[Ffmpeg-devel-irc] ffmpeg.log.20150709

burek burek021 at gmail.com
Fri Jul 10 02:05:01 CEST 2015

[00:05:33 CEST] <djems54>    it works! thank you!
[00:06:03 CEST] <capostrike93> Hey
[00:06:14 CEST] <capostrike93> someone please can help me with ffmpeg + vp9 + dash
[00:06:22 CEST] <djems54> ffmpeg command in one, is it possible to put more quality in the stream .m3u8 file?
[00:06:25 CEST] <capostrike93> i posted a topic here http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2218
[00:07:04 CEST] <OstlerDev> djems54 yes, just change the -ab and -b:v
[00:07:13 CEST] <OstlerDev> that is audio bitrate and video bitrate I believe
[00:07:50 CEST] <capostrike93> My problem is Could not find codec parameters for stream 0 (Video: vp9, none, 1280x720): unspecified pixel format
[00:08:15 CEST] <capostrike93> i tried the pix_fmt without success
[00:08:26 CEST] <capostrike93> also  'analyzeduration' and 'probesize' without success
[00:08:27 CEST] <capostrike93> :(
[00:08:33 CEST] <OstlerDev> not sure :/
[00:08:35 CEST] <OstlerDev> sorry :/
[00:08:39 CEST] <capostrike93> :/
[00:08:42 CEST] <capostrike93> :'(
[00:08:58 CEST] <djems54> yes, but I do not put multiple quality in the same file m3u8 sortiee (360p, 720p, 1080p ...)
[00:11:03 CEST] <djems54> I read that with HLS can provide several quality stream, and they are automatically selected based on the quality of the network
[00:11:39 CEST] <DHE> that is true, but my own testing shows that players tend to be pretty bad at switching if the network speed changes dramatically and suddenly
[00:12:21 CEST] <capostrike93> you can edit some flash player like flashls.org
[00:12:43 CEST] <capostrike93> make a better implementation of the bitrate switching
[00:13:41 CEST] <capostrike93> i did a penalty code, when reaching low speed, more penalty = avoid switching up untril some time (network stable for some time based on penalty..)
[00:15:23 CEST] <djems54> ok, thanks
[00:15:47 CEST] <capostrike93> https://github.com/mangui/flashls/blob/dev/src/org/mangui/hls/controller/LevelController.as
[00:16:00 CEST] <capostrike93> the getnextlevel function
[00:16:05 CEST] <capostrike93> is what you need improve
[00:16:06 CEST] <capostrike93> ;)
[00:16:47 CEST] <capostrike93> about the setup of the m3u8 is easy
[00:16:50 CEST] <djems54> interesting :)
[00:17:06 CEST] <capostrike93> you need to have example 720p.m3u8 480p.m3u8 etc..
[00:17:26 CEST] <capostrike93> and a playlist.m3u8 that points to the bitrates playlists
[00:18:32 CEST] <capostrike93> #EXTM3U #EXT-X-VERSION:3 #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1396000,RESOLUTION=848x480 616-chunklist_b1396000.m3u8 #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=896000,RESOLUTION=640x360 616-chunklist_b896000.m3u8 #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=596000,RESOLUTION=427x240 616-chunklist_b596000.m3u8 #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=446000,RESOLUTION=320x160 616-chunklist_b446000.m3u8
[00:18:53 CEST] <capostrike93> http://pastebin.com/EyKB7Hi0
[00:19:01 CEST] <djems54> oh! great idea!
[00:19:23 CEST] <capostrike93> its simple
[00:19:34 CEST] <capostrike93> you use ffmpeg to create each m3u8 playlist quality
[00:19:49 CEST] <djems54> yes it looks
[00:19:51 CEST] <capostrike93> and later you create the playlist with the bitrates playlist
[00:20:09 CEST] <capostrike93> but like ostler say, players sucks a bit in bitrate switching
[00:20:18 CEST] <djems54> Ok
[00:20:20 CEST] <capostrike93> you need improve your player, so pick someone open source
[00:21:00 CEST] <capostrike93> if u want more help
[00:21:02 CEST] <capostrike93> send me a mail
[00:21:07 CEST] <capostrike93> to caposguanatosfut at gmail.com
[00:21:30 CEST] <capostrike93> ok, now, someone please HEEEELP ME
[00:21:38 CEST] <capostrike93> http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2218
[00:22:14 CEST] <capostrike93> TRYING to create a DASH vod with VP9, http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2218
[00:22:32 CEST] <capostrike93> TRYING to create a DASH vod with VP9, unspecified pixel format
[00:22:41 CEST] <djems54>    a big thank you! I note
[00:23:02 CEST] <djems54> and sorry to not be able to help you in return
[00:23:20 CEST] <capostrike93> no problem
[00:23:51 CEST] <Nolski> capostrike93: I think Mozilla is working on adaptive bitrate streaming in native browser
[00:24:23 CEST] <capostrike93> what you mean? for hls?
