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• fixing (with ffmpeg) the chrominance position on a video after capturing

  23 mai 2016, par APLU

  I’m trying to convert some video from VHS to digital using an (old) video capture card (and obviously an old VHS player). Due to the input from my video capture card and the output available from the VHS, I have no other choice than capture with an S-Video cable to a computer.

  Almost everything works except a little mis-synchronization between chroma and luma which do not happen on TV.

  For example, in the original video, I have something like that :
  

  After capturing the video looks like this :
  

  As you may see, there is a little desynchronization of the chroma with the luma channel (I will say about 10 lines errors).

  I’m capturing with ffmpeg on a Linux system with the following commands :

  $ v4lctl setnorm PAL-BG

  $ v4lctl setinput S-video

  $ ffmpeg -y -f alsa -ac 2 -i pulse -f video4linux2 -i /dev/video0 -c:a pcm_s16le -vcodec rawvideo -t $duration -r 25 -loglevel error -stats /tmp/tmp.mkv

  I tried other input norm in v4l, tried an other VHS player, tried an other conversion cable from SCART to S-Video but it didn’t change anything,

  My question is simple : is there a way to fix this with a post-processing video filter in ffmpeg ?

  I already looked at the long list of video filter available in ffmpeg but I didn’t find anything.

  Also, please note I can’t apply filter during the capture commands (old capture cards, old cpu, ..), this is why I capture in rawvideo and native audio. When the capture is done I convert the video/audio into h264/vorbis, at this step I can apply as much as audio/video filtering needed (even if it include extracting chroma & luma to new files, fixing and merging again).

  Thanks !

• Extract audio file from a webm file using ffmpeg

  29 janvier 2016, par Hemant Kumar

  I am recording an audio using webRTC from firefox browser and then creating a file which is in webm format. Creating a webm is not my choice as webRTC automatically record on it that format. Its mimeType is "video/webm". I want to extract audio from this file in an mp3 file.

  I am using following command :

  "ffmpeg -i {$audioFile} -vn -ar 44100 -ac 2 -ab 192 -f mp3 -write_xing 0 {$outPutFile}"

  Its working on my localhost PC but unable to create a valid file on server.

  I am using ffmpeg version 2.4 on my localhost while 2.2 on my server.

  Here’s an input command :

  ffmpeg -i /26k20y1o4aj531k21syq.webm -vn -ar 44100 -ac 2 -ab 192 -f mp3 -write_xing 0 /26k20y1o4aj531k21syq_1.mp3

  Output on console on server :

  ffmpeg version 2.2.5 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Aug  1 2014 09:24:02 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libtheora --enable-libx264 --enable-libvpx --enable-libmp3lame
  
    libavutil      52. 66.100 / 52. 66.100
  
    libavcodec     55. 52.102 / 55. 52.102
  
    libavformat    55. 33.100 / 55. 33.100
  
    libavdevice    55. 10.100 / 55. 10.100
  
    libavfilter     4.  2.100 /  4.  2.100
  
    libswscale      2.  5.102 /  2.  5.102
  
    libswresample   0. 18.100 /  0. 18.100
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, ogg, from '/26k20y1o4aj531k21syq.webm':
  
    Duration: 00:00:05.82, start: 0.000000, bitrate: 52 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono
  
      Metadata:
  
        ENCODER         : Mozilla43.0.4
  
  [abuffer @ 0x1d94660] Unable to parse option value "(null)" as sample format
  
      Last message repeated 1 times
  
  [abuffer @ 0x1d94660] Error setting option sample_fmt to value (null).
  
  [graph 0 input from stream 0:0 @ 0x1d92c80] Error applying options to the filter.
  
  Error opening filters!

  For the same, output on my localhost :

  ffmpeg version 2.4.3-1ubuntu1~trusty6 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Nov 22 2014 17:07:19 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --prefix=/usr --extra-version='1ubuntu1~trusty6' --build-suffix=-ffmpeg --toolchain=hardened --extra-cflags= --extra-cxxflags= --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-fontconfig --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --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-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-opengl --enable-x11grab --enable-libxvid --enable-libx265 --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libx264 --enable-libsoxr --enable-openal --enable-libopencv
  
    libavutil      54.  7.100 / 54.  7.100
  
    libavcodec     56.  1.100 / 56.  1.100
  
    libavformat    56.  4.101 / 56.  4.101
  
    libavdevice    56.  0.100 / 56.  0.100
  
    libavfilter     5.  1.100 /  5.  1.100
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  0.100 /  3.  0.100
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  0.100 / 53.  0.100
  
  Input #0, ogg, from '/26k20y1o4aj531k21syq.webm':
  
    Duration: 00:00:05.82, start: 0.000000, bitrate: 52 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
  
      Metadata:
  
        ENCODER         : Mozilla43.0.4
  
  [libmp3lame @ 0x172cc20] Bitrate 192 is extremely low, maybe you mean 192k
  
  The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
  
  Output #0, mp3, to '/26k20y1o4aj531k21syq_3.mp3':
  
    Metadata:
  
      TSSE            : Lavf56.4.101
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp, 0 kb/s
  
      Metadata:
  
        encoder         : Lavc56.1.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (opus (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  size=      91kB time=00:00:05.81 bitrate= 128.8kbits/s    
  
  video:0kB audio:91kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.046998%

  There’s one more problem, On live server I can’t update ffmpeg library version because a live website is already working there. Can you please suggest / fix my command i.e.

