[Ffmpeg-devel-irc] ffmpeg.log.20160207

[burek]

[01:14:25 CET] <spidersgeorg> what are the parameters to adjust mpegts stream? I have specified anything yet the stream I'm seeing from using simply -f mpegts udp://127.0.0.1:1234 looks very low quality
[01:20:31 CET] <spidersgeorg> I'm looking for a lossless way to stream over localhost to another application
[01:21:44 CET] <tp_> -c copy
[01:21:45 CET] <J_Darnley> If you *really* want lossless then you'd better ask for it.
[01:24:47 CET] <spidersgeorg> @J_Darnley I'm afraid I don't understand
[01:25:11 CET] <spidersgeorg> @tp_ does -c copy require outputting to a file?
[01:26:45 CET] <spidersgeorg> I'm getting an error 'filtering and streamcopy cannot be used together'
[01:27:08 CET] <tp_> then dont use filtering at the same time if you want lossless :)
[01:27:39 CET] <spidersgeorg> makes sense haha
[01:27:55 CET] <spidersgeorg> basically I'm opening my capture device deinterlacing and trying to send it to another application
[01:29:06 CET] <spidersgeorg> is there a way to filter and stream without losing too much quality
[01:29:26 CET] <J_Darnley> Encode at high quality, obviously
[01:29:38 CET] <J_Darnley> Be more precise!
[01:29:55 CET] <J_Darnley> What format are you trying out output?  h264?
[01:30:12 CET] <spidersgeorg> this is what I have right now
[01:30:21 CET] <spidersgeorg> ffmpeg -f dshow -framerate 29.97 -i video="GV-USB2, Analog Capture" -top 1 -vf "separatefields, crop=644:224:40:8, scale=600:224:flags=bicubic, scale=600:448:flags=neighbor" -f mpegts udp://127.0.0.1:1234
[01:30:41 CET] <spidersgeorg> I'm trying to add it as a media source in open broadcaster
[01:30:55 CET] <J_Darnley> OMFG!  If you want to deinterlace the use a deinterlacer!
[01:31:06 CET] <spidersgeorg> this does work but the quality loss is terrrible and I haven't even specified a bitrate or anything
[01:31:21 CET] <J_Darnley> I'm not surprised it's shit
[01:31:23 CET] <spidersgeorg> I'm also resizing it as well
[01:31:37 CET] <J_Darnley> yadif is a deinterlacer
[01:31:44 CET] <spidersgeorg> it looks perfect using ffplay
[01:31:44 CET] <J_Darnley> deinterlace then scale
[01:32:01 CET] <spidersgeorg> yadif doesn't work well I'm deinterlacing snes output
[01:32:07 CET] <spidersgeorg> looks washy
[01:32:33 CET] <J_Darnley> Now you say that!
[01:32:44 CET] <J_Darnley> Then specify some encoding options
[01:32:59 CET] <J_Darnley> I assume libx264 is used so: -crf 18
[01:34:51 CET] <spidersgeorg> getting 'past duration 0.xxxxxx is too large' spam
[01:40:06 CET] <spidersgeorg> http://pastebin.com/X82w8WjP
[01:41:11 CET] <spidersgeorg> I can let it run for a bit longer and copy the output if you'd like
[01:43:59 CET] <DHE> if you don't specify defaults you get ~1 megabit mpeg2 video. and that looks like crap
[01:44:39 CET] <spidersgeorg> ah ok
[01:44:45 CET] <spidersgeorg> this sent my cpu through the roof
[01:44:49 CET] <spidersgeorg> -codec:v libx264 -crf 0 -preset veryslow
[01:45:09 CET] <spidersgeorg> which makes sense
[01:45:32 CET] <spidersgeorg> how does a program normally output video to other applications
[01:45:43 CET] <spidersgeorg> surely this is the worst way to go about doing it
[01:45:44 CET] <tp_> try veryfast
[01:45:47 CET] <spidersgeorg> but I don't know any other way
[01:46:07 CET] <J_Darnley> oh lord mpeg2
[01:46:17 CET] <J_Darnley> -qscale 2
[01:46:40 CET] <DHE> try -crf 10 for nearly lossless but presentable bitrate
[01:46:59 CET] <J_Darnley> Only if he was usinglibx264
[01:48:19 CET] <J_Darnley> I wouldn't try to "output to other applications"
[01:48:48 CET] <spidersgeorg> what would you suggest
[01:49:00 CET] <J_Darnley> What else are you trying to do?