[00:24:38 CEST] <capostrike93> hls is not supported with browsers (only on ios)
[00:24:42 CEST] <Nolski> not for hls
[00:24:51 CEST] <capostrike93> oh, for dash
[00:24:55 CEST] <capostrike93> ?
[00:25:22 CEST] <capostrike93> well i think the bitrate switching is done via javascript in the current players
[00:25:35 CEST] <Nolski> capostrike93: yeah for DASH
[00:25:51 CEST] <Nolski> using mediaSource and stuff like that
[00:26:07 CEST] Action: Nolski only heard bits and pieces about it
[00:26:36 CEST] <Nolski> but it's still pretty experimental I think
[00:26:36 CEST] <capostrike93> its all handled via player code
[00:26:52 CEST] <capostrike93> the browser only play what the player sends to him via mediasource api
[00:27:06 CEST] <Nolski> Right.
[00:27:07 CEST] <capostrike93> but the player pick the quality to use
[00:27:30 CEST] <capostrike93> some time ago i see the bitrate switcher of dashif player
[00:27:55 CEST] <Nolski> I think the issue they're trying to solve is to do that without using a flash player
[00:28:04 CEST] <capostrike93> the problem is now, i want to use dash in production
[00:28:14 CEST] <capostrike93> but i cant even generate the fucking manifest.mpd
[00:28:16 CEST] <Nolski> also I think this was the guy working on it: https://air.mozilla.org/tech-talk-3/
[00:28:17 CEST] <Nolski> Oh.
[00:28:26 CEST] <capostrike93> because ffmpeg outputs a f***king error
[00:28:27 CEST] <capostrike93> :(
[00:28:43 CEST] <Trieste> hi, just a quick check - if, when using the C ffmpeg API, I decode an AVPacket using avcodec_decode_video2, and the AVPacket's flags are set to PKT_FLAG_KEY, does that mean I just decoded an I frame, and if they're not, did I just get a P/B frame?
[00:28:44 CEST] <Nolski> I don't think browser support is there yet
[00:29:05 CEST] <Nolski> capostrike93: ^
[00:30:06 CEST] <capostrike93> already here, with chrome
[00:30:16 CEST] <capostrike93> youtube is using dash + vp9
[00:30:20 CEST] <capostrike93> since some months ago
[00:30:27 CEST] <Nolski> only in chrome though
[00:30:40 CEST] <capostrike93> and dash + vp8 since maybe 2 years
[00:30:44 CEST] <capostrike93> or 1 year
[00:30:49 CEST] <capostrike93> yeah, but for me
[00:30:51 CEST] <capostrike93> chrome is like
[00:30:59 CEST] <capostrike93> 70% of customers
[00:31:28 CEST] <capostrike93> firefox is supporting vp9 too i think
[00:31:31 CEST] <capostrike93> and also dash
[00:31:47 CEST] <capostrike93> so i can use dash for 80-90%
[00:31:59 CEST] <capostrike93> and for the 15% hls/rtsp
[00:32:16 CEST] <capostrike93> but i need this first http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2218
[00:32:16 CEST] <Nolski> huh you may be right
[00:32:20 CEST] <capostrike93> :(
[00:33:49 CEST] <capostrike93> where ffmpeg devs are
[00:35:50 CEST] <Nolski> is there #ffmpeg-dev?
[00:36:01 CEST] <capostrike93> is devel
[00:36:13 CEST] <capostrike93> but the website says is only for talk about development
[00:36:14 CEST] <capostrike93> so..
[00:36:19 CEST] <capostrike93> anyways i am going to try
[00:36:25 CEST] <Nolski> glhf ;)
[00:39:02 CEST] <capostrike93> well, i typed in the ffmpeg-devel room
[00:39:08 CEST] <capostrike93> noone online i think
[00:39:11 CEST] <capostrike93> :(
[00:42:01 CEST] <OstlerDev> Is there a way to tell the segmenter where to start?
[00:59:21 CEST] <OstlerDev> I am currently using the segemter to convert a file for a livestream, however it cuts off before I want it to. Is there a way to tell ffmpeg to either never stop, or to continue a conversion at a certain point without destroying the old files?
