[Ffmpeg-devel-irc] ffmpeg.log.20160410

[burek]

[01:23:44 CEST] <ethe> Is there a flag for ./configure to disable fixed math stuff only?
[01:25:09 CEST] <JEEB> --disable-decoder=aacdec_fixed
[01:25:17 CEST] <JEEB> since that was the one you were having an issue with
[01:25:29 CEST] <JEEB> unless it pops up differently in the configure output
[01:25:35 CEST] <JEEB> in which case I got the decoder name incorrect :P
[01:25:47 CEST] <ethe> hmm, I thought it would be an issue with all the fixed encoders, but actually I think the other ones were fine
[01:26:53 CEST] <ethe> JEEB yeah it's an encoder
[01:27:26 CEST] <ethe> thanks anyways
[01:28:40 CEST] <ethe> well, apparently now it's a decoder .-.
[01:29:44 CEST] <JEEB> I saw CC      libavcodec/aacdec_fixed.o
[01:29:49 CEST] <JEEB> in your output :P
[01:30:56 CEST] <JEEB> and yeah, there is just one aac encoder in FFmpeg afaik :)
[01:30:59 CEST] <ethe> yeah... --disable-decoder=aac_fixed was the flag which worked in the end
[01:31:36 CEST] <JEEB> http://fatebeta.ffmpeg.org/ there's a couple of aarch64 machines in FATE nowadays
[01:32:13 CEST] <JEEB> some iOS devices, a qemu instance at least
[01:35:16 CEST] <ethe> my suspicions were correct, ac3dec_fixed failed as well, so it looks like it's all of the fixed decoders which are not working
[01:35:55 CEST] <ethe> although I think those are the only two fixed decoders
[04:00:04 CEST] <drwho7817> I need help recording desktop audio from headset with built in microphone. Can someone help point me in the right direction?
[04:00:42 CEST] <drwho7817> This is my .asoundrc # .asoundrc pcm.!default { type plug slave.pcm hw:1 }  defaults.ctl.card 1 defaults.pcm.card 1
[04:01:57 CEST] <drwho7817> What I am really trying to do is stream. I get the stream to work and it picks up my microphone but not my desktop sound. I followed this guide https://trac.ffmpeg.org/wiki/EncodingForStreamingSites
[04:02:33 CEST] <drwho7817> I use Streaming your desktop  Without scaling the output and get video to stream picks up mic but no desktop sound.
[04:03:58 CEST] <c_14> https://trac.ffmpeg.org/wiki/Capture/ALSA#Recordaudiofromanapplication
[04:04:34 CEST] <drwho7817> c_14: I have tried this but I must be doing something wrong since it does not pick up audio.
[04:05:07 CEST] <drwho7817> I make sure to load module first copied that script then edited my sound card.
[04:05:09 CEST] <c_14> Well, the part of your asoundrc that you pasted definitely doesn't capture desktop  audio
[04:05:31 CEST] <c_14> What's your ffmpeg command?
[04:06:33 CEST] <drwho7817>  ffmpeg -f alsa -ac 2 -ar 44100 -i hw:1 out.wav
[04:06:48 CEST] <drwho7817> ffmpeg -f alsa -ac 2 -ar 44100 -i hw:Loopback,0,0 out.wav
[04:06:56 CEST] <drwho7817> I have tried both
[04:07:52 CEST] <c_14> The first one will record your microphone, the second one won't do anything because you never modified your asoundrc
[04:08:53 CEST] <drwho7817> I changed my asoundrc per the guide you gave me https://trac.ffmpeg.org/wiki/Capture/ALSA#Recordaudiofromanapplication now where the card <Your Output Device Name>, I edited to say card 1 per my arecord -l output
[04:10:52 CEST] <c_14> If you modified your asoundrc like in the guide you have to do -i hw:Loopback,1,0
[04:11:25 CEST] <drwho7817> ffmpeg -f alsa -ac 2 -ar 44100 -i hw:Loopback,1,0 out.wav <--- this command right?