[01:49:12 CET] <spidersgeorg> amarectv is a popular solution that most people use to capture their older consoles but it sucks imo
[01:49:30 CET] <J_Darnley> You've just captured with ffmpeg.
[01:49:40 CET] <spidersgeorg> I'm trying to add my console as a source in obs
[01:49:51 CET] <spidersgeorg> yes but I can't pipe ffmpeg to obs
[01:50:23 CET] <J_Darnley> Isn't obs just ffmpeg with some more fancy features?  Can't that capture from dshow?
[01:51:22 CET] <spidersgeorg> you can add a media source by specifying a source like I'm trying to do udp://127.0.0.1
[01:51:50 CET] <spidersgeorg> or add a capture device but it doesn't support deinterlacing atm
[01:51:57 CET] <spidersgeorg> and it's very limited
[01:52:11 CET] <spidersgeorg> doesn't support the use of external filters
[01:52:23 CET] <spidersgeorg> maybe if there was a way for ffmpeg to output to a virtual device
[01:53:21 CET] <spidersgeorg> I guess I could just stream with ffmpeg by itself but how would I add window capture sources?
[01:53:55 CET] <spidersgeorg> alongside my console capture I also capture my webcam and a few application windows
[01:54:07 CET] <spidersgeorg> obs provides a gui for resizing and arranging everything
[01:54:23 CET] <spidersgeorg> could all of this be done through ffmpeg?
[01:56:00 CET] <J_Darnley> ffmpeg can do all of that.  The fancy arranging will be hard though.
[01:57:23 CET] <J_Darnley> capturing specific windows is hard on Windows I think
[01:57:57 CET] <spidersgeorg> looks like you can capture the desktop but then I would need to somehow shift the source over to hide everything else I wouldn't want to show...
[01:58:20 CET] <J_Darnley> input devices can be found here: http://ffmpeg.org/ffmpeg-devices.html
[01:59:32 CET] <spidersgeorg> I could also just keep using obs and capture an ffplay window but it's playback is horrible
[01:59:48 CET] <spidersgeorg> it drops a ton of frames I'm not sure if that's because I'm configuring something wrong
[02:00:06 CET] <J_Darnley> it's probably just too slow
[02:00:30 CET] <J_Darnley> continue with obs if it works for you
[02:00:33 CET] <spidersgeorg> if obs uses ffmpeg to display my capture device how can it keep up?
[02:01:21 CET] <spidersgeorg> can I allocate more priority to ffplay when capturing + filtering?
[02:01:43 CET] <spidersgeorg> hardly uses any cpu when running atm
[02:02:45 CET] <spidersgeorg> ffplay -f dshow -framerate 29.97 -i video="GV-USB2, Analog Capture" -top 1 -vf "separatefields, crop=644:224:40:8, scale=600:224:flags=bicubic, scale=600:448:flags=neighbor"
[02:03:03 CET] <spidersgeorg> this displays everything nicely minus the dropping frames
[02:03:24 CET] <spidersgeorg> I do get a warning that the data is not aligned and can lead to speedloss
[02:09:01 CET] <caveat-> I'm trying to record audio from an application, following this https://trac.ffmpeg.org/wiki/Capture/ALSA#Recordaudiofromanapplicationwhilealsoroutingtheaudiotoanoutputdevice  but when I do: mpv out.wav I get http://sprunge.us/TShH
[02:09:47 CET] <J_Darnley> Blame mpv?