[01:15:52 CEST] <MetaPhaze> ffmpeg -f alsa -acodec pcm_s16le -ac 2 -i pulse -f x11grab -r 30 -s 1920x1080 -i :0.0+0,0 -vcodec libx264 -preset ultrafast -crf 0 ScreenCast-Output.mkv
[01:16:03 CEST] <MetaPhaze> Capture area 1920x1080 at position 0.0 outside the screen size 3840x1080
[01:16:05 CEST] <MetaPhaze> :0.0+0,0: Invalid argument
[01:16:33 CEST] <klaxa> also use -framerate 30 instead of -r 30
[01:16:52 CEST] <MetaPhaze> i'll just say it
[01:16:54 CEST] <MetaPhaze> you got a bug
[01:17:27 CEST] <klaxa> no bug without a command + output paste
[01:17:31 CEST] <MetaPhaze> ffmpeg reports that 1920x1080 is outside screen size 3840x1080, but 1920x1079 works fine
[01:17:40 CEST] <MetaPhaze> bug exists regardless
[01:17:47 CEST] <MetaPhaze> you want red tape, go elsewhere
[01:17:50 CEST] <MetaPhaze> report done
[01:18:53 CEST] <Chuckles32> I'm trying to understand the documentation for creating a video from stills.  Using: ffmpeg -framerate 1/5 -i img%03d.png -c:v libx264 -r 30 -pix_fmt yuv420p out.mp4  Can someone explain to me what the -framerate 1/5 means?  I understand this is the input framrate and the -r 30 is the output framerate.  Documentation says each image will ahve druation of 5 seconds (inverse of 1/5 framse per
[01:18:53 CEST] <Chuckles32> second).  However, when i change this to 1/15, i don't get a 15 seconds for each still?  What am i missing?
[01:19:03 CEST] <durandal_1707> no proof
[01:20:42 CEST] <durandal_1707> 1/15 is number of frames per single second
[01:24:10 CEST] <durandal_1707> MetaPhaze: what version?
[01:24:44 CEST] <MetaPhaze> ffmpeg version 2.6.3 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8.4 (Gentoo 4.8.4 p1.6, pie-0.6.1)
[01:25:16 CEST] <MetaPhaze> 64 bit
[01:26:27 CEST] <Chuckles32> I'm having a hardtime understanding this.  I will be creating a 3 video panel.  Left 1/3 is 15 second video.  Middle 1/3 is 15 second video.  Right 1/3 will be a video made up of stills.  Depending on the process, I will have anywhere from 2-16 stills to merge into a video.  I need to make sure that the video created from the stills is 15 sec duration and that each still is displayed at least
[01:26:27 CEST] <Chuckles32> once.  If i had 2 stills, i'd create a video that displayed each still for 7.5 seconds.  If i have 15 stills, the video would display each still for 1 second.  I think -framerate is what i'm looking for, but can't figure out the math.
[01:26:34 CEST] <MetaPhaze> sorry for being blunt but i'm in the middle of a 300,000 line coding project and this is just slowing me down recording things for my partners, all over 1 line of pixels
[01:28:14 CEST] <debianuser> MetaPhaze: "-i :0.0+0,0" ? Shouldn't that be "-i :0.0+0+0"?
[01:29:02 CEST] <debianuser> or maybe just ":0"
[01:29:23 CEST] <durandal_1707> MetaPhaze: maybe that's what x11/wm reports
[01:31:07 CEST] <durandal_1707> Chuckles32: try with -vf fps filter instead of -r
[01:31:14 CEST] <MetaPhaze> debianuser, i've tried both ways, according to man ffmpeg it's now +0,0 and not +0+0 like it used to be
[01:32:24 CEST] <MetaPhaze> durandal_1707, it could be that openbox is reporting 1079, but why then does ffmpeg say that it's 3840x1080, it's getting that screen size from somewhere
[01:33:50 CEST] <klaxa> MetaPhaze: you could speed up the process by pasting your command + output, maybe also with -loglevel debug
[01:33:57 CEST] <durandal_1707> You use ffmpeg directly?
[01:37:20 CEST] <MetaPhaze> http://pastebin.com/htkpKgpW
[01:37:25 CEST] <MetaPhaze> yeah directly
[01:37:31 CEST] <MetaPhaze> scripted and directly
[01:37:39 CEST] <MetaPhaze> command highlighted for ya there
[01:41:42 CEST] <Chuckles32> I'm doing a poor job explaining.  I don't understand what values to use.  Let's start with this.  I have 16 jpgs in the folder from which i am looking to build a 15 second video.  Each jpg should be displayed in the video for the same duration.  i set framerate = 1/5 i get a video duration of 00:01:15.03.  Here is pastebin:  http://pastebin.com/N7CijgZi
[01:43:46 CEST] <klaxa> MetaPhaze: i cannot reproduce your problem, is your desktop resolution 1080p? i'll switch to that and try again
[01:44:15 CEST] <Chuckles32> if i want a 15 total duration of the resulting video, what do i use for framerate - assuming 16 jpgs will be in the video?
[01:45:15 CEST] <klaxa> MetaPhaze: i cannot reproduce even at 1080p, can you try a more recent version of ffmpeg, best would be latest git
[01:45:50 CEST] <Chuckles32> For 15 second video, the duration of EACH of the 16 jpgs would .9375 seconds.  How do i translate this to framerate?
[01:46:20 CEST] <durandal_1707> Chuckles32: 15/16
[01:46:43 CEST] <Chuckles32> that's what i thought.  .9375, right?
