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• How to change Chromuim alsa sound device [closed]

  25 mai 2020, par Edik Amperyan

  I need to run multiple browser windows at multiple alsa sound devices to record them using snd-aloop and ffmpeg in Ubuntu 18.04. I created three loopback devices :

  



  sudo modprobe snd-aloop enable=1,1,1 index=10,11,12

  



  and tried to run chromium-browser with —alsa-output-device flag with different values (such as device name, number, "plug :" + name) but it always uses default device (checking it using pavucontrol).

  



  Does this flag work ? Is there any other way to select sound device for application ? I can do it using pavucontrol GUI but I need to do it programmatically.

  



  Not working examples :

  



  chromium-browser https://youtu.be/Rb0UmrCXxVA?t=2744 --new-window --autoplay-policy=no-user-gesture-required --incognito --alsa-output-device=hw:11,1

  



  chromium-browser https://youtu.be/Rb0UmrCXxVA?t=2744 --new-window --autoplay-policy=no-user-gesture-required --incognito --alsa-output-device=plug:surround51

  



  chromium-browser https://youtu.be/y65necIJU2Y?t=1434 --new-window --autoplay-policy=no-user-gesture-required --incognito --alsa-output-device=Loopback_2

  


• FFmpeg eats all the memory and crash within a minute while recording chrome activity

  17 avril 2020, par Thomas Petit

  I'm using ffmpeg to capture the activity of chrome (thanks to chrome driver) and record it into an mp4 file. However, the memory consumed by ffmpeg is quickly blowing up, and after a minute or so my 8GB of memory get saturated and I have to reboot the PC.

  



  Unbuntu 16.0
ALSA loopback (installed by modprobe snd-aloop)
ffmpeg version 4.2.2-1ubuntu1 16.04.york0

  



  This is the command line to ffmpeg :

  



      ffmpeg -y -v info -f x11grab -draw_mouse 0 -r 30 -s 1280x720 -thread_queue_size 4096 
-i :0.0+0,0 -f alsa -thread_queue_size 4096 -i plug:cloop -acodec aac -strict -2 -ar 44100 
-c:v libx264 -preset veryfast -profile:v main -level 3.1 -pix_fmt yuv420p -r 30 
-crf 25 -g 60 -tune zerolatency -f mp4 file.mp4


  



  If I remove all the sound input (-i plug:cloop -acodec aac -strict -2 -ar 44100) then the memory is OK, stable, but the file generated can't be played with VLC or media player.

  



  The logs from ffmpeg looks normal to me :

  