[02:12:43 CET] <caveat-> J_Darnley: same thing using whatever player
[02:13:05 CET] <J_Darnley> The blame your system?
[02:13:18 CET] <J_Darnley> What does ffmpeg say aboutthe file?
[02:14:18 CET] <jarainf> It's not the file
[02:14:27 CET] <jarainf> mpv can't initialize the audio output
[02:15:01 CET] <jarainf> You need to have jack installed and if I recall you also have to load the loopback module
[02:15:23 CET] <jarainf> Did you do "modprobe snd-aloop pcm_substreams=1"?
[02:15:45 CET] <caveat-> jarainf: yes
[02:15:52 CET] <caveat-> and jack is installed
[02:16:18 CET] <caveat-> jarainf: you mean ffprobe file.wav? http://sprunge.us/XbZO
[02:16:30 CET] <caveat-> sorry, for J_Darnley
[02:17:11 CET] <J_Darnley> Yeah.  So what's wrong with the file?
[02:17:15 CET] <J_Darnley> Nothing.
[02:17:25 CET] <jarainf> ^this
[02:17:33 CET] <jarainf> It's the .asoundrc
[02:17:35 CET] <J_Darnley> You can't play it because your system seems to lack any audio outout.
[02:17:51 CET] <caveat-> also, if I remove the .asoundrc file created for the recording using the loopback, I don't get that message anymore, the player starts, but it's mute
[02:18:31 CET] <jarainf> did you overwrite an .asoundrc?
[02:18:46 CET] <caveat-> no, no, with other audio files I have sound
[02:19:04 CET] <jarainf> I still think you didn't execute the modprobe command
[02:19:22 CET] <caveat-> just for the ones with sound recorded from applications using the loopback alsa device
[02:19:30 CET] <jarainf> Well yes
[02:19:36 CET] <jarainf> because your loopback isn't working
[02:19:42 CET] <jarainf> probably
[02:20:13 CET] <caveat-> lsmod | grep aloop    http://sprunge.us/FRHE
[02:20:49 CET] <jarainf> aplay -L output pls
[02:21:13 CET] <caveat-> http://sprunge.us/GTZB
[02:21:32 CET] <jarainf> and your .asoundrc/
[02:21:33 CET] <jarainf> ?
[02:22:09 CET] <caveat-> jarainf: I have copy pasted from https://trac.ffmpeg.org/wiki/Capture/ALSA#Recordaudiofromanapplicationwhilealsoroutingtheaudiotoanoutputdevice
[02:22:22 CET] <caveat-> http://sprunge.us/MbER
[02:22:27 CET] <jarainf> well
[02:22:47 CET] <jarainf> you need to insert your audio device in pcm.output { }
[02:23:15 CET] <jarainf> It may be "card Intel"
[02:23:49 CET] <caveat-> jarainf: I get it from alsamixer?
[02:23:51 CET] <debianuser> caveat-: You can collect and paste them all at once using alsa-info script: https://wiki.ubuntu.com/Audio/AlsaInfo (that dumps `aplay -l`, `aplay -L`, `cat /proc/asound/cards`, your mixer settings, etc.) Run it as a regular user, it doesn't need root.
[02:24:04 CET] <jarainf> caveat-, that should work too
[02:26:46 CET] <debianuser> (caveat-: also I'd suggested you a bit different dmix-based config: http://pastebin.com/B7iEKBwH, just replace "RealCardName" in lines 5 and 25 with the name of your card, you can see card names with `cat /proc/asound/cards`, the "Loopback" card should be there too)
[02:29:14 CET] <caveat-> http://www.alsa-project.org/db/?f=605b2dd1f2dc5cf33d1649820e5a9a33e8a1dd48
[02:29:20 CET] <jarainf> I'd suggest not using Loopback at all :33333333333
[02:30:40 CET] <debianuser> caveat-: jarainf was right, your card is named "Intel". So you need to replace "card Headset" with "card Intel" in your config.