[01:47:22 CEST] <durandal_1707> yes
[01:49:06 CEST] <Chuckles32> OK.  Take a look a this.  It comes up with a 00:00:16.03 duration.   http://pastebin.com/JJALWEu2
[01:55:11 CEST] <Chuckles32> Where does the extra second come from?
[01:57:08 CEST] <durandal_1707> rounding of some kind try vfs filter
[02:01:06 CEST] <durandal_1707> setfps filter
[02:04:02 CEST] <Chuckles32> ok.  thanks.  i'll take a look.
[02:07:57 CEST] <Chuckles32> durandal_1707:  No change - http://pastebin.com/nJL3e39p
[02:09:08 CEST] <Chuckles32> do i need to change the output framerate?  Is there some formula between input framerate and output framerate that i'm missing?
[02:14:23 CEST] <DHE> Do you think I could do multiple transcodes of the same video at different resolutions/bitrates from the same first pass encoding? (libx264)
[02:14:39 CEST] <DHE> ie. do pass 1, then do pass 2, then do pass 2 at a different resolution, perhaps again and more
[02:17:42 CEST] <MetaPhaze> klaxa, yeah i'll try it
[02:18:02 CEST] <MetaPhaze> klaxa, using 1080p x 2
[02:18:07 CEST] <MetaPhaze> dual monitor
[02:31:06 CEST] <Chuckles32> drandal_1707:  Any other suggestions?
[02:31:49 CEST] Action: Chuckles32 slaps ChazNunz around a bit with a large trout
[02:32:02 CEST] <klaxa> DHE: why do you want to encode 2-pass? do you have to meet a specific filesize?
[02:34:29 CEST] <klaxa> DHE: https://trac.ffmpeg.org/wiki/Encode/H.264
[02:35:27 CEST] <OstlerDev> Is there a way to pause an ongoing ffmpeg encode?
[02:35:41 CEST] <dystopia_> ctrl+x
[02:35:42 CEST] <dystopia_> iirc
[02:35:48 CEST] <klaxa> on linux you can try ctrl+z
[02:35:51 CEST] <OstlerDev> its ctrl+z
[02:35:57 CEST] <OstlerDev> but I am meaning prgramatically
[02:36:51 CEST] <dystopia_> can you explain a bit more what you mean / what your trying to do
[02:37:27 CEST] <klaxa> send SIGTSTP to the process?
[02:37:34 CEST] <klaxa> that's what ctrl+z does
[02:37:44 CEST] <klaxa> and if you want to continue send SIGCONT
[02:37:57 CEST] <klaxa> that's what i gather from wikipedia
[02:38:03 CEST] <OstlerDev> well I am trying to convert a file while it is downloading, so I need to pause the encode before it gets to the end of the written bytes
[02:38:05 CEST] <klaxa> you can send signals to processes with the kill command
[02:38:12 CEST] <dystopia_> no you don't
[02:38:16 CEST] <dystopia_> sec OstlerDev
[02:38:19 CEST] <OstlerDev> Ok
[02:38:22 CEST] <dystopia_> let me find the line
[02:38:25 CEST] <OstlerDev> k
[02:38:35 CEST] <dystopia_> you can limit encoding fps, to video fps
[02:38:43 CEST] <klaxa> you can use the -re flag
[02:38:43 CEST] <dystopia_> so the encode never catechs up to the end of the file
[02:38:46 CEST] <dystopia_> thats it
[02:38:54 CEST] <dystopia_> -re
[02:38:56 CEST] <klaxa> but that will not guarantee that you do not seek beyond the end of the file
[02:39:06 CEST] <dystopia_> it should
[02:39:08 CEST] <klaxa> suppose you are downloading a file slower than its playback speed
[02:39:15 CEST] <klaxa> you'll seek beyond the end of the file
[02:39:24 CEST] <dystopia_> that is unlikely in 2015 heh
[02:39:34 CEST] <klaxa> that really depends on the file
[02:39:35 CEST] <OstlerDev> so I could do like -re 30
[02:39:39 CEST] <OstlerDev> to cap it to 30fps?
[02:39:49 CEST] <klaxa> no -re is a parameterless flag
[02:39:58 CEST] <OstlerDev> Ok
[02:40:05 CEST] <klaxa> it reads packets from the input at the same speed it would be played back
[02:40:35 CEST] <OstlerDev> so my overall command would be ffmpeg -i in.mkv out.mp4 -re ?
