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• Visual Studio LNK2001 Error, despite using extern "C" for ffmpeg libraries in C++ [duplicate]

  1er février 2020, par Faizan Cassim

  This is what it looks like :

  ’’’

  extern "C"
  
  {
  
  #include "libavcodec/avcodec.h"
  
  #include "libavformat/avformat.h"
  
  #include "libavutil/avutil.h"
  
  };

  ’’’

  But I still get these errors when using the FFmpeg libraries in Visual Studio 2019

  1>AudioFile.obj : error LNK2001: unresolved external symbol avformat_open_input
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_read_frame
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_free
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_get_sample_fmt_name
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol avformat_close_input
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_init_packet
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol avcodec_receive_frame
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol avcodec_open2
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_sample_fmt_is_planar
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol avcodec_close
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_get_bytes_per_sample
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_packet_unref
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol avformat_find_stream_info
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_find_best_stream
  
  1>AudioFile.obj : error LNK2001: unresolved external symbol av_frame_alloc

  I would be greatfull to anyone who could help me resolve this. I am using C++ for Windows.

• 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 ?

  


• Rotate Android video for display on web

  16 juin 2017, par Kim

  I’m building a website. I’m trying to load my first video onto it. This mp4 video is in portrait orientation and was recorded on an Android device.

  When I load this mp4 onto my website, it plays perfectly (in portrait orientation) on Chrome ; however, it plays in landscape orientation on Firefox & IE. I realise that this is because Chrome is using the rotation metadata that is stored within the mp4 to play the video in the correct orientation, while FF & IE don’t use this metadata.

  I’ve tried to correct this issue using FFMPEG, and various combinations of the Transpose, Metadata & Rotate commands. I am using MediaInfo to view the metadata associated with the files that I produce using FFMPEG.

  My problem is :
  1) If I use Transpose=1, then on FF & IE & Windows Media Player, the resulting video is actually rotated through 180 degrees, and the metadata shows the file width and depth to be the same as the original file.
  2) If I use -metadata:s:v rotate=0 (I’ve also tried rotate=90), then MediaInfo tells me that the resulting files width & depth are now portrait, but FF & IE & WMP will not play the resulting file. WMP plays the sound but no image, and FF gives an error message "Error Loading This Resource".

  What ffmpeg command do I need to use to convert a portrait MP4 recorded an android device into an mp4 that will play in the correct orientation on Firefox ?

  Thanks in advance