[02:31:06 CET] <caveat-> jack is not running, if that matters
[02:31:28 CET] <jarainf> debianuser, that's the first time I've ever seen anyone using the Intel Sound
[02:31:39 CET] <jarainf> I'm genuinely surprised right now
[02:32:36 CET] <debianuser> caveat-: Optionally you can use my config (they're based on the same idea, but mine uses dmix/dsnoop and allows multiple apps to play sound at the same time), where you'd need to replace "RealCardName" in lines 5 and 25 with the name "Intel".
[02:33:05 CET] <debianuser> caveat-: Also, my config doesn't have "loopout" pcm, you need to append it:     pcm.loopout { type plug; slave.pcm "dsnoop:Loopback,1" }    Then your command should finally work. I hope!
[02:35:13 CET] <debianuser> caveat-: If it still doesn't work - feel free to ask, we'll fix that!
[02:37:07 CET] <caveat-> ok, thanks, I have to metabolize your suggestions
[03:49:15 CET] <solrize> any idea what would cause "Unrecognized option 'headers'"?  i found some slightly similar messages with web search that were related to missing modules
[03:49:22 CET] <solrize> but this was just about an mp4 file
[03:56:19 CET] <c_14> solrize: there is no such option?
[03:56:44 CET] <solrize> i think it may be because i had networking turned off because of that bug a while ago
[03:56:54 CET] <solrize> that's fixed now right?  i'm recompiling with networking back on
[03:57:41 CET] <solrize> the compile will take a couple more minutes
[03:59:19 CET] <c_14> The HLS one?
[04:00:20 CET] <solrize> c_14, not sure what HLS is, there was a security announcement a month or so ago that said recompile with --disable-network because of some vulnerability
[04:01:18 CET] <solrize> hmm now it says https not found, recompile with openssl, but i used --enable-openssl.  pastebin coming, sec
[04:01:55 CET] <solrize> here's my current config command: http://lpaste.net/9190805150980112384
[04:02:36 CET] <c_14> Can you pastebin the config.log?
[04:04:32 CET] <solrize> sec
[04:05:25 CET] <solrize> um it's 300kb
[04:05:32 CET] <solrize> 11000 lines
[04:06:08 CET] <c_14> The first and last 50 lines should be enough
[04:06:17 CET] <solrize> oh ok.  or i can upload it to a web server
[04:06:44 CET] <c_14> Might be better in case the openssl check is somewhere in the middle
[04:08:17 CET] <solrize> http://lpaste.net/3470756646626525184
[04:08:28 CET] <solrize> oh hmm yeah i can search it
[04:09:26 CET] <solrize> ok will try to upload it.  i see several messages indicating openssl is there and no obvious errors
[04:09:28 CET] <solrize> sec
[04:10:26 CET] <c_14> Are you sure you're running the correct binary then? Check the version and compilation flags
[04:14:56 CET] <solrize> hmm
[04:15:42 CET] <solrize> i run ffmpeg --help and it doesn't show openssl
[04:16:18 CET] <solrize> ffmpeg version N-78335-g9ee4c89 Copyright (c) 2000-2016 the FFmpeg developers
[04:16:28 CET] <solrize> that's supposed to be from master as of about 20 minutes ago
[04:17:34 CET] <c_14> That is the latest commit, yes
[04:18:32 CET] <c_14> Can you upload the whole config.log?
[04:22:31 CET] <solrize> yeah just a sec it's taking some messing around to do that but it's almost there
[04:22:46 CET] <solrize> http://configlog.phr.fastmail.fm/config.log
[04:24:35 CET] <c_14> It should definitely be there... make distclean and try again?
[04:26:10 CET] <solrize> will try though i've done that several times... about 3 minutes to recompile, brb
[04:30:00 CET] <solrize> hmm now openssl shows up in the help message
[04:30:24 CET] <solrize> whoa it's working now, wonder what was wrong before... thanks!