  root$ ffmpeg -rtbufsize 15M -y -v info -f x11grab -draw_mouse 0 -r 30 -s 1280x720 -thread_queue_size 4096 -i :0.0+0,0 -f alsa -thread_queue_size 4096 -i hw:0 -acodec ac3_fixed -strict -2 -ar 44100 -c:v libx264 -preset veryfast -profile:v main -level 3.1 -pix_fmt yuv420p -r 30 -crf 25 -g 60 -tune zerolatency -f mp4 /tmp/recordings/stuff.mp4
ffmpeg version 2.8.15-0ubuntu0.16.04.1 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609
  configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
Input #0, x11grab, from ':0.0+0,0':
  Duration: N/A, start: 1587042601.295126, bitrate: N/A
    Stream #0:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1280x720, 30 fps, 30 tbr, 1000k tbn, 30 tbc
Guessed Channel Layout for  Input Stream #1.0 : stereo
Input #1, alsa, from 'hw:0':
  Duration: N/A, start: 1587042601.304179, bitrate: 1536 kb/s
    Stream #1:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
[libx264 @ 0x928ee0] using cpu capabilities: MMX2 SSE Cache64
[libx264 @ 0x928ee0] profile Main, level 3.1
Output #0, mpegts, to '/tmp/recordings/stuff.ts':
  Metadata:
    encoder         : Lavf56.40.101
    Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720, q=-1--1, 30 fps, 90k tbn, 30 tbc
    Metadata:
      encoder         : Lavc56.60.100 libx264
    Stream #0:1: Audio: ac3 (ac3_fixed), 44100 Hz, stereo, s16p, 192 kb/s
    Metadata:
      encoder         : Lavc56.60.100 ac3_fixed
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  Stream #1:0 -> #0:1 (pcm_s16le (native) -> ac3 (ac3_fixed))
Press [q] to stop, [?] for help
frame=    7 fps=0.0 q=21.0 size=       5kB time=00:00:00.23 bitrate= 180.5kbits/s frame=   15 fps= 15 q=21.0 size=      15kB time=00:00:00.50 bitrate= 240.6kbits/s frame=   21 fps= 14 q=21.0 size=      21kB time=00:00:00.70 bitrate= 249.2kbits/s frame=   28 fps= 14 q=21.0 size=      28kB time=00:00:00.93 bitrate= 246.5kbits/s frame=   35 fps= 13 q=21.0 size=      35kB time=00:00:01.16 bitrate= 244.9kbits/s frame=   42 fps= 13 q=21.0 size=      44kB time=00:00:01.40 bitrate= 260.0kbits/s frame=   49 fps= 13 q=21.0 size=      51kB time=00:00:01.63 bitrate= 256.9kbits/s frame=   56 fps= 13 q=21.0 size=      58kB time=00:00:01.86 bitrate= 254.6kbits/s frame=   63 fps= 13 q=22.0 size=      66kB time=00:00:02.10 bitrate= 255.7kbits/s frame=   70 fps= 13 q=21.0 size=      75kB time=00:00:02.33 bitrate= 263.0kbits/s frame=   77 fps= 13 q=21.0 size=      82kB time=00:00:02.56 bitrate= 261.3kbits/s frame=   79 fps= 13 q=21.0 Lsize=      85kB time=00:00:02.63 bitrate= 264.4kbits/s
video:8kB audio:61kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 22.604090%
[libx264 @ 0x928ee0] frame I:2     Avg QP: 5.10  size:   760
[libx264 @ 0x928ee0] frame P:77    Avg QP: 6.78  size:    88
[libx264 @ 0x928ee0] mb I  I16..4: 99.9%  0.0%  0.1%
[libx264 @ 0x928ee0] mb P  I16..4:  0.0%  0.0%  0.0%  P16..4:  0.0%  0.0%  0.0%  0.0%  0.0%    skip:100.0%
[libx264 @ 0x928ee0] coded y,uvDC,uvAC intra: 0.0% 0.1% 0.0% inter: 0.0% 0.0% 0.0%
[libx264 @ 0x928ee0] i16 v,h,dc,p: 91%  0%  9%  0%
[libx264 @ 0x928ee0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu:  0%  0% 100%  0%  0%  0%  0%  0%  0%
[libx264 @ 0x928ee0] i8c dc,h,v,p: 98%  2%  0%  0%
[libx264 @ 0x928ee0] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x928ee0] kb/s:25.23
Exiting normally, received signal 2.


  



  This is my .asoundrc file.

  



  # playback PCM device: using loopback subdevice 0,0
pcm.amix {
  type dmix
  ipc_key 219345
  slave.pcm "hw:Loopback,0,0"
}

# capture PCM device: using loopback subdevice 0,1
pcm.asnoop {
  type dsnoop
  ipc_key 219346
  slave.pcm "hw:Loopback,0,1"
}

# duplex device combining our PCM devices defined above
pcm.aduplex {
  type asym
  playback.pcm "amix"
  capture.pcm "asnoop"
}

# ------------------------------------------------------
# for jack alsa_in and alsa_out: looped-back signal at other ends
pcm.ploop {
  type plug
  slave.pcm "hw:Loopback,1,1"
}

pcm.cloop {
  type dsnoop
  ipc_key 219348
  slave.pcm "hw:Loopback,1,0"
}

# ------------------------------------------------------
# default device

pcm.!default {
  type plug
  slave.pcm "aduplex"
}


  



  I'm not sure how to debug this kind of issue, any idea why the memory blow up so fast like that ?

  


• Decode audio file with missing header

  21 mai 2020, par Eyah

  I have a audio file (according to ffmpeg, it's a MP3) and I tried to decode it with ffmpeg but it's not working. I tried to repair the file with ffmpeg before I convert it but it didn't work either.

  



  Here's the hexadecimal data representation :

  



  048c 63b8 8017 0020 3ac1 7b96 a4a3 87bf
be1f 08ff e102 ba51 d85b 1485 3ec7 ff32
78c1 7726 6bbc ef67 2e7d 7672 fdd8 7df9
e1dd aff7 87e7 e58f 7edf bf7c f654 d5ee
f6f7 dc4b 299d fd83 f3db 06aa 65a6 edb3
7ac7 f9ef 659b 9d9f 3fad ddb6 7a17 38ad
0341 2e47 f643 83b6 cbd4 7343 9bf3 e4bf
d8e8 777e 3a78 6ae5 ef14 7590 df92 d3ed
a472 f52b 9afd eab1 b360 e584 9db8 f99d
d863 d8b2 73de edbb 7513 e7d7 0dec ee1d
6bc6 bf47 b0f4 92f7 46bf 30b5 dafd 2efb
c7c1 aa05 dff1 e7c7 6d71 6fc8 973a e6d5
d3a7 2d1b ebb3 d621 e8cc cfe1 f978 d745


  



  Would someone have an idea on how i could decode it please ?