John Riselvato

This is one of the most asked questions on google about FFMPEG and for a good reason, by default automatic compression is added depending on the filter. This is usually done to perform the filter quicker as the higher the quality, the longer the command completion time. 

Compression can be avoided using recommended codecs and setting quality. It is recommended that the H.264 aka libx264 encoder is used with most video formats. Libx264 is used entirely throughout this book. This assumes that your FFMPEG installation is configured with –enable-libx264.

The H.264 codec can be set as seen in the example below:

 $ ffmpeg -i input.mp4 -c:v libx264 output.mp4 

Another useful command is setting the constant rate factor (-crf) a rate control mode. The setting for a lossless output is 0, the lower the number the higher the quality but larger the file size. The default value is 23 while 51 is the worst quality. According to the documentation, values between 17-28 are “virtually lossless” visually but technically not. 

$ ffmpeg -i input.mp4 -c:v libx264 -crf 18 output.mp4

In addition to the previous two, selecting a preset can also increase quality but will increase file size. Setting the preset will determine which encoding speed and compression ratio. The slower the preset the longer the computation time. The presets are as follows:

• ultrafast 
• superfast 
• veryfast 
• faster 
• fast 
• medium – default 
• slow 
• slower 
• veryslow

For the highest video quality with all 3 settings are as follows:

 $ ffmpeg -i input.mp4 -c:v libx264 -crf 0 -preset veryslow output.mp4 

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Adding an additional audio source on top of the audio already in a video can be accomplished with the amix filter. This might be useful for adding background music to a commentary video as seen in the example below:

  $ ffmpeg -i input.mp4 -i input.mp3 -filter_complex "amix" -map 0:v -map 0:a -map 1:a output.mp4  

A common issue that comes up with amix is the volume of one input over powers the second, to solve this, setting the louder input volume down can resolve this. In the example below the volume of the first input audio [0:a] will be reduce 40%:

  $ ffmpeg -i input.mp4 -i input.mp3 -filter_complex [0:a]volume=0.40,amix -map 0:v -map 0:a -map 1:a -shortest output.mp4  

Now you have two audio streams playing in a single video. Quick and easy.

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On Ubuntu or Debian based linux operating systems using apt-get installed makes installation of FFMPEG is easy:

$ sudo apt-get update 
$ sudo apt-get install ffmpeg 

This will install the latest FFMPEG package but some linux users will want to install from source. FFMPEG requires compilation and installation of various codecs and libraries not bundled with FFMPEG source code.

Linux users who want to configure extra dependencies are required to install from source. Here’s an example of install dependencies, given from the FFMPEG wiki:

cd ~/ffmpeg_sources && \
wget -O ffmpeg-snapshot.tar.bz2 https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 && \
tar xjvf ffmpeg-snapshot.tar.bz2 && \
cd ffmpeg && \
PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
  --prefix="$HOME/ffmpeg_build" \
  --pkg-config-flags="--static" \
  --extra-cflags="-I$HOME/ffmpeg_build/include" \
  --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  --extra-libs="-lpthread -lm" \
  --bindir="$HOME/bin" \
  --enable-gpl \
  --enable-libaom \
  --enable-libass \
  --enable-libfdk-aac \
  --enable-libfreetype \
  --enable-libmp3lame \
  --enable-libopus \
  --enable-libvorbis \
  --enable-libvpx \
  --enable-libx264 \
  --enable-libx265 \
  --enable-nonfree && \
PATH="$HOME/bin:$PATH" make && \
make install && \
hash -r

Source: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu

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At this point FFMPEG should be installed on your operating system of choice and ready for use. To double check installation, it’s always good to view the current version installed. 

$ ffmpeg -version 

The output not only details the FFMPEG version but also the various configurations enabled and disabled as well as what codec are installed.

$ ffmpeg -version
ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
built with Apple clang version 11.0.0 (clang-1100.0.33.17)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libflite --enable-libzvbi --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack 
libavutil      56. 31.100 / 56. 31.100
libavcodec     58. 54.100 / 58. 54.100
libavformat    58. 29.100 / 58. 29.100
libavdevice    58.  8.100 / 58.  8.100
libavfilter     7. 57.100 /  7. 57.100
libavresample   4.  0.  0 /  4.  0.  0
libswscale      5.  5.100 /  5.  5.100
libswresample   3.  5.100 /  3.  5.100
libpostproc    55.  5.100 / 55.  5.100

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Now that you understand the importance of knowing which codecs your file formats should conform to. It is useful to have the ability to check the input file format before using FFMPEG commands. Luckily this functionality is native to FFMPEG, the command is as follows:

$ ffmpeg -i input.mp3
ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Input #0, mp3, from 'input.mp3':
  Metadata:
    title           : 1989 MAZDA FAMILIA // Car
    artist          : テレビCM
    track           : 16
    album           : Visual Signals
    date            : 2020
    encoder         : Lavf58.29.100
  Duration: 00:00:16.58, start: 0.023021, bitrate: 128 kb/s
    Stream #0:0: Audio: mp3, 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.54
    Side data:
      replaygain: track gain - 13.100000, track peak - unknown, album gain - unknown, album peak - unknown, 
At least one output file must be specified

After the installed FFMPEG version and configuration information the Input section details information like metadata, duration, encoding and more. In the above example it can be noted that input.mp3 is indeed an mp3 with a Lavc/Lavf encoder which I guess is the standard encoding for Bandcamp downloads.

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