[04:31:46 CET] <solrize> might have gotten compiles and configs out of sync
[04:37:00 CET] <solrize> hmm now i'm trying to transcode the mp4 to something smaller and i get a pile of messages like "Past duration 0.643990 too large"  is there a way to get rid of that?  i remember encountering that before and spending a while looking into it and not getting anywhere definite
[04:37:59 CET] <solrize> oh i see a bunch of web hits, that say to ignore it.  ok
[10:17:30 CET] <dorp> When it comes to transcoding duration time, x264 with preset=ultrafast and qp=0, is supposed to be much more efficient than x265?
[12:29:32 CET] <markvandenborre> hi!
[12:30:10 CET] <markvandenborre> I have a pile of somewhat older quad socket hexacore servers with oozes of ram (512 Gb)
[12:30:48 CET] <markvandenborre> I wonder if there would be a way to do live webm vp9 streaming from those
[12:31:26 CET] <markvandenborre> in other words, is there a way to do multithreaded vp9 encoding for live streams using ffmpeg?
[12:31:35 CET] <markvandenborre> any hints appreciated!
[12:36:59 CET] <pzich> https://www.google.com/search?q=multithreaded+vp9+ffmpeg
[12:37:03 CET] <pzich> looks like yes? maybe?
[12:52:21 CET] <kolo> ffmpeg -i input.mp4 -vf subtitles=input.srt out.mp4
[12:54:13 CET] <kolo> https://up1.ca/#0JqnVoi7Bsf_-IAx-Fqc5g --> Is it possible to burn subtitles in the ~ middle of screen?
[12:56:31 CET] <kolo> not possible?!
[13:21:54 CET] <kolo> Is it possible to burn subtitles in the ~ middle of screen?
[13:21:59 CET] <kolo> https://www.filepicker.io/api/file/BiG9VCYNThymE1OA6DlZ
[13:38:05 CET] <kolo> I want to burn a transparent  picture file (.png) at a given timecode on video, is it possible?
[13:38:29 CET] <kolo> I do it for subtitles in this way: : ffmpeg -i input.mp4 -vf subtitles=input.srt out.mp4
[13:38:47 CET] <kolo> Can I do that for PNGs?
[13:39:31 CET] <Mavrik> overlay filter I guess.
[13:39:35 CET] <Mavrik> There's a guide in the wiki.
[13:41:02 CET] <kolo> I this methode the best way to write sth on the video?
[13:42:50 CET] <kolo> Mavrik: www.ffmpeg.org/ffmpeg-filters.html#overlay-1 --> It's saying video on video; Is it the same for image on video?
[14:17:47 CET] <PowerKiller> Is it possible to reencode NTSC into a MP4
[14:17:53 CET] <PowerKiller> so it's playable on computer?
[14:28:32 CET] <PowerKiller> OK got it
[14:28:50 CET] <PowerKiller> Gotta use WinXDVD in a VM and then reencode the mp4 into webm
[14:28:57 CET] <PowerKiller> shd. work
[15:06:57 CET] <MarkusDB1> I'm looking for a good tutorial to stream mpeg-dash or webm. I've been doing lots of googling, but all this seems like very new tech, is it? Is there a good guide out there?
[16:04:46 CET] <NicCo> HI
[16:05:10 CET] <NicCo> I would like to use -report on Windows
[16:05:53 CET] <NicCo> I would like to report a log for each file converted, is it possible to specify a path for log file or I must use environment variable ?
[16:09:49 CET] <J_Darnley> Are you starting many commands at the exact same second?
[16:10:50 CET] <NicCo> No I'm launching only one conversion at the same time
[16:11:03 CET] <J_Darnley> Then what's wrong with the default filename?
[16:11:35 CET] <NicCo> I would like to convert files on different folder and I would like to put each log on each folder
[16:12:11 CET] <J_Darnley> When why not cd to the right folder?
[16:12:59 CET] <J_Darnley> To answer your first question: no, the env var is the only way.