[04:11:37 CEST] <c_14> yes
[04:11:40 CEST] <drwho7817> I just adjusted my asound.
[04:11:44 CEST] <drwho7817> asoundrc*
[04:12:05 CEST] <drwho7817> I am getting this error now [alsa @ 0x23a5480] cannot open audio device hw:Loopback,1,0 (Invalid argument) hw:Loopback,1,0: Input/output error
[04:12:24 CEST] <drwho7817> I already ran alsactl restore
[04:12:28 CEST] <drwho7817> Should I have to reboot?
[04:12:57 CEST] <c_14> no
[04:13:29 CEST] <c_14> your asoundrc has pcm.!default { type plug slave.pcm "hw:Loopback,0,0" } right?
[04:13:58 CEST] <drwho7817> No it only has that long script that mentions the loopback
[04:14:06 CEST] <drwho7817> Should I add that command if so where?
[04:14:32 CEST] <c_14> That's not a command, that's a line in your .asoundrc
[04:14:41 CEST] <c_14> Can you upload your current .asoundrc to a pastebin service?
[04:15:35 CEST] <drwho7817> 1 sec please
[04:16:45 CEST] <drwho7817> http://pastebin.com/7sqUiTaz
[04:17:44 CEST] <c_14> Ok, delete line 2
[04:17:58 CEST] <c_14> Then do ffmpeg -f alsa -ac 2 -ar 44100 -i loopout out.wav
[04:19:13 CEST] <drwho7817> testing now
[04:19:52 CEST] <drwho7817> no sound
[04:20:52 CEST] <drwho7817> This is my arecord -l output http://pastebin.com/cSY7cdj8
[04:22:27 CEST] <c_14> are you running an application that produces sound?
[04:22:53 CEST] <c_14> Can you hear anything on the output pcm? (card 1 in your case)
[04:22:56 CEST] <drwho7817> sure I test youtube or music stream on vlc
[04:23:05 CEST] <drwho7817> Yeah I can
[04:23:52 CEST] <c_14> Does arecord -D loopout -t wav -c 2 -f cd out.wav work?
[04:24:59 CEST] <drwho7817> I had pulseaudio install just removed it in case it was a conflict
[04:26:09 CEST] <drwho7817> I run your command and get no error but nothing happens either
[04:27:44 CEST] <drwho7817> I think my sound card does not have stereo mix capabilities does this matter or is this what the loopback is for?
[04:27:57 CEST] <eftm> some opencv files compiled during installation for their python library include <ffmpeg/avformat.h>, and *as they are now* these headers files all actually reside under /usr/lib/<avlibraryname>/<avlibraryfiles>. how would i collect them all under the ffmpeg label if they have files that are named the same (like version.h)? or how else would i fix t
[04:27:57 CEST] <eftm> he library organization so they can be recognized by the <ffmpeg/[[avlibraryname.h]]> format? my initial thought was to make the ffmpeg folder and provide symlinks to all of them, but they appear to also need whatever other files belong in each of their current folders
[04:31:39 CEST] <c_14> drwho7817: shouldn't matter afaik, the installed pulse might be it though, it might have left behind some configs, maybe check /usr/share/alsa ?
[04:31:47 CEST] <c_14> eftm: sed the opencv files
[04:33:01 CEST] <drwho7817> c_14: shouldnt the loopin also be set to pcm.loopin {         type plug         slave.pcm "hw:Loopback,1,0"  as my pcm.loopout? and my card 1 in that script?
[04:33:45 CEST] <gcl5cp> i have had problem applying png watermark (overlay) pre-processed by imagemagick, raise a libav core dump error.
[04:34:17 CEST] <gcl5cp> but using gimp to save the alpha png file works well.
[04:34:28 CEST] <drwho7817> c_14: in other words is my asoundrc correct? http://pastebin.com/Sm9xVdPy
[04:34:57 CEST] <drwho7817> or should i leave it default and only edit my card 1 to reflect my sound card being my headset
[04:35:19 CEST] <c_14> drwho7817: no, the slave pcms for loopin and loopout have to use different subdevices
[04:35:37 CEST] <drwho7817> okay I will change it back i am getting an error now anyway
[04:35:49 CEST] <c_14> gcl5cp: what version are you getting the error with?
