FFnorm

automatic audio Normalization tool powered by FFmpeg for normalizing large batch of files.

Installations

Required dependencies are FFmpeg, FFprobe (shipped with FFmpeg), AWK and Node.js (if using Executables, Node.js is unnecessary).

didn't have them?

Download FFmpeg or Node.js

for AWK see installation guide

Usage

you can either run the Executable directly:

ffnorm norm -i "path/to/input/folder" "path/to/output/folder"

or run the source code var Node.js:

node ffnorm.js norm -i "path/to/input/folder" "path/to/output/folder"

Scan files Loudness

input path can be File or Folder

ffnorm scan -i "path/to/input/folder"

you can also specify Target loudness as guideline

ffnorm scan -i "path/to/input/folder" --target -9

FFnorm will mark how far-off each file's loudness from Target

Normalize Files

lets start by normalizing a single Video file, we defined Max Offset to 2LUFS meaning if this file is only 2LUFS away from default Target loudness (-14.4LUFS) it would be skipped.

ffnorm norm -i ./test/video.mp4 -of 2 ./test/norm_vid.mp4

for multiple files we can specify Folder path instead of File, FFnorm will look for supported files in the given folder and normalize them

ffnorm norm -i ./test -of 2 ./test/norm/

Options

Option	Descriptions	Type	Default	Note
'-i', '--input'	specify input file/folder	string	None	this option is required
'-o','--output'	specify output file/folder	string	None	(if none set will use the last command-line argument as output)
'norm', '--norm', '-n'	Normalize mode - scan Audio loudness of file/folder contents and normalize them according to the Target loudness set by '-t' option.	(No Value needed for this option)	None	(this option requires Output file/folder)
'scan','--scan', '-s'	Scan Audio loudness and report them on the terminal.	(No Value needed for this option)	None
'-t', '--target'	Target Loudness in LUFS	float	-14.4 (LUFS) (YouTube standard loudness)
'-q', '--qscale', -qscale	FFmpeg Quality Scale flag, this is to prevent losses during normalization. while the program tells FFmpeg to keep output bitrate as the same as input, there's still some losses being generated (if the media isn't lossless) due to compression over compression, each time the quality is going to get worse for the same bitrate. set qscale to a lower values result in a better quality 0 is lossless.	int	2	2 or 3 is what I think is the sweet spot, the file is quite heavy but it genuinely makes the losses invisible. For someone who'd like to Disable this option: set qscale to -1 (-q -1)
'-of', '--offset'	Max offset fron Target loudness before normalization become active.	float	1.3 (LUFS)
'-r', '--ratio'	How much Normalization is apply in percentage, 1.0 is 100% lower this value to prevent over-shooting	float	0.78 (78%)
'-f', '--format'	wrapper for ffmpeg '-f' option. this option specify Output format of output files	string	(whatever the input format is)	this option is omitted for scan mode
'-st','--scanthread'	Max number of Threads for loudness scanning	int	64 (threads)
'-nt', '--normthread'	Max number of Threads for audio normalization	int	32 (threads)
'-v', '--version'	prints program version and exit.	(No Value needed for this option)	None	has to be the first option to work
'-h', '--help', 'help'	display help message.	(No Value needed for this option)	None	has to be the first option to work

Supported Files Extentions

• aiff
• aif
• aifc
• flac
• mp3
• mp4
• mp4a
• m4a
• mkv
• mov
• wav
• webm
• ogg

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FFmpeg Commands use:

• getting audio loudness

ffmpeg -hide_banner -i audio.wav -af ebur128=framelog=verbose -f null - 2>&1 | awk "/I:/{print $2}"

• getting audio bitrate

ffprobe -v fatal -select_streams a:0 -show_entries stream=bit_rate -of compact=p=0:nk=1 audio.wav"

• modifying audio Gains

ffmpeg -hide_banner -y -i input.wav -movflags use_metadata_tags -map_metadata 0 -id3v2_version 3 -q:a (QSCALE) -af "volume=(GAIN)dB" -id3v2_version 3 -c:v copy output.wav

or

ffmpeg -hide_banner -y -i input.wav -movflags use_metadata_tags -map_metadata 0 -id3v2_version 3 -af "volume=(GAIN)dB" -id3v2_version 3 -b:a (BITRATE) -c:v copy output.wav