[02:40:43 CEST] <klaxa> ffmepg -re -i in.mkv out.mp4
[02:40:47 CEST] <OstlerDev> Ok
[02:40:49 CEST] <dystopia_> yeah before input
[02:41:19 CEST] <klaxa> if you are fine with taking as long to encode as it takes to watch the file and are confident that your download speed is fast enough that should be okay
[02:41:21 CEST] <dystopia_> works well for tv encodes
[02:41:35 CEST] <OstlerDev> Awesome
[02:41:50 CEST] <klaxa> be aware that you are in no place to complain if your download speed falls and the encode breaks though ;)
[02:41:56 CEST] <OstlerDev> The only worry that I would have when encoding would be if the user attempts to fast forward the video
[02:42:11 CEST] <OstlerDev> My application livestreams torrents to an Apple TV
[02:42:25 CEST] <OstlerDev> so I must make sure that they are in .mp4 format
[02:42:40 CEST] <OstlerDev> And yeah I understand
[02:42:42 CEST] <OstlerDev> Thanks a ton though
[05:11:20 CEST] <OstlerDev> When using the sequence creator is there a way to make sure that "#EXT-X-MEDIA-SEQUENCE:1" is inside the .m3u8 file?
[05:18:14 CEST] <OstlerDev> nbm
[05:18:16 CEST] <OstlerDev> *nvm
[05:23:21 CEST] <OstlerDev> Is there a way to include "#EXT-X-PLAYLIST-TYPE:EVENT" in the header of the .m3u8 file while using the segmenter?
[06:29:07 CEST] <FlorianBd>  --enable-random ??? hahaha
[08:14:08 CEST] <hendry> i have two jpegs i want to loop between a second apart in a webm. -loop 1 seems to encode forever. Am I missing something?
[08:52:47 CEST] <durandal_1707> hendry: -d ?
[09:08:15 CEST] <hendry> thinking i was missing the duration and the fact you need to loop it in the player itself
[12:53:48 CEST] <Lokie> hey I was using a quite old static build of ffmpeg to transcode some videos to webm for universal browser playback.
[12:53:53 CEST] <Lokie> Since I upgraded to a almost latest build, the webms can't be playbacked on a browser. According to mediainfo the only difference is that the new has format version 2 while the old one is format version 4 / 2
[12:54:28 CEST] <Lokie> any way to force ffmpeg to use that kind of format to verify if that is the problem?
[12:58:29 CEST] <relaxed> Lokie: pastebin the output of "ffmpeg -i blah.webm" on one of these files that won't play
[12:59:40 CEST] <Lokie> dled it to my machine
[12:59:47 CEST] <Lokie> I think the syntax has changed
[12:59:55 CEST] <Lokie> gonna pastebin my command first
[13:00:17 CEST] <Lokie> https://p.lokie.eu/eUZzAk5td6nB.txt
[13:00:47 CEST] <Lokie> video has 9 seconds at start with no video / audio
[13:02:08 CEST] <Lokie> ffmpeg -i https://p.lokie.eu/slKDI8IxMKfo.txt
[13:03:02 CEST] <Lokie> on the older version I could put -ss before -i to make it faster if there weren't any subtitles to be added
[13:04:16 CEST] <Lokie> I don't understand what part of my syntax makes it start at 8.737 since it used to work fine
[13:05:54 CEST] <relaxed> do you have a url to one that doesn't play?
[13:05:59 CEST] <relaxed> the output looks fine
[13:06:11 CEST] <Lokie> you mind if I pm the link?
[13:06:20 CEST] <relaxed> sure
[13:06:59 CEST] <relaxed> plays fine with chrome
[13:07:14 CEST] <Lokie> no empty content at start?
[13:07:53 CEST] <Lokie> even mpc shows the first 9sec blank
[13:07:55 CEST] <relaxed> oh, yes there is
[13:08:38 CEST] <dystopia_> -ss is to set the start time, just omit it
[13:08:53 CEST] <relaxed> it's probably because of the included subs, yes?
[13:09:14 CEST] <Lokie> yea but I need to define the starting point dystopia_ I want to ignore the first 36 seconds
[13:09:22 CEST] <relaxed> Lokie: remove the subs and see if it still happens
[13:09:38 CEST] <Lokie> I don't specify anything about subs in my command
[13:10:10 CEST] <relaxed> there are subs in that webm, remove them
[13:10:17 CEST] <Lokie> hardsubs
[13:10:26 CEST] <Lokie> so already burned in the video stream
[13:10:47 CEST] <relaxed> I see Stream #0:2(eng): Subtitle: webvtt (default)
[13:10:55 CEST] <dystopia_> use -map to select just audio and video
[13:11:12 CEST] <relaxed> so you have hardsubbed and soft
[13:11:17 CEST] <Lokie> that is true
[13:11:22 CEST] <Lokie> is there a flag to ignore subs?