[04:36:28 CEST] <gcl5cp> the gimp's file weight 2.25KB, and the imagemagick's file 471B
[04:37:33 CEST] <drwho7817> c_14: should i uninstall alsa remove scripts reboot and start fresh?
[04:37:46 CEST] <gcl5cp> ffmpeg version 2.8.1-2~trusty Copyright (c) 2000-2015
[04:38:41 CEST] <c_14> gcl5cp: if you can reproduce with a build from git, open a bug report on trac
[04:39:20 CEST] <c_14> drwho7817: you could do that, what error are you getting though?
[04:39:41 CEST] <drwho7817> im using the git build ffmpeg version git-2016-03-15-7725210 Copyright (c) 2000-2016 the FFmpeg developers
[04:40:03 CEST] <drwho7817> [alsa @ 0x15c3480] cannot open audio device loopout (Device or resource busy) loopout: Input/output error
[04:40:12 CEST] <gcl5cp> ok c_14
[04:40:45 CEST] <drwho7817> when i run ffmpeg -f alsa -ac 2 -ar 44100 -i loopout out.wav
[04:41:49 CEST] <c_14> drwho7817: hmm, strange. do you still have the arecord running or anything else which might be trying to access that device. Maybe an output program from when you had loopin and loopout with the same subdevice?
[04:42:28 CEST] <drwho7817> ah yeah i see arecord running let me kill process
[04:43:15 CEST] <drwho7817> good now
[04:44:07 CEST] <drwho7817> still no audio
[04:44:21 CEST] <drwho7817> Maybe I need wav codecs installed?
[04:44:37 CEST] <c_14> no
[04:45:03 CEST] <c_14> All you should need is a kernel built with alsa, alsa-lib, alsa-utils and maybe alsa-plugins but I'm not sure about the last one
[04:45:46 CEST] <drwho7817> should I recompile ffmpeg from guide? maybe i did something wrong
[04:46:04 CEST] <drwho7817> how can i check to see if kernel has those headers compiled?
[04:48:04 CEST] <c_14> If you can listen to audio your kernel is most probably built with alsa (unless it uses OSS, which would be unusual)
[04:48:13 CEST] <c_14> If arecord can't pick it up, it's probably not ffmpeg
[04:48:55 CEST] <drwho7817> yep I can hear audio from internet stream, local music file and youtube via browser
[04:49:55 CEST] Action: c_14 just copied the .asoundrc to his .asoundrc and it works fine here
[04:51:00 CEST] <drwho7817> hmm should i check my bios?
[04:51:05 CEST] <drwho7817> could it be a setting there?
[04:51:36 CEST] <c_14> very unlikely
[04:51:38 CEST] <c_14> hmm
[04:51:50 CEST] <c_14> Do you have more than one sound card? Something like usb headphones + speakers or something?
[04:52:02 CEST] <drwho7817> then its got to be my headset or soundcard
[04:52:12 CEST] <drwho7817> i have a usbheadset
[04:52:17 CEST] <drwho7817> let me connect it and try that
[04:53:01 CEST] <c_14> What I want to try is setting the "loopin" slave of the multi pcm to the other sound card / usb headset and checking to make sure that sound plays on both
[04:53:10 CEST] <c_14> To make sure that the route is actually doing its job
[04:54:15 CEST] <drwho7817> okay i connected my usb headset
[04:55:35 CEST] <drwho7817> slave.pcm "hw:Loopback,3,0"
[04:55:45 CEST] <drwho7817> arecord -l says my usb headset is card 3
[04:56:34 CEST] <c_14> then it should just be hw:3
[04:56:39 CEST] <drwho7817> i am not getting any audio now out of my usb headset
[04:57:21 CEST] <c_14> are you sure you uninstalled pulse?
