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ffmpeg

ffmpeg is a complete, cross-platform solution to record, convert and stream audio and video.

This is my personal and ever growing collection of useful ffmpeg commands.

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ffmpeg-commands

Video Size Syntax Overview

Audio Filter Documentation

ffmpeg by example

ffmpeg command collection

Get File Information

ffmpeg -i video.avi

ffmpeg -i audio.mp3

Wav to Mp3

ffmpeg -i input_file.wav -write_id3v1 1 -id3v2_version 3 -dither_method rectangular -out_sample_rate 48k -qscale:a 1 output_file.mp3

ffmpeg -i input.mp3 output.wav
  • ffmpeg starts the command
  • -i input_file path and name of the input file
  • -write_id3v1 1 This will write metadata to an ID3v1 tag at the head of the file, assuming you’ve embedded metadata into the WAV file.
  • -id3v2_version 3 This will write metadata to an ID3v2.3 tag at the tail of the file, assuming you’ve embedded metadata into the WAV file.
  • -dither_method rectangular Dither makes sure you don’t unnecessarily truncate the dynamic range of your audio.
  • -out_sample_rate 48k Sets the audio sampling frequency to 48 kHz. This can be omitted to use the same sampling frequency as the input.
  • -qscale:a 1 This sets the encoder to use a constant quality with a variable bitrate of between 190-250kbit/s. If you would prefer to use a constant bitrate, this could be replaced with -b:a 320k to set to the maximum bitrate allowed by the MP3 format. For more detailed discussion on variable vs constant bitrates see here.
  • output_file path and name of the output file

WAV to AAC/MP4

ffmpeg -i input_file.wav -c:a aac -b:a 128k -dither_method rectangular -ar 44100 output_file.mp4
  • ffmpeg starts the command
  • -i input_file path and name of the input file
  • -c:a aac sets the audio codec to AAC
  • -b:a 128k sets the bitrate of the audio to 128k
  • -dither_method rectangular Dither makes sure you don’t unnecessarily truncate the dynamic range of your audio.
  • -ar 44100 sets the audio sampling frequency to 44100 Hz, or 44.1 kHz, or “CD quality”
  • output_file path and name of the output file

Mix a Video With a Sound File

ffmpeg -i sound.wav -i original_video.avi final_video.mpg

Batch Conversion .mp4 to WAV

cd to directory, then:

for i in *.mp4; do ffmpeg -i "$i" -acodec pcm_s16le -ar 48000 "${i%.*}.wav"; done

Batch Conversion .wav to .mp3

for i in *.wav; do ffmpeg -i "$i" -dither_method rectangular -qscale:a 1 -ar 48000 "${i%.*}.mp3"; done

(keeps the file names and creates new files in the same directory)

WAV sample rate conversion

ffmpeg -i sound.wav -ar 44100 sound-44100.wav

Extract audio from an AV file

ffmpeg -i *input_file* -c:a copy -vn *output_file*
  • ffmpeg starts the command
  • -i input_file path, name and extension of the input file
  • -c:a copy re-encodes using the same audio codec
  • -vn no video stream
  • output_file path, name and extension of the output file

Create an allround Visualizing Video from Audio File

ffmpeg -i input.mp3 -filter_complex \
"[0:a]avectorscope=s=640x518[left]; \
 [0:a]showspectrum=mode=separate:color=intensity:scale=cbrt:s=640x518[right]; \
 [0:a]showwaves=s=1280x202:mode=line[bottom]; \
 [left][right]hstack[top]; \
 [top][bottom]vstack,drawtext=fontfile=/Library/Fonts/Arial.ttf:fontcolor=white:x=10:y=10:text='\"Song Title\" by Artist'[out]" \
-map "[out]" -map 0:a -c:v libx264 -preset fast -crf 18 -c:a copy output.mkv
  • input.mp3 can also be .wav or maybe other formats as well
  • output must be .mkv
  • -preset fast can be exchanged in:
  • ultrafast
  • superfast
  • veryfast
  • faster
  • fast
  • medium – default preset
  • slow
  • slower
  • veryslow

Audio Waveform Video from Audio File

ffmpeg -i input.wav -filter_complex \
"[0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" \
-map "[v]" -map 0:a output.mp4

ffplay -f lavfi "amovie=input.m4a, asplit [a][out1]; [a] showwaves [out0]"

Audio Spectrum Video from Audio File

ffmpeg -i input.wav -filter_complex \
"[0:a]showspectrum=s=1280x720:color=rainbow:start=0:stop=2500:legend=enabled,format=yuv420p[v]" \
-map "[v]" -map 0:a output.mp4

some Options for showspectrum

  • slide=replace or =scroll =fullframe =rscroll (How spectrum slides along the window)

  • mode=combined or =separate

  • color=channel or =rainbow =magma =green (different color schemes)

