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• Change input source based on intensity of audio using FFmpeg

  24 mai 2020, par Harry Blue

  I have script that allows me to encode an audio stream and an mp4 file with visual effects applied over the top.

  



  The effects are synced to the music and whilst the effects are simple, they work well.

  



  I am using ffmpeg, mediainfo, randomize-lines, bc.

  



  #!/bin/bash

#########################
## START CONFIGURATION ##
#########################

# path to video playlist.txt
playlist="playlist.txt";

# output frame size
outsize="1920x1080";

# frames per second
fps="60";

# video bitrate
bv="20M";

# audio bitrate
ba="96k";

# video codec
cv="h264";

# audio codec
ca="libfdk_aac";

# output container format
fmt="mp4";

# base frequency (Hz)
bfreq="20";

# end frequency (Hz)
efreq="1420";

#######################
## END CONFIGURATION ##
#######################

viz="volume=2,showcqt=s=1920x144:text=0:r=$fps:axis=0:basefreq=$bfreq:endfreq=$efreq:count=15:sono_g=4:bar_g=4:bar_v=35:sono_h=144:sono_v=bar_v*a_weighting(f):tc=0.1,rotate=1200*sin(200*PI/200*t):ow=24:oh=24:c=none,scale=$outsize,setsar=1/1[viz];[1:v]scale=$outsize,setsar=1/1[vid1];[viz][vid1]blend=all_mode=heat:shortest=1:repeatlast=0,hue="H="2*PI*t/420""";
enc="-s $outsize -c:a $ca -b:a $ba -c:v $cv -preset ultrafast -b:v $bv -profile:v high -level 4.2 -g "$(bc <<< $fps*2)" -bf 2 -x264opts keyint="$(bc <<< $fps*2)":min-keyint="$(bc <<< $fps*2)":8x8dct=1 -pix_fmt yuv420p -r $fps";
ffmpeg -hide_banner -i "$1" -r $fps -stream_loop -1 -i one.mp4 -filter_complex $viz $enc -f $fmt "$2";


  



  This can be invoked using something like

  



  ./stream.sh http://ic2255.c471.fast-serv.com/autodj file.mp4

  



  What I would like to do is use different input videos based on the intesity of the music, in the same way the effects are applied.

  



  I created a playlist.txt file that contains my mp4 video input files

  



  one.mp4
two.mp4


  



  and made a change to this like

  



  ffmpeg -hide_banner -i "$1" -r $fps -stream_loop -1 -i one.mp4 -filter_complex $viz $enc -f $fmt "$2";


  



  So it uses a random file

  



  randfile="$(cat "$playlist" | rl | head -1)";
ffmpeg -hide_banner -i "$1" -r $fps -stream_loop -1 -i "$randfile" -filter_complex $viz $enc -f $fmt "$2";


  



  However this only works for the initial input, it does not yet change the input, only loops that mp4.

  



  Is this possible ?

  


• ffmpeg HE-AAC 'unsupported codec' despite being compiled with support

  12 mai 2020, par Sebastian

  Lack of default support of AAC for ffmpeg is really annoying :

  



  My ffmpeg version :

  



  ffmpeg version git-2020-05-02-0d81edc Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD-0d81edc_1 --enable-shared --cc=clang --host-cflags=-fno-stack-check --host-ldflags= --enable-gpl --enable-libaom --enable-libmp3lame --enable-libopus --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --disable-libjack --disable-indev=jack --enable-opencl --enable-videotoolbox --disable-htmlpages --enable-libbluray --enable-libfdk-aac --enable-libopenh264 --enable-libopenjpeg --enable-librsvg --enable-libspeex --enable-libsrt --enable-libtwolame --enable-libwavpack --enable-libwebp --enable-libxvid --enable-nonfree --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb


  



  I have several segment.ts files in edit.txt and I try to concatenate and convert to mpg :
ffmpeg -f concat -i edit.txt -c copy   output.mpg

  



  this does not work

  



  Input #0, concat, from 'edit.txt':
  Duration: N/A, start: 0.000000, bitrate: 121 kb/s
    Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720, 30 fps, 29.97 tbr, 90k tbn, 60 tbc
    Stream #0:1: Audio: aac (HE-AAC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 121 kb/s
File 'output.mpg' already exists. Overwrite? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (aac (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
[mpeg @ 0x7f8e44807e00] VBV buffer size not set, using default size of 230KB
If you want the mpeg file to be compliant to some specification
Like DVD, VCD or others, make sure you set the correct buffer size
[mpeg @ 0x7f8e44807e00] Unsupported audio codec. Must be one of mp1, mp2, mp3, 16-bit pcm_dvd, pcm_s16be, ac3 or dts.
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Error initializing output stream 0:1 --
Conversion failed!


  



  so I tried with ffmpeg -f concat -i edit.txt -c:v copy -c:a libfdk_aac  output.mpg -> same effect.

  



  Why fraunhoffer codec exists and is mentioned on FFMPEG site if it is not supported ?
Is it any way to do this ?

  


• Two variables for ffmpeg ; use for loop ?

  16 juillet 2020, par Zorba

  I would like to silence sequences in a number of audio files with ffmpeg, which requires to give two variables, the starting point and the end point.

  



  It works nice when you add the two variables in a script like so

  



  
echo "which file do you want to silence"
read filename

echo "When does the muted period start in seconds"
read A

echo "When does the muted period end in seconds"
read B

ffmpeg -i $filename -af "volume=enable='between(t,$A,$B)':volume=0" output_silenced.mp3


  



  I have read all the "similar questions" that come up in a search for "use two variables in bash" and similar search terms, but I can't get my head around this.

  



  I put the variables in two files, with new lines for each number, and tried to call the files in a for loop, which did not work. I also tried to call them in a for loop like so

  



  echo "which file do you want to silence"
read filename

for A in 22 33 57; do

for B in 27 45.5 58.5; do

ffmpeg -i $filename -af "volume=enable='between(t,$A,$B)':volume=0" output_silenced.mp3

mv output_silenced.mp3 filename.mp3
done
done


  



  but the output sounds just like the original input audio !

  



  Can anybody help me out with this ? I have to partly silence many files and it already takes forever to figure out all the start and end moments, and it would be great to get through with this a bit easier !

  



  Thanks a lot in advance !