[16:13:50 CET] <NicCo> Ok thank you so 1) I use env var 2) log is allways into ffmpeg.exe folder ?
[16:14:06 CET] <J_Darnley> No.  RTFM
[16:14:19 CET] <J_Darnley> -report
[16:14:19 CET] <J_Darnley>     Dump full command line and console output to a file named program-YYYYMMDD-HHMMSS.log in the current directory.
[16:14:27 CET] <J_Darnley> *the current directory*
[16:15:10 CET] <NicCo> oh osrry
[16:15:13 CET] <NicCo> sorry
[16:16:00 CET] <NicCo> so I understand why you said to use cd
[16:18:58 CET] <NicCo> thank you ;)
[17:14:06 CET] <thebombzen> what happened to libquvi support?
[17:17:02 CET] <chungy> removed
[17:17:25 CET] <chungy> libquvi hasn't been updated for a long time. Both security issues and usability issues (doesn't work on many sites anymore)
[17:18:30 CET] <thebombzen> I see.  I guess I'll have to use quvi's frontend and pipe it
[17:18:32 CET] <JEEB> youtube-dl should in general be used instead for similar functionality
[17:18:41 CET] <JEEB> (not limited to youtube)
[17:18:49 CET] <thebombzen> is that a CLI program?
[17:18:51 CET] <JEEB> yes
[17:19:06 CET] <JEEB> the mpv video player has integration for it, for example
[17:19:52 CET] <thebombzen> does FFmpeg support that, or is that the kind of thing I would have to pipe &/| save to a file first?
[17:20:53 CET] <JEEB> ffmpeg doesn't have any integration as far as I know, so piping would be it
[17:21:43 CET] <bencoh> youtube-dl as --exec
[17:21:52 CET] <bencoh> has*
[17:21:58 CET] <bencoh> (just like quvi)
[17:23:01 CET] <thebombzen> it appears youtube-dl has the same problem that TeX has. that it's a program rather than a library
[18:50:17 CET] <valdikss> Hello guys! I have the following video: http://b.1339.cf/abpdfut.webm
[18:50:50 CET] <valdikss> The first part of it (before opening) is 23.98 fps and opening is 29.97. How can I re-encode it with ffmpeg without framedrop or stuttering?
[18:54:30 CET] <ChocolateArmpits> valdikss: probably -vf fps=30000/1001
[18:57:08 CET] <valdikss> ChocolateArmpits: this will produce file with constant fps. I need to save vfr.
[18:57:49 CET] <valdikss> ChocolateArmpits: I tried vsync 0, copyts, copytb and it's variations.
[18:58:45 CET] <ChocolateArmpits> uh then sorry, I'm not familiar with vfr encoding
[18:58:46 CET] <valdikss> ChocolateArmpits: I have either stuttering in opening or non-strictly-monotonic PTS notices.
[18:59:35 CET] <ChocolateArmpits> are you sure the opening is actually at 29.97 rather than just playing at that framerate ?
[18:59:46 CET] <valdikss> ChocolateArmpits: it seems ffmpeg ignores containers' pkt_duration
[19:00:15 CET] <valdikss> ChocolateArmpits: yes i'm sure, I've encoded episode and opening and joined it with mkvmerge.
[19:00:34 CET] <ChocolateArmpits> what I mean the opening doesn't have duplicated frames
[19:00:57 CET] <valdikss> ChocolateArmpits: &ignores pkt_duration and sets pkt_duration=41 in both episode and op, while it should be set to 33 in op
[19:01:07 CET] <valdikss> ChocolateArmpits: yes, all frames in opening are unique.
[21:31:42 CET] <spookypeanut> has anybody tried using audio & video filters on a .vob (ripped with vobcopy, in case it matters)?
[21:32:01 CET] <spookypeanut> i'm getting my video and audio out of sync, trying to troubleshoot
[21:33:46 CET] <Zitter> Hi, I have a video file with audio. Now I would like to add another audio to that video (I have in a .mp3 file). any hint on how to do that?