  • scale=sqrt or =lin =log
  • orientation=vertical or =horizontal
  • start=0 set start Frequency for Display
  • stop=0 set stop frequency for display
  • legend=disabled or =enabled (show legend in video)
ffplay -f lavfi "amovie=input.wav, asplit [a][out1]; \
[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]"

Audio-Spectrum Picture from Audio File

ffmpeg -i input.wav -lavfi showspectrumpic=s=hd720 output.jpg



ffmpeg -i /Users/tobias.hartmann/Documents/UniSiegen/Lehrprobe/Apotos_Björk.wav -lavfi showspectrumpic=s=hd720 /Users/tobias.hartmann/Documents/UniSiegen/Lehrprobe/Apotos_Björk.jpg
  • input can also be other files

Audio Vector Scope Video from Audio File

ffmpeg -i input.wav -filter_complex \
"[0:a]avectorscope=s=1280x720:r=25:draw=dot:mode=lissajous,format=yuv420p[v]" \
-map "[v]" -map 0:a output.mp4

some Options for vectorscope

  • s=1024X768 any size for video output
  • draw=dot or line
  • mode=lissajous or polar
  • scale=lin or log (amplitude scale of audio samples)
  • rc=40:gc=160:bc=80:ac=255 (red, green, blue, alpha contrast DEFAULTS. Range is [0, 255])
  • rf=15:gf=10:bf=5:af=5 (red, green, blue, alpha fade DEFAULTS. Range is [0,255])
ffplay -f lavfi "amovie=input.wav, asplit [a][out1]; \
[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]"

Audio Frequency-Spectrum Video from Audio File

ffmpeg -i input.wav -filter_complex \
"[0:a]showfreqs=mode=line:fscale=log:win_size=1024:win_func=hanning,format=yuv420p[v]" \
-map "[v]" -map 0:a output.mp4

some Options for showfreqs

  • mode=bar Options are =line =dot
  • ascale=log or =lin (amplitude scale)
  • fscale=linor =log (frequency scale)

  • win_size=2048 Range is [16, 65536]

  • win_func=hanning or =rect =sine `=gauss``

  • cmode=combined or =seperated (channel display mode)
  • colors=Azure|Lime (set colors. Check ffmpeg -colors for available color names)
ffplay -f lavfi "amovie=input.mp4, asplit [a][out1]; [a]  showfreqs=mode=line:fscale=log [out0]"

Video from Spatial Relationship between two Audio Channels

ffmpeg -i input.wav -filter_complex \
"[0:a]showspatial=s=1024x768:win_size=2048,format=yuv420p[v]" \
-map "[v]" -map 0:a output.mp4

Audio Volume Video from Audio File

ffmpeg -i input.wav -filter_complex \
"[0:a]showvolume=f=1:b=4:w=720:h=68,format=yuv420p[vid]" \
-map "[vid]" -map 0:a output.mp4

ffplay -f lavfi "amovie=input.mka, asplit [a][out1]; [a] showvolume=f=255:b=4:w=720:h=68 [out0]"

Video to Analyse Musical Notes from Audio File

ffmpeg -i input.wav -filter_complex \
"[0:a]showcqt,format=yuv420p[v]" \
-map "[v]" -map 0:a output.mp4

ffplay -f lavfi "amovie=input.mp4, asplit [a][out1]; [a] showcqt [out0]"

Volume Histogram Video from Audio File

ffmpeg -i input.wav -filter_complex \
"[0:a]ahistogram,format=yuv420p[v]" \
-map "[v]" -map 0:a output.mp4

ffplay -f lavfi "amovie=input.flac, asplit [a][out1]; [a] ahistogram [out0]"

VR 180 Format für Google Video

Verwendeter Code 

ffmpeg -i /SOURCE-FILELOCATION/SOURCE-FILENAME -c:v libx264 -preset fast -crf 18 -x264-params mvrange=511 -maxrate 120M -bufsize 25M -vf "scale=5120x2560:in_range=full:out_range=full:in_color_matrix=bt709:out_color_matrix=bt709" -pix_fmt yuv420p -c:a aac -b:a 160k -r 29.97 -movflags faststart /TARGET-FILELOCATION/TARGET-FILENAME

von google vorgeschlagenes Script:

ffmpeg -i /SOURCE-FILELOCATION/SOURCE-FILENAME -c:v libx264 -preset fast -crf 18 -x264-params mvrange=511 -maxrate 120M -bufsize 25M -vf "scale=5120x2560:out_range=full:out_color_matrix=bt709" -pix_fmt yuv420p -c:a aac -b:a 160k -r 30 -movflags faststart /TARGET-FILELOCATION/TARGET-FILENAME

Convert Video with Audio for youtube Upload

Video conversion with audio stream copying:

ffmpeg -i input.avi -c:v libx264 -preset slow -crf 18 -c:a copy -pix_fmt -movflags +faststart yuv420p output.mkv
  • add for youtube faststart: -movflags +faststart

Video conversion with Audio reencoding (AAC):

ffmpeg -i input.mov -c:v libx264 -preset slow -crf 18 -c:a aac -b:a 192k -pix_fmt yuv420p output.mkv

