How can I normalize audio using ffmpeg?

Option 1: Built-in Normalization Filters

Current ffmpeg has two filters that can be directly used for normalization – although they are already quite advanced, so they do not simply apply gain to reach a peak level. Here they are:

  • loudnorm: loudness normalization according to EBU R128. You can set an integrated loudness target, a loudness range target, or maximum true peak. This is recommended for publishing audio and video and it is used by broadcasters all over the world.
  • dynaudnorm: “intelligent” loudness normalization without clipping, which applies normalization dynamically over windowed portions of the file. This may change the characteristics of the sound, so it should be applied with caution.

Also, the volume filter can be used to perform simple volume adjustments. See the Audio Volume Manipulation wiki entry for more.

The loudnorm filter can be used with one pass, but it is recommended to perform two passes, which enables more accurate linear normalization. This is a little hard to automate. Also, if you want a “simple” RMS-based or peak normalization to 0 dBFS (or any other target), read on.

Option 2: Use the ffmpeg-normalize tool

I created a Python program to normalize media files, available on PyPi as well. You simply:

  • download ffmpeg (choose a static build, version 3.1 or higher)
  • put the ffmpeg executable in your $PATH by either adding it in, for example, /usr/local/bin, or adding its directory to $PATH
  • Run pip install ffmpeg-normalize
  • Use ffmpeg-normalize

For example:

ffmpeg-normalize input.mp4 -o output.mp4 -c:a aac -b:a 192k

Or, to simply batch-normalize a number of audio files and write them as uncompressed WAV to an output folder:

ffmpeg-normalize *.m4a -of /path/to/outputFolder -ext wav

The tool supports EBU R128 (default), RMS and peak. Have a look at ffmpeg-normalize -h for more options and check the README for some examples.

Also, it supports re-encoding with other encoders (e.g., AAC or MP3), or automatic merging of the audio back into the video.

Option 3: Manually normalizing audio with ffmpeg

In ffmpeg you can use the volume filter to change the volume of a track. Make sure you download a recent version of the program.

This guide is for peak normalization, meaning that it will make the loudest part in the file sit at 0 dB instead of something lower. There is also RMS-based normalization which tries to make the average loudness the same across multiple files. To do that, do not try to push the maximum volume to 0 dB, but the mean volume to the dB level of choice (e.g. -26 dB).

Find out the gain to apply

First you need to analyze the audio stream for the maximum volume to see if normalizing would even pay off:

ffmpeg -i video.avi -af "volumedetect" -vn -sn -dn -f null /dev/null

Replace /dev/null with NUL on Windows.
The -vn, -sn, and -dn arguments instruct ffmpeg to ignore non-audio streams during this analysis. This drastically speeds up the analysis.

This will output something like the following:

[Parsed_volumedetect_0 @ 0x7f8ba1c121a0] mean_volume: -16.0 dB
[Parsed_volumedetect_0 @ 0x7f8ba1c121a0] max_volume: -5.0 dB
[Parsed_volumedetect_0 @ 0x7f8ba1c121a0] histogram_0db: 87861

As you can see, our maximum volume is -5.0 dB, so we can apply 5 dB gain. If you get a value of 0 dB, then you don't need to normalize the audio.

Apply the volume filter:

Now we apply the volume filter to an audio file. Note that applying the filter means we will have to re-encode the audio stream. What codec you want for audio depends on the original format, of course. Here are some examples:

  • Plain audio file: Just encode the file with whatever encoder you need:

    ffmpeg -i input.wav -af "volume=5dB" output.mp3

    Your options are very broad, of course.

  • AVI format: Usually there's MP3 audio with video that comes in an AVI container:

    ffmpeg -i video.avi -af "volume=5dB" -c:v copy -c:a libmp3lame -q:a 2 output.avi

    Here we chose quality level 2. Values range from 0–9 and lower means better. Check the MP3 VBR guide for more info on setting the quality. You can also set a fixed bitrate with -b:a 192k, for example.

  • MP4 format: With an MP4 container, you will typically find AAC audio. We can use ffmpeg's build-in AAC encoder.

    ffmpeg -i video.mp4 -af "volume=5dB" -c:v copy -c:a aac -b:a 192k output.mp4

    Here you can also use other AAC encoders. Some of them support VBR, too. See this answer and the AAC encoding guide for some tips.

In the above examples, the video stream will be copied over using -c:v copy. If there are subtitles in your input file, or multiple video streams, use the option -map 0 before the output filename.


