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• ffmpeg - How to filter video when record Window Application in the same time

  16 mai 2018, par user3181176

  I have a problem when record Window Application and filter (overlay, audio mixing...) with others filter video.

  This code work perfectly :

  ffmpeg -rtbufsize 1500M -f dshow -i audio="virtual-audio-capturer" -f gdigrab -framerate 30 -draw_mouse 0 -i title="Main" -vf crop=850:480:156:100 -pix_fmt yuv420p -profile:v baseline -y ok.mp4   (With Main is my Application)

  But this code bellow doesn’t :

  ffmpeg  -rtbufsize 1500M -f dshow -y -i audio="virtual-audio-capturer" -f gdigrab -framerate 30 -draw_mouse 0 -i title="Main" -vf crop=850:480:156:100 -pix_fmt yuv420p -profile:v baseline -stream_loop 999 -i "filter/filter.mp4" -filter_complex "[2:v]scale=385:216, setdar=dar=16/9[v1]; [2:v]scale=385:216, setdar=dar=16/9[v2]; [v1][v2]blend=shortest=1:all_opacity=1[v3]; movie=filter/nds_bg.mp4:loop=999,setpts=N/(FRAME_RATE*TB), scale=640:360[v4] ;[v4][v3]overlay=shortest=1:x=20:y=130;[2:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,asetrate=8.5/10*44100,atempo=10/8.5,lowpass=f=2500,highpass=f=400,volume
  
  =3,bass=g=-30,equalizer=f=10.5:width_type=o:width=1:g=-30, equalizer=f=31.5:width_type=o:width=1:g=-
  
  30,equalizer=f=63:width_type=o:width=1:g=-10, equalizer=f=125:width_type=o:width=1:g=-20,equalizer=f=250:width_type=o:width=1:g=-1.5,equalizer=f=500:width_type=o:width=1:g=-20,equalizer=f=1000:width_type=o:width=1:g=-20,equalizer=f=8000:width_type=o:width=3:g=1,equalizer=f=16000:width_type=o:width=3:g=1" -vcodec libx264 -pix_fmt yuv420p -r 26 -g 30 -b:30 
  
  800k -shortest -acodec libmp3lame -b:a 128k -preset:v ultrafast -ar 44100 -f flv -bufsize 3000k -s 640x360 out.mp4

  I think this code need to output to video first

  ffmpeg -rtbufsize 1500M -f dshow -y -i audio="virtual-audio-capturer" -f gdigrab -framerate 30 -draw_mouse 0 -i title="Main" -vf crop=850:480:156:100 -pix_fmt yuv420p -profile:v baseline

  Please show me how to merge these code or any solution. Thank you so much !

• Different namefiles rather those provided using ffmpeg and bash

  13 mars 2020, par takeTo

  I have this code :

  input=""
  
  touch file.tmp
  
  echo "Drag and drop destination folder, and press enter:"
  
  
  
  read destination_folder
  
  folder="$destination_folder/"
  
  while [ "$input" !=  "e" ]
  
  do
  
    echo ""
  
    echo "Drag and drop file, and press enter (or press e and enter to end):"
  
    read input
  
    if [ "$input" != "e" ]
  
    then
  
      echo "$input"  | tee -a file.tmp >/dev/null
  
    fi
  
  done
  
  echo ""
  
  echo "Processing..."
  
  echo ""
  
  input="file.tmp"
  
  while IFS= read line
  
  do
  
    FILE="$line"
  
    basenames=$(basename "$FILE")
  
    echo "$basenames"
  
    folders="$folder$basenames"
  
    echo "$folders"
  
    ffmpeg -y -loglevel quiet -hwaccel vdpau -i "$line" -filter:a "crossfeed=strength=0.3:range=0.5","extrastereo=m=0.0:c=false","highpass=f=80:t=q:w=0.8","afftdn=nr=8.5:nt=c:bn=125|250|520|745|920","deesser=i=0.12","equalizer=f=110:t=q:w=1.12:g=2","equalizer=f=285:t=q:w=3:g=-3","equalizer=f=500:t=q:w=2.72:g=-4","equalizer=f=1030:t=q:w=1.95:g=2","highshelf=g=3:f=4250:t=q:w=0.6","loudnorm=i=-16:lra=14:tp=-0.8:dual_mono=true" -c:a aac -ab 320k -ar 48000 -c:v libx264 -x264-params b=10000:keyint_min=15:coder=ac:b-bias=2 -preset slow "$folders"
  
  done <"$input"
  
  echo "Done."