  [ "ffmpeg -i {$audioFile} -vn -ar 44100 -ac 2 -ab 192 -f mp3 -write_xing 0 {$outPutFile}"  ]

  so that it create file perfectly.

  EDIT :
  Thanks for replying @Mulva. I tried your solution but the error is still the same. Here is the output console :

  ffmpeg version 2.2.5 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Aug  1 2014 09:24:02 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libtheora --enable-libx264 --enable-libvpx --enable-libmp3lame
  
    libavutil      52. 66.100 / 52. 66.100
  
    libavcodec     55. 52.102 / 55. 52.102
  
    libavformat    55. 33.100 / 55. 33.100
  
    libavdevice    55. 10.100 / 55. 10.100
  
    libavfilter     4.  2.100 /  4.  2.100
  
    libswscale      2.  5.102 /  2.  5.102
  
    libswresample   0. 18.100 /  0. 18.100
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, ogg, from '/26k20y1o4aj531k21syq.webm':
  
    Duration: 00:00:08.40, start: 0.000000, bitrate: 52 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono
  
      Metadata:
  
        ENCODER         : Mozilla43.0.4
  
  [abuffer @ 0x24257e0] Unable to parse option value "(null)" as sample format
  
      Last message repeated 1 times
  
  [abuffer @ 0x24257e0] Error setting option sample_fmt to value (null).
  
  [graph 0 input from stream 0:0 @ 0x24256c0] Error applying options to the filter.

  EDIT 2 :

  @Mulvya I tried your re-edited command but no it is not working as well.
  As my Question says the command run properly on my local server (on my own pc) and its running ffmpeg version 2.4.3. I was wondering if I can update the ffmpeg installed online. (Since there is no other way to do create this file ). How can i upgrade ffmpeg on my server. My server is running on ubuntu.

  I tried "sudo apt-get update" and "sudo apt-get install ffmpeg" (This command say ffmpeg is already updated) so I used command "sudo apt-get install ffmpeg 2.8.5"

  It gave me list of many codecs/library to add/update but today when i see ffmpeg version, nothing is changed. its is still showing me the same version with same list of library configured as previously. And yes the commands I tried earlier still are not working on server.

  Am i missing anything ? Anything I should do apart from upgrading ffmpeg ?
  How can i update existing ffmpeg software on server without removing it ?

• using ffmpeg to create a wavefile image from opus

  29 décembre 2015, par edwardsmarkf

  I have been trying to use ffmpeg to create a wavefile image from an opus file. so far i have found three different methods but cannot seem to determine which one is the best.

  The end result is hopefully to have a sound-wave that is only approx. 55px in height. The image will become part of a css background-image.

  Adapted from Generating a waveform using ffmpeg :

  ffmpeg -i file.opus -filter_complex
  "showwavespic,colorbalance=bs=0.5:gm=0.3:bh=-0.5,drawbox=x=(iw-w)/2:y=(ih-h)/2:w=iw:h=1:color=black@0.5"
  file.png

  which produces this image :
  

  Next, I found this one (and my favorite because of the simplicity) :

  ffmpeg -i test.opus -lavfi showwavespic=split_channels=1:s=1024x800 test.png

  And here is what that one looks like :

  

  Finally, this one from FFmpeg Wiki : Waveform, but it seems less efficient using a second utility (gnuplot) rather than just ffmpeg :

  ffmpeg -i file.opus -ac 1 -filter:a
  aresample=4000 -map 0:a -c:a pcm_s16le -f data - | \
  gnuplot -e "set
  terminal png size 525,050 ;set output
  ’file.png’ ;unset key ;unset tics ;unset border ; set
  lmargin 0 ;set rmargin 0 ;set tmargin 0 ;set bmargin 0 ; plot ’

  

  Option two is my favorite, but i dont like the margins on the top and bottom of the waveforms.

  Option three (using gnuplot) makes the best ’shaped’ image for our needs, since the initial spike in sound seems to make the rest almost too small to use (lines tend to almost disappear) when the image is sized at only 50 pixels high.

  Any suggestions how might best approach this ? I really understand very little about any of the options I see, except of course for the size. Note too i have 10’s of thousands to process, so naturally i want to make a wise choice at the very beginning.