[13:11:47 CEST] <relaxed> -sn
[13:11:49 CEST] <Lokie> similar to -vn
[13:11:55 CEST] <Lokie> make sense :P
[13:12:02 CEST] <Lokie> makes *
[13:12:38 CEST] <Lokie> yea that was the problem
[13:12:53 CEST] <Lokie> on the old version I was using if you didn't specify the substitles flag
[13:12:57 CEST] <Lokie> none was added
[13:13:12 CEST] <Lokie> in the current some def or something is added so I need to include the -sn
[13:13:13 CEST] <Lokie> cheers
[13:14:20 CEST] <Lokie> gonna check -map dystopia_
[13:15:04 CEST] <Lokie> also anyone knows x264 flags / profiles that makes it playable on almost any device / browser? (that is why I am using webm cause it seems pretty much anything can play it)
[13:23:42 CEST] <JEEBsv> vp8/9 have very limited hw decoding capabilities
[13:23:52 CEST] <JEEBsv> so you are pretty much throwing out all mobile devices
[13:24:18 CEST] <JEEBsv> if you are limited to firefox/chrome and PC, then vp8 or vp9 might be ok
[13:25:11 CEST] <Lokie> my old 5yr mobile can play them back
[13:25:18 CEST] <Lokie> same for my 2yr chinese tablet :P
[13:26:12 CEST] <JEEBsv> then you are doing something very low resolution and rate, and the decoding is being done in software
[13:26:20 CEST] <JEEBsv> it will eat battery but possibly work, yes
[13:28:09 CEST] <Lokie> it's 7209 @ ~ 2.5Mbps
[13:28:58 CEST] <Lokie> If you could point me to a link for some good h264 profile / settings that works better I would gladly play with it and try to switch to mp4
[13:29:12 CEST] <Lokie> 7209 = 720p
[13:29:16 CEST] <JEEBsv> what kind of stuff do you intend to support is the question
[13:29:25 CEST] <Lokie> as much as possible
[13:29:52 CEST] <JEEBsv> as much as possible would be -profile:v baseline -level:v 30 (constrained baseline; level 3.0)
[13:30:23 CEST] <Lokie> realistically 95% is pcs or modern mobile devices that have no problem playing them but .. :P
[13:30:25 CEST] <JEEBsv> then a bit less than that but gaining compression is -profile:v main -level:v 40 (main; level 4.0)
[13:30:51 CEST] <Lokie> cheers will try to play with them
[13:30:57 CEST] <Lokie> the () are comments right?
[13:31:15 CEST] <JEEBsv> and then for most of modern devices except for weird china stuff is (leave profile to whatever) -level 41 for up to 1080p and 42 for >1080p
[13:31:35 CEST] <JEEBsv> of course, remember to use -maxrate and -bufsize when encoding so it can enter actual buffering model
[13:31:43 CEST] <JEEBsv> (but you should have been doing this with libvpx already)
[13:32:06 CEST] <Lokie> I used -crf and c:b
[13:32:16 CEST] <Lokie> eh -b:v *
[13:35:33 CEST] <JEEBsv> -b:v is generally not recommended unless you are doing 2pass
[13:35:49 CEST] <JEEBsv> and you in any case need to do -maxrate + -bufsize if you are going to do things over a limited bandwidth
[13:36:12 CEST] <JEEBsv> so either 2pass + bit rate + VBV (maxrate+bufsize), or CRF + VBV
[13:37:05 CEST] <Lokie> mhm will try to read upon them a bit more
[13:37:38 CEST] <Lokie> gonna part to keep stuff handy on the logs and will come back once I had time to try and play with the options
[13:37:49 CEST] <Lokie> cheers for the help!
[17:46:40 CEST] <livingBEEF> Can I somehow use one stream in -filter_complex as input for multiple streams? When I try it by using the label multiple times, it doesn't work....
[17:46:59 CEST] <c_14> try split?
[17:49:29 CEST] <livingBEEF> oh, that should work. ty
[18:20:41 CEST] <chutchut> hey all! X)
[18:23:02 CEST] <chutchut> getting the followiing error with the following cmd line, can anyone see an obvious issue? Im thinking it may be bitrate related.. "Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height" cmd: "nohup ffmpeg -analyzeduration 2147483647 -probesize 2147483647 -acodec libfdk_aac -vcodec h264 -i rtsp://witness:tHNItkNfjwImrerlQGjVSP6hMnnqgqgv@ -acodec libfdk_aac -vcodec libx26
[18:23:02 CEST] <chutchut> 4 -ar 128000 -b:a 128k -b:v 250k -bufsize 350k -pix_fmt yuv420p -r 15 -s 640x480 /var/www/.../231721598_2015-07-09_17-15-32.mp4"
[18:36:01 CEST] <chutchut> thx n sorry!
[18:36:15 CEST] <oomkiller> if I use -vcodec to change one thing at the video codec (i.e. -r 25 to reduce the fps) does ffmpeg copy  the other settings (quality  and so on) from the file given, or does it use defaults?
[18:38:08 CEST] <oomkiller> of course I select the same video codec as the source file
[18:39:26 CEST] <chutchut> actually loooking again at the output ive just spotted this Nolski: [libfdk_aac @ 0x199c060] Specified sample rate 128000 is not supported
[18:39:42 CEST] <chutchut> how can I find out what supported rates are?
[18:40:46 CEST] <Nolski> chutchut: I think it depends on your encoding (not sure though)
[18:41:33 CEST] <chutchut> full output Nolski: http://pastie.org/10282323
[18:42:50 CEST] <Nolski> Do you have h264 encoder on your box?
