simply converting the pts from timebase units to milliseconds. If you want the difference between frames, you have to subtract the old from new pts beforehand
[19:34:00 CEST] <ac_slater> oooo
[19:37:47 CEST] <ac_slater> kepstin: doing that, I get negative values. But I think the abs() of those is correct
[19:38:23 CEST] <kepstin> ac_slater: well, your frames are in dts order, so if you are just using the relative values as-is, it'll be all over the place
[19:38:28 CEST] <ac_slater> oh right
[19:38:36 CEST] <ac_slater> I just realized that souce has b-frames
[19:38:54 CEST] <ac_slater> kepstin: it's exactly right for my non-b-frame video!
[19:38:56 CEST] <ac_slater> thanks!!!
[20:03:14 CEST] <Mavrik> Hmm... so how does HE-AACv2 64kb compare to 192kb MP2? :)
[20:10:32 CEST] <durandal_1707> compare it?
[20:12:08 CEST] <Mavrik> I will, had in my head there was a graph of SnR vs. Bitrate somewhere and can't find it on Google. Hence my question here - hoping someone will save me some tedious time doing comparisons.
[20:29:28 CEST] <DHE> SNR is a mathematical measurement of the difference. hearing tests are what people actually give a damn about
[20:57:16 CEST] <SchrodingersScat> I'm trying to match video frames from two different videos, it spits out a dts and pts but what does this actually mean?
[20:58:24 CEST] <SchrodingersScat> is there a way to translate this into a time position?
[21:00:41 CEST] <kepstin> SchrodingersScat: dts is an implementation detail of b-frames in codecs. pts is the 'presentation time stamp', aka time frame is shown. Multiply pts by the time_base to get seconds.
[21:02:52 CEST] <`Orum> I'm trying to encode with h264_nvenc, and I can figure out almost everything, but how do I set the qp to use with cqp ratecontrol?
[21:06:05 CEST] <SchrodingersScat> kepstin: awesome!
[21:17:32 CEST] <SchrodingersScat> kepstin: wow, k, that does get pretty close, nice.
[21:27:53 CEST] <`Orum> oh, I see, -q:v and then it looks like it should have a value 0 <= x <= 1
[23:16:22 CEST] <Anova> hi friends. I'm managed to use ffmpeg to download a HLS stream in m3u8 format. However, it seems very slow (speed = 1.5) vs other download tools like chrome extension fvd for example. What is the reason for the speed discrepancy and is there a way I can speed it up to use the full bandwidth of my internet connection?
[23:23:06 CEST] <DHE> live feed?
[23:23:24 CEST] <DHE> ffmpeg will do transcoding if you don't specify -c copy
[23:23:31 CEST] <DHE> two different possibilities there
[23:24:48 CEST] <pomaranc> or other download tools download files in parallel and ffmpeg doesn't
[23:26:04 CEST] <DHE> possible, but you'd need a pretty good connection for that to be the issue
[23:27:43 CEST] <Anova> I am including -c copy
[23:28:13 CEST] <Anova> ffmpeg -i "URL" -c copy -bsf:a aac_adtstoasc "C:\Output.mp4"
[23:28:26 CEST] <Anova> ^^ this is the command I'm using. It is not for a livestream, it is for a recorded video
[23:42:11 CEST] <TwinTailed> I am using LibAV (FFMPEG's) to remux and transcode an mp4 file. But the generated H264 MP4 (AVC/ACC) only plays sound on Windows Media Player and QuickTime Player. It plays normally on VLC and Classic Media Player.
[00:00:00 CEST] --- Thu Sep 29 2016