  Running the script on this directory/files structure :

  /done/1.mp4
  
  /done/2.mp4
  
  /done/3.mp4

  The output is :

  1.mp4
  
  3.mp4

  Why I’m getting as output only 1.mp4 and 3.mp4 ? And, why "2.mp4" basename is echoed as ".mp4" and not as "2.mp4" ?

  EDIT : Now all files are processed normally, but i’m getting processed also .mp4 files for each file processed.

• Add, remove and tune filters on running ffmpeg

  28 mai 2023, par Turmund

  Preface

  


  I have been fiddling around with ffmpeg and ffplay using command line adding and tuning filters and effects.

  


  It quickly becomes rather tiresome to

  


  

  1. start playback with some audio file

  


  2. stop

  


  3. modify command

  


  4. back to 1.

  


  


  When for example fine-tuning noise reduction or adding effects and equalizer.

  


  I have played around with using zmq to tune filters by executing commands in a different terminal, but this also becomes somewhat cumbersome.

  


  I want some interface where I can add, remove and tune filters during runtime / while listening to the changes taking effect.

  


  FFMPEG

  


  I use filter to mean effect / filter from here on out. For example afftdn, rubberband, ...

  


  ffmpeg is somewhat intimidating. It's powerful but also complex, at least when starting to dig into it. :0

  


  Looking at the library and examples I am looking at API example for audio decoding and filtering, - which at least at first looks promising as a starting platter.

  


  Output

  


  I imagine it would be best to have multiple sinks or some container with multiple audio tracks :

  


  

  1. Raw audio

  


  2. Audio with effects applied

  


  


  Optionally :

  


  

  1. Raw audio

  


  2. Audio with all filters

  


  3. Audio with filter group 1

  


  4. Audio with filter group 2

  


  5. ... etc.

  


  


  ---

  


  Processing

  


  I imagine the routine would have to be something like :

  


  

  1. Read packet from stream/file/url

  


  2. Unpack the sample

  


  3. Copy / duplicate sample for each filter group / or one for filters

  


  4. Apply filter(s) to these “effect sample” (s)

  


  5. Write raw audio, filtered audio 1, filtered audio 2, filtered audio N, ... to out

  


  


  Or for step 3 - 5 (as one would only be listening to one track at a time (But this is perhaps not the best if one decide to jump back / forth in the audio stream) :

  


  

  3. Apply currently active filter(s)

  


  4. Write raw audio, filtered audio to out

  


  


  Simultaneously one would read and check changes to filters by some interface. I.e. input :

  


  afftdn=rf=-20:nr=20


  


  then, if afftdn is not present in filters add it, else set new values.

  


  Idea is to output "raw-audio". I.e. used in a sampling and tuning phase - then produce a line with filter-options one can use with the ffmpeg-tool to process the audio files once satisfied.

  


  Questions section

  


  

  • Does something similar exist ?

  


  


  General :

  


  

  • Does this seem like a way to do it and use the ffmpeg library ?

    

    • Can one add, remove and change filter values during runtime or do one have to re-initialize the entire stream for each added / removed filter etc ?

    


    • Is the “Processing” part sound ?

    


    


  


  • Would using a container that supports multiple audio tracks be the likely best solution ? E.g. mp4.

    

    • Any container preferred over others ?

    


    • Any drawbacks (i.e. jumping back / forth in the stream)

    


    


  


  


  ---

  


  Sub-note

  


  Dream is to have a Arduino interfacing with this program where I use physical rotary switches, incremental rotary encoders, buttons and whistles. Tuning the various options for the filters using physical knobs. But at first I need some working sample where I use FIFO or what ever to test ffmpeg itself.