Create Video from still image and audio file:

ffmpeg -loop 1 -framerate 2 -i input.png -i audio.m4a -c:v libx264 -preset medium -tune stillimage -crf 18 -c:a copy -shortest -pix_fmt yuv420p output.mkv

ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut

Convert Videos for Web Use

ffmpeg -i input.mp4 -vcodec libx264 -crf 23 -preset veryfast -acodec aac -movflags +faststart output.mp4
  • i input.mp4: specifies the input file
  • vcodec libx264: specifies the video codec to be used (libx264)
  • crf 23: sets the Constant Rate Factor (CRF) which influences the quality of the video (lower is better quality but larger file size, 23 is a good middle ground)
  • preset veryfast: sets the encoding speed. Slower encoding provides better compression (smaller file size) and can give better quality for the same filesize. The options are: ultrafast, superfast, veryfast, faster, fast, medium, slow, slower, veryslow. 'Veryfast' is a good middle ground.
  • acodec aac: specifies the audio codec (AAC)
  • movflags +faststart: relocates some of the video data to the beginning of the file, which allows the video to start playing before it is completely downloaded by the client.

Batch conversion sh script

for file in /path/to/your/directory/*.m4v
do
  echo "Processing $file"
  ffmpeg -i "$file" -vcodec libx264 -crf 23 -preset fast -acodec aac -movflags +faststart "${file%.mp4}_output.mp4"
done

Put this script in the folder containing files. cd to this directory.

This script will take every .m4v file in the specified directory, and create a new converted file with _output appended to the original filename. Make sure to replace /path/to/your/directory/ with the actual path to the directory containing the video files you want to convert.

This script should be put in a .sh file, and you need to ensure the file is executable. You can make it executable by running chmod +x script.sh in the terminal, where script.sh is the name of your script. Then you can run the script using ./script.sh.

Create looped Insta Video from Picture and audio file

ffmpeg -loop 1 -t 20 -framerate 30 -i input.png -i input.mp3 -vf "scale=1080:1080:force_original_aspect_ratio=decrease,pad=1080:1080:(ow-iw)/2:(oh-ih)/2" -c:v libx264 -c:a aac -strict experimental -shortest output.mp4

Cut a video

ffmpeg -ss 00:04:40.00 -i input.mp4 output.mp4

Cut last 50 Seconds of a file

ffmpeg -i input.mp4 -t 50 -ss $(ffprobe -v error -show_entries format=duration -of default=noprint_wrappers=1:nokey=1 input.mp4 -sexagesimal | awk -F: '{ print $1 ":" $2 ":" $3 - 50 }') last50_new_output.mp4

ffmpeg -ss $(ffmpeg -i input.mp4 2>&1 | grep "Duration" | cut -d ' ' -f 4 | sed s/,// | awk -F: '{ print $1*3600 + $2*60 + $3 - 50 }') -i Anna_3D_vid_compressed.mp4 -c copy last50_output.mp4

Change Video Resolution

ffmpeg -i input.mov -vf scale=1920:1080 -c:v prores_ks -profile:v 3 output.mov

Batch Conversion to change video resolution:

for f in *.mov; do ffmpeg -i "$f" -vf scale=1920:1080 -c:v prores_ks -profile:v 3 "$(basename "${f}" .mov)_LoRes.mov"; done

Spoken word / voice processing

see: tratt.net/laurie/blog/2024/recording_and_processing_spoken_word.html

Two steps optimising spoken word recordings:

Compression with ffmpeg

ffmpeg -i in.wav \
  -af "acompressor=threshold=-24dB:attack=2.5:release=15:ratio=3" \
  out.wav

This command takes in a file in.wav (change .wav to .mp3, or whatever, if necessary), compresses it, and writes the output to out.wav, using an audio filter (-af) with a single stage (acompressor). The arguments to acompressor are a threshold (note the dB units and be aware that “dB” is case-sensitive in ffmpeg); the attack time (2.5ms); the release time (15ms); and the ratio (3:1)

Normalize using ffmpeg-normalize

github.com/slhck/ffmpeg-normalize

ffmpeg-normalize <input file> -o normalised.mp3 \
  -c:a mp3 -b:a 96 -pr -t -16 -tp -1

The options are: -o normalised.mp3 is the output file; -c:a mp3 specifies the output format (in this case MP3) and -b:a 96 specifies a bitrate of 96 (for spoken word, a reasonable trade-off between file size and audio quality); -pr shows a progress bar (normalisation is not a fast job, in part because it makes multiple passes over the audio); -t -16 specifies a target loudness of -16 (in LUFS, which we can think of as dB; the default is -23, which was probably appropriate for television back in the day, but -16 is recommended for podcasts and the like); -tp -1 specifies a maximum peak (in DBFS, which we can think of as dB; the default is -2 but I find that unduly conservative). The last of these options (-tp -1) I would consider optional, but for most of us -t -16 is recommended.