[18:42:55 CEST] <chutchut> yea
[18:43:19 CEST] <chutchut> been using it for a while for low quality now just want to up the output a bit
[18:43:32 CEST] <Nolski> I see
[18:44:21 CEST] <Nolski> hmm, I'm not quite sure. Maybe one of the more experienced folk in here could help you. I'm still a bit new to video processing myself.
[18:44:38 CEST] <chutchut> same, thanks anyway X)
[18:47:05 CEST] <c_14> oomkiller: defaults
[18:47:59 CEST] <c_14> chutchut: line 29
[18:48:22 CEST] <chutchut> thx c_14 got it ^^!
[18:48:33 CEST] <chutchut> just wondering what the supported rates were..
[18:48:47 CEST] <c_14> probably up to 96k or so
[18:49:05 CEST] <chutchut> ok was using 96 before for the other qual so ill stick with that cheers
[18:49:33 CEST] <chutchut> no sorry scratch that, 160000 worked before
[18:49:42 CEST] <chutchut> for -ar
[18:50:02 CEST] <chutchut> oops ar=16000
[18:50:27 CEST] <Mavrik> Are you sure you want 160kHz samplerate? That you're not trying to set bitrate?
[18:50:55 CEST] <chutchut> im not sure to be honest, whats a sensible sample rate?
[18:51:22 CEST] <c_14> 44.1k 48k or 96k (usually)
[18:51:38 CEST] <Mavrik> chutchut, CDs are 44100, standard is 48000
[18:51:47 CEST] <chutchut> cool thx
[18:51:50 CEST] <Mavrik> 96k is high quality studio equiment only
[18:52:09 CEST] <chutchut> coolio may as well use 48000 then
[18:52:09 CEST] <Mavrik> and these are supported rates for fdk_aac: https://www.ffmpeg.org/doxygen/2.2/libfdk-aacenc_8c.html#a0167df555be7476084b2968f0a5884a6
[18:52:17 CEST] <chutchut> thanks! X)
[18:52:49 CEST] <c_14> >0
[18:53:07 CEST] <c_14> Is that for something special?
[18:53:35 CEST] <chutchut> huh?
[18:54:07 CEST] <c_14> I was just wondering why '0' is a supported samplerate.
[19:29:27 CEST] <kubast2> Hey I've a song[m4a aac 256 kbps] ,and I would like to convert it to one of the following formats wma/ogg/mp3[A game supports m4a-it's described as "iTunes M4A"- ,but requires apple quicktime player].
[19:30:27 CEST] <kubast2> How can I accomplish that?[to 256 ogg(vorbis ,propablly)/320 stereo mp3] ?
[19:31:06 CEST] <kubast2> My internet is very slow right now ,and I can't do a proper research on google
[19:35:57 CEST] <kuba__> any ideas?
[19:36:33 CEST] <c_14> ffmpeg -i file -c:a libvorbis -b:a 256k out.ogg
[19:53:13 CEST] <kuba__> thx :)
[20:10:10 CEST] <livingBEEF> so split apparently can't handle more than 10 copies... is that a feature of a bug?
[20:11:12 CEST] <livingBEEF> cuz when I did split=3 and then 3 times split=10, it worked
[20:26:29 CEST] <durandal_1707> livingBEEF: it can handle more than 10
[20:47:25 CEST] <livingBEEF> durandal_1707: that's weird... because both, split and asplit said No output pad can be associated to link label 'vi11' (resp. 'ai11') and when I did a split to 3 and then 3 times to 10, it worked without problems. And it was literally the only change I made.
[20:48:55 CEST] <durandal_1707> your command is missing something...
[21:54:53 CEST] <Kaedenn> How do I determine in a script (python script, but it could just as well be sh) that the user is running the ffmpeg release of ffmpeg, and not the libav fork?
[21:55:21 CEST] <Kaedenn> I'm thinking of processing the output of the version argument
[21:56:39 CEST] <Kaedenn> I guess I could do ffmpeg --version | head -1 | grep -i ' libav ', and if that's true then I error out
[22:01:09 CEST] <JEEBsv> yes, any binary released from the libav sources bears the name there
[22:02:09 CEST] <JEEBsv> basically you should just poke the user towards avconv in that case, since that is the updated thing (I really have no idea why they left the ffmpeg binary there for 0.8 - it has since stirred more derp than given gain)
[22:03:20 CEST] <JEEBsv> of course that probably made more sense in like 2011, when people had no idea of the new syntax...
[22:03:41 CEST] <Kaedenn> avconv doesn't work with my project
[22:03:50 CEST] <Kaedenn> There are visual artifacts and blurring that make the output horrendous
[22:03:58 CEST] <Kaedenn> So I'm just saying screw it, use the real ffmpeg
[22:04:05 CEST] <JEEBsv> that just probably means it's ancient
[22:04:09 CEST] <Kaedenn> !deprecated
[22:04:14 CEST] <Kaedenn> Where's that URL?