[21:34:16 CET] <Zitter> I mean I don't want to lose the original audio, only adding the new one
[21:35:18 CET] <Mavrik> Pass original file and the mp3 file as input, use "-map" parameters to map all three streams and use "-codec copy" so things won't get encoded.
[21:38:24 CET] <Mavrik> ask which parts are unclear ;)
[21:46:57 CET] <Zitter> I've tried: ffmpeg -i second_part_volume05.mp4 -i 20160128-144446.mp4 -map 0:0 -map 0:1 -map 1:0 -codec copy output.mp4 and I hear only audio from first source (second_part_volume05)
[21:47:28 CET] <Mavrik> hmm
[21:47:41 CET] <Mavrik> are timestamps on the tracks synced?
[21:47:58 CET] <Mavrik> also pastebin command and output of ffmpeg
[21:48:49 CET] <Zitter> If I do -map 0:0 -map 1:0 I can hear output from second file but not from the first one
[21:53:36 CET] <Zitter> ah, I undestand. It add a "track" audio so it means that I can switch from an audio source to another one
[21:53:50 CET] <Zitter> but it isn't what I need
[21:54:01 CET] <Mavrik> well, you said that's what you need
[21:54:05 CET] <Mavrik> :)
[21:54:44 CET] <Zitter> mmm
[21:55:04 CET] <Zitter> sorry I'm not so good to explain my needs in english
[21:55:06 CET] <Zitter> :D
[21:55:33 CET] <Zitter> I need to mixing two audio (one with a lower volume, of course)
[21:55:34 CET] <Mavrik> ^^
[21:55:36 CET] <Mavrik> Ah.
[21:55:50 CET] <Mavrik> So you want to hear both audios at once and you want them to be a single stream in the output?
[21:56:26 CET] <Zitter> yes
[21:58:24 CET] <Mavrik> hmm, you'll have to reencode the audio then
[21:58:31 CET] <Mavrik> try something like
[21:59:13 CET] <Zitter> maybe amix filter? I'm looking on docs
[21:59:35 CET] <Mavrik> ffmpeg -i <video> -i <audio> -codec:v copy -codec:a aac -filter_complex "[0:1][1:0] amix=inputs=2" -b:a <audio_bitrate> output.mp4
[21:59:43 CET] <Mavrik> yeah, amix seems the one you need :)
[22:01:00 CET] <Zitter> which value for <audio_bitrate>? sorry, I'm really new to these things
[22:03:28 CET] <Mavrik> well
[22:03:34 CET] <Mavrik> which ever you want
[22:03:44 CET] <Mavrik> if you want to keep quality the same, use the same one as the input has
[22:03:56 CET] <Mavrik> but I can't tell you the number because you didn't show the ffmpeg output that shows those values :P
[22:06:50 CET] <Zitter> bitrate 1394 kb/s
[22:07:15 CET] <Zitter> I've tried with first 2 minutes to see if it works... and it works :)
[22:07:29 CET] <Zitter> Only problem is that audio of first stream is too high
[22:10:33 CET] <Mavrik> huh
[22:10:38 CET] <Mavrik> 1394 is like 10x too much
[22:10:44 CET] <Mavrik> Are you mixing 5.1 AC3 or something?
[22:12:39 CET] <Zitter> maybe :)
[22:17:07 CET] <Mavrik> Well..
[22:17:13 CET] <Zitter> *cough*
[22:17:26 CET] <Mavrik> if you won't tell what you're dealing with I can't help you anymore.
[22:18:01 CET] <Zitter> listen, where can I learn basics from audio stuff? I mean: what codec is, bitrate and so on? Is there a nice guide on the web?
[22:18:18 CET] <Zitter> audio/video I mean
[22:45:40 CET] <slimyman> hi
[22:46:03 CET] <slimyman> fu
[22:46:04 CET] <slimyman> jk
[00:00:00 CET] --- Mon Feb  8 2016