I can not comment on the best message so that is my ugly bash based on it to do that

ffmpeg -i sound.mp3 -af volumedetect -f null -y nul &> original.txt
grep "max_volume" original.txt > original1.tmp
sed -i 's|: -|=|' original1.tmp
if [ $? = 0 ]
 then
 sed -i 's| |\r\n|' original.tmp
 sed -i 's| |\r\n|' original.tmp
 sed -i 's| |\r\n|' original.tmp
 sed -i 's| |\r\n|' original.tmp
 grep "max_volume" original1.tmp > original2.tmp
 sed -i 's|max_volume=||' original2.tmp
 yourscriptvar=$(cat "./original2.tmp")dB
 rm result.mp3
 ffmpeg -i sound.mp3 -af "volume=$yourscriptvar" result.mp3
 ffmpeg -i result.mp3 -af volumedetect -f null -y nul &> result.txt
fi

Here's a script to normalize sound levels of .m4a files. Watch out if the sound levels are too quiet to start with. The final sound can be better if you use something like Audacity in that case.

#!/bin/bash

# Purpose: Use ffmpeg to normalize .m4a audio files to bring them up to max volume, if they at first have negative db volume. Doesn't process them if not. Keeps bitrate same as source files.
# Parameters: $1 should be the name of the directory containing input .m4a files.
#   $2 should be the output directory.

INPUTDIR=$1
OUTPUTDIR=$2

<<"COMMENT"

# For ffmpeg arguments http://superuser.com/questions/323119/how-can-i-normalize-audio-using-ffmpeg
# and
# https://kdecherf.com/blog/2012/01/14/ffmpeg-converting-m4a-files-to-mp3-with-the-same-bitrate/
ffmpeg -i test.m4a -af "volumedetect" -f null /dev/null

ffmpeg -i test.m4a -af "volumedetect" -f null /dev/null 2>&1 | grep max_volume
# output: max_volume: -10.3 dB

ffmpeg -i test.m4a -af "volumedetect" -f null /dev/null 2>&1 | grep 'max_volume\|Duration'
# Output:
#  Duration: 00:00:02.14, start: 0.000000, bitrate: 176 kb/s
# [Parsed_volumedetect_0 @ 0x7f8531e011a0] max_volume: -10.3 dB

ffmpeg -i test.m4a -af "volumedetect" -f null /dev/null 2>&1 | grep max_volume | awk -F': ' '{print $2}' | cut -d' ' -f1
# Output: -10.3

ffmpeg -i test.m4a 2>&1 | grep Audio
# output: Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 170 kb/s (default)

ffmpeg -i test.m4a 2>&1 | grep Audio | awk -F', ' '{print $5}' | cut -d' ' -f1
# output: 170

# This works, but I get a much smaller output file. The sound levels do appear normalized.
ffmpeg -i test.m4a -af "volume=10.3dB" -c:v copy -c:a aac -strict experimental output.m4a

# Operates quietly.
ffmpeg -i test.m4a -af "volume=10.3dB" -c:v copy -c:a aac -strict experimental -b:a 192k output.m4a -loglevel quiet

COMMENT

# $1 (first param) should be the name of a .m4a input file, with .m4a extension
# $2 should be name of output file, with extension
function normalizeAudioFile {
    INPUTFILE=$1
    OUTPUTFILE=$2

    DBLEVEL=`ffmpeg -i ${INPUTFILE} -af "volumedetect" -f null /dev/null 2>&1 | grep max_volume | awk -F': ' '{print $2}' | cut -d' ' -f1`

    # We're only going to increase db level if max volume has negative db level.
    # Bash doesn't do floating comparison directly
    COMPRESULT=`echo ${DBLEVEL}'<'0 | bc -l`
    if [ ${COMPRESULT} -eq 1 ]; then
        DBLEVEL=`echo "-(${DBLEVEL})" | bc -l`
        BITRATE=`ffmpeg -i ${INPUTFILE} 2>&1 | grep Audio | awk -F', ' '{print $5}' | cut -d' ' -f1`

        # echo $DBLEVEL
        # echo $BITRATE

        ffmpeg -i ${INPUTFILE} -af "volume=${DBLEVEL}dB" -c:v copy -c:a aac -strict experimental -b:a ${BITRATE}k ${OUTPUTFILE} -loglevel quiet

    else
        echo "Already at max db level:" $DBLEVEL "just copying exact file"
        cp ${INPUTFILE} ${OUTPUTFILE}
    fi
}

for inputFilePath in ${INPUTDIR}/*; do
    inputFile=$(basename $inputFilePath)
    echo "Processing input file: " $inputFile
    outputFilePath=${OUTPUTDIR}/$inputFile
    normalizeAudioFile ${inputFilePath} ${outputFilePath}
done

Tags:

Audio

Ffmpeg

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