[22:04:25 CEST] <Kaedenn> That explains the different versions of ffmpeg
[22:04:36 CEST] <JEEBsv> which packaged versions tend to be, and 0.8 was there in debian based shit for a frightening amount of time
[22:04:52 CEST] <JEEBsv> and when people grab or build ffmpeg or libav they generally get the latest
[22:05:03 CEST] <JEEBsv> (as in, from outside the official packages)
[22:05:10 CEST] <Kaedenn> Yes I know all of this
[22:05:40 CEST] <JEEBsv> just saying that it might as well not be a case of avconv not being able of it, but rather just that avconv being fuck old
[22:06:12 CEST] <JEEBsv> also we'll probably get some of that now that debian switched back to ffmpeg as default provider for the libs/tools
[22:06:54 CEST] <Kaedenn> stackoverflow.com/questions/9477115/w/9477756#9477756
[22:06:58 CEST] <Kaedenn> http://stackoverflow.com/questions/9477115/w/9477756#9477756
[22:07:19 CEST] <JEEBsv> I am involved in both ffmpeg and libav, so you really don't need to poke me with those links
[22:07:22 CEST] <JEEBsv> löl
[22:08:38 CEST] <JEEBsv> granted, the major changes in API etc are probably behind us in a lot of ways, and that was one of the reasons why it took a lot of time for debian to move to a newer libav version, so we might have less ancient versions in debian-based distros in the future
[22:08:58 CEST] <JEEBsv> but still, distro versions will always be old Quite Soon
[22:09:55 CEST] <JEEBsv> also debian was really bad at throwing away unmaintained shit, so you literally waited for some package to get updated for newer APIs for months
[22:12:02 CEST] <JEEBsv> but as I am veering off a tangent here, unless you are using one of those video filters or input or output formats that only ffmpeg supports (there are some, generally it's fringe stuff as far as the formats go), it's not avconv itself that is your issue but rather the age of the binaries.
[22:12:26 CEST] <JEEBsv> but of course you could have also found a bug on the libav side :)
[22:15:18 CEST] <JEEBsv> I guess if I was making a tool that was using ffmpeg or avconv I'd probably have version checks
[22:15:55 CEST] <JEEBsv> which would barf at the user if the version would be too old and warn if it was too new
[22:18:10 CEST] Action: c_14 probably wouldn't check for too new
[22:18:17 CEST] <c_14> Haven't been hit with any serious regressions though.
[22:18:58 CEST] <JEEBsv> which is why I didn't say "barf" at it
[22:19:04 CEST] <JEEBsv> just warning would be enough
[22:19:34 CEST] <c_14> Mhm, overread that.
[22:19:49 CEST] <c_14> *overlooked
[22:22:15 CEST] <JEEBsv> I think I should regrep my logs one of these days: should see what two things I herped a derp about
[22:22:36 CEST] <JEEBsv> one was trailing options, which ffmpeg can notice, but will do absolutely nothing about
[22:22:58 CEST] <JEEBsv> I'd rather prefer a "you're doing it wrong, boy" approach to such things and a nonzero exit code
[22:23:27 CEST] <JEEBsv> might throw a patch at that if i get the time :/
[23:05:47 CEST] <techtopia> when using libx265 are all the options / switches / commands ect... the same as when using libx264?
[23:06:25 CEST] <JEEBsv> no
[23:06:59 CEST] <JEEBsv> but you generally shouldn't have to care much aside from the basics like preset and CRF and possibly VBV (do note that I have no effing idea how good x265's VBV is)
[23:07:00 CEST] <techtopia> oh hey jeeb
[23:23:37 CEST] <techtopia> hmm
[23:23:55 CEST] <techtopia> im encoding at 8fps to x265
[23:24:09 CEST] <techtopia> and at x264 i encode this content at around 55fps
[23:24:27 CEST] <techtopia> i know there is a performence hit but thats more than it should be right?
[23:25:01 CEST] <techtopia> it's almost 7x slower
[23:29:17 CEST] <ILEoo> I think it's less that I would expect
[23:30:57 CEST] <klaxa> seven times slower is actually pretty fast for h265
[23:31:41 CEST] <klaxa> s/h265/x265/
[23:31:50 CEST] <klaxa> have you looked at benchmarks beforehand?
[23:32:16 CEST] <techtopia> no just doing some tests with a 1min ts sample
[23:58:02 CEST] <techtopia> http://i.imgur.com/aA1k1Ev.png
[23:58:17 CEST] <techtopia> my initial test heh looks like there are some massive savings to be had
[23:58:30 CEST] <techtopia> but the encoding speed is a hasstle
[23:58:35 CEST] <techtopia> guess i could run them overnight
[00:00:00 CEST] --- Fri Jul 10 2015

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