[04:58:08 CEST] <drwho7817> this is my new pastebin asoundrc http://pastebin.com/FCN5jAdV
[04:58:19 CEST] <drwho7817> yes i ran apt-get purge pulseaudio pavucontrol
[04:58:58 CEST] <drwho7817> it looks like pulseutils is still installed'
[04:59:37 CEST] <c_14> can you just switch the slave.pcm for pcm.!default to "output" ?
[05:00:32 CEST] <c_14> and then see if you get audio on the usb headphones
[05:01:35 CEST] <drwho7817> pcm.output { <------>type hw <------> slave.pcm hw:3 like this?
[05:02:03 CEST] <c_14> line 28 in the config you pasted, just change the part in quotes to output
[05:04:00 CEST] <drwho7817> done
[05:04:12 CEST] <c_14> then try playing something and seeing if the output goes to your usb headphones
[05:05:32 CEST] <drwho7817> nah
[05:05:36 CEST] <drwho7817> i may have to reboot
[05:05:46 CEST] <drwho7817> its not acting right since i switched headphones
[05:05:50 CEST] <drwho7817> or log out
[05:06:15 CEST] <drwho7817> Hardware is initialized using a generic method No state is present for card Headset
[05:08:15 CEST] <drwho7817> be right back
[05:21:59 CEST] <drwho7817> c_14: my pastebin with new usb headset http://pastebin.com/uUk4yVMA
[05:22:14 CEST] <drwho7817> i just rebooted and not getting any sound
[05:22:38 CEST] <c_14> When playing audio normally?
[05:22:43 CEST] <drwho7817> yep
[05:23:08 CEST] <c_14> Did you check /usr/share/alsa ?
[05:24:07 CEST] <c_14> especially alsa.conf.d
[05:24:50 CEST] <drwho7817> what should i look for?
[05:24:52 CEST] <drwho7817> i have not yet
[05:25:15 CEST] <c_14> anything mentioning pulse
[05:25:27 CEST] <drwho7817> oh
[05:25:31 CEST] <drwho7817> i found pulse stuff
[05:25:33 CEST] <drwho7817> :s
[05:25:45 CEST] <drwho7817> 50-pul50-pulseaudio.conf
[05:25:54 CEST] <drwho7817> 99-pulseaudio-default.conf.example
[05:26:23 CEST] <drwho7817> removing now
[05:28:37 CEST] <drwho7817> still no audio logging off be right back
[05:33:35 CEST] <drwho7817_> sound works again i got it to work from a script online http://pastebin.com/GqBVpXrV
[05:33:45 CEST] <drwho7817_> it wont work with the ffmpeg script :/
[05:33:55 CEST] <drwho7817_> im using the usb headset
[05:35:17 CEST] <c_14> for some reason the route pcm just doesn't seem to want to work for you
[05:36:54 CEST] <drwho7817_> hmm it could be my kernel setup
[05:37:16 CEST] <drwho7817_> maybe ill reinstall debian and build from scratch
[05:37:27 CEST] <drwho7817_> ill try that and update here if i get it working
[05:37:35 CEST] <drwho7817_> i really appreciate you helping me look
[05:50:07 CEST] <c_14> np
[13:39:04 CEST] <brad987> Some help with audio conversion?
[13:39:17 CEST] <brad987> (new here)
[13:41:09 CEST] <brad987> I have a video with audio recorded only to the left channel. I need to convert to stereo. Tried -map_channel 0.1.0 and it worked but think it sounds a bit spacey
[13:42:07 CEST] <brad987> Tried amerge but the way I'm doing it the left channel ends up louder - I think because it's getting added twice: once from the input file and second from the amerge filter
[13:42:18 CEST] <brad987> ffmpeg -i trimmed.avi -vn -af "amovie=trimmed.avi [l] ; [l] [l] amerge" amerge4a.avi
[13:42:59 CEST] <brad987> Can I modify this last command to exclude the base audio from the input file?
[14:28:29 CEST] <brad987> meh actually the map_channel seems ok
[15:40:16 CEST] <volar> /join #
[17:28:20 CEST] <fling> Could not open libavcodec encoder for saving images
[17:28:44 CEST] <fling> How to fix? ^ Do I need to enable png somehow?
[17:29:38 CEST] <JEEB> that's why you generally start off with a build that doesn't have things specifically disabled
[17:29:48 CEST] <JEEB> you make sure your code works with such a build
[17:29:57 CEST] <JEEB> then you start optimizing the build for features and size :P
[17:34:03 CEST] <fling> rebuilt ffmpeg and mpv and the error gone
[17:36:58 CEST] Action: JEEB finally got mpv on android kind of nicely going
[17:37:07 CEST] <JEEB> (except for the OSC which only gets half rendered)
[17:39:43 CEST] <fling> The problem was a broken gentoo ebuild probably
[17:40:05 CEST] <fling> As encode flag on mpv does not change the behavior of this&
[20:27:15 CEST] <Mavrik> JEEB, how do you render? OGL?
[20:27:26 CEST] <JEEB> yeah
[20:27:34 CEST] <Mavrik> Bah sorry, forgot to check timestamps on messages
[20:27:47 CEST] <JEEB> it was fun kind of understanding the android gl view workflow
[20:28:15 CEST] <Mavrik> *sigh*
[20:28:18 CEST] <JEEB> (I basically threw stuff at the walls and checked what was sticking)
[20:28:22 CEST] <Mavrik> For certain meanings of the word :D
[20:28:51 CEST] <JEEB> (my WIP branch for making the context stuff work was/is "garbage_day")
[20:29:08 CEST] <JEEB> but yeah, now I get the OSC working, too https://github.com/jeeb/mpv-android/releases/tag/mpv-android-2016-04-10v2
[20:29:24 CEST] <JEEB> (I broke it while making the stuff in general time right)
[20:29:38 CEST] <JEEB> I must say it was much simpler to get libmpv built :P
[20:32:41 CEST] <Mavrik> Yeah, if it helps I had to do that stuff for an app and made a demo: https://github.com/izacus/AndroidOpenGLVideoDemo :)
[20:32:45 CEST] <Mavrik> If you want shaders and whatnot.
[20:32:49 CEST] <Mavrik> (Uses Andorid codecs tho=
[20:33:03 CEST] <JEEB> I'm leaving most of the work to libmpv
[20:33:08 CEST] <JEEB> which is how it should be
[20:33:29 CEST] <JEEB> I just have to stick the cables into the right inputs and outputs
[20:33:35 CEST] <JEEB> between android and libmpv
[20:33:40 CEST] <Mavrik> yep
[20:33:48 CEST] <Mavrik> SW decoding only though :)
[20:33:50 CEST] <JEEB> nah
[20:33:52 CEST] <JEEB> mediacodec just fine
[20:34:08 CEST] <JEEB> this was the thing I tested mateo's thing with before it got merged
[20:34:49 CEST] <Mavrik> Ahh.
[20:34:55 CEST] <Mavrik> So the MediaCodec stuff was merged?
[20:34:58 CEST] <JEEB> yes
[20:35:26 CEST] <Mavrik> Neat. Need to test device compat to see if I can use it somewhere else :)
[20:35:49 CEST] <JEEB> compatibility itself for the mediacodec thing should be >=4.2
[20:36:19 CEST] <Mavrik> Yeah, but that doesn't guarantee it's not broken or funny sadly :/
[20:36:24 CEST] <JEEB> sure
[20:36:27 CEST] <Mavrik> I've had quite a bit of fun on 4.2 on some devices
[20:36:39 CEST] <Mavrik> Either they had strange pix fmts or plainly just didn't work.
[20:36:41 CEST] <JEEB> but that would be not limited to the libavcodec mediacodec implementation
[20:36:44 CEST] <Mavrik> It got better in 4.4+ though.
[20:36:51 CEST] <Mavrik> Yeah, it doesn't work anywhere then :)
[20:37:18 CEST] <JEEB> in my case the first thing you notice between devices is the way those things fall back
[20:37:43 CEST] <Mavrik> As long as they fall back instead of failing horribly with segfault it's great
[20:37:48 CEST] <JEEB> aka "I feed you 10bit AVC and let's see how badly you are able to let me fall back onto sw dec"
[20:38:04 CEST] <JEEB> my oneplus one is pretty straightforward and works
[20:38:13 CEST] <JEEB> then I think some nexus device just stuck there for 30s
[20:38:19 CEST] <JEEB> others can be even worse :)
[20:38:42 CEST] <JEEB> which is why I think most apps have a profile/level check before feeding to hwdec
[20:39:25 CEST] <Mavrik> <3 MediaTek!
[20:39:27 CEST] <Mavrik> <3 AllWinner
[20:39:35 CEST] <JEEB> yup
[20:39:38 CEST] <Mavrik> "My Hofer Medion tab doesn't work!"
[20:46:57 CEST] <mateo`> JEEB, Mavrik: I've send the mediacodec hwaccel part to the ml to render the output buffers to a surface (in case you didn't notice)
[20:52:02 CEST] <JEEB> yeah, I did notice it :)
[20:52:11 CEST] <JEEB> haven't gotten to writing code for it yet
[20:53:57 CEST] <mateo`> I also have some code to manipulate Surface+SurfaceTexture but i'm not sure I will submit it for upstreaming as it depends on a java class bundled with the application (wich is mandatory, as you have to implement an interface to received SurfaceTexture callbacks)
[20:57:55 CEST] <mateo`> (I think I will send the patch anyway)
[21:49:40 CEST] <Guest69441> Hiiiiiii all
[21:49:58 CEST] <Guest69441> Could you please guide me
[21:50:34 CEST] <Guest69441> i wish to encode a video by ffmpeg
[21:51:05 CEST] <Guest69441> i like that the result be a coded audio and a coded video
[21:51:14 CEST] <Guest69441> is it possible?
[21:52:00 CEST] <Guest69441> suppose my original file is yuv
[21:53:06 CEST] <Guest69441> i mean that  i have one file (yuv) and i like two have to encoded (one for video and one for audio)
[21:55:47 CEST] <furq> Guest69441: -i src.yuv -an -c:v foo video.mkv -vn -c:a bar audio.mkv
[21:56:55 CEST] <Guest69441> Oh thanksssss
[22:02:23 CEST] <JEEB> ugh
[22:02:35 CEST] <JEEB> Mavrik: people are trying to remove libfaac wrapper
[22:02:49 CEST] <JEEB> and they're being blocked by "BUT CONSIDER THE USER CHOICE!!!!!!"
[22:02:51 CEST] <Mavrik> Hmm, does anyone still use that? :)
[22:03:02 CEST] <Mavrik> Sometimes the users are morons :P
[22:03:09 CEST] <JEEB> supposedly because faac is 60x real time
[22:03:18 CEST] <JEEB> while aac/fdk-aac are ~30x real time
[22:03:47 CEST] <Mavrik> Hrmf.
[22:03:56 CEST] <JEEB> but yeah, ended up posting in that mailing list thread
[22:03:58 CEST] <Mavrik> Trying to think of an esoteric case where that matters.
[22:04:13 CEST] <JEEB> "running mencoder or ffmpeg on your phone" is the only thing that was brought up
[22:04:17 CEST] <JEEB> as in, cli
[22:04:19 CEST] <JEEB> :V
[22:04:24 CEST] <JEEB> "better battery life"
[22:04:37 CEST] <JEEB> like, I've seen quite a few users here throughout the years
[22:04:59 CEST] <JEEB> and the only reason people have been using faac is due to them reading an old tutorial from before fdk-aac was a thing
[22:05:06 CEST] Action: JEEB sighs
[22:29:19 CEST] <tp_> or they use a package manager to install ffmpeg
[23:39:33 CEST] <kepstin> people who used a package manager to install it used to get vo-aacenc :/
[00:00:00 CEST] --- Mon Apr 11 2016