Am 28.05.25 um 00:49 schrieb Connor Grooms:
Hello,
I am currently working on converting a large number of files from wav (Max freq
64000 Hz) to mp3 (max Freq 48000 Hz).
My current call is:
ffmpeg -i "INPUTFILENAME.WAV" -af "aresample=resampler=soxr:precision=33" -q:a 2 -ar
48000 -first_pts 0 "OUTPUTFILENAME.mp3"
Also without the "first_pts 0" call I was noticing that there was a taper of
the waveform amplitude at the start and end of the output files.
I want to use the full dynamic range of the file and each file has a varying
maximum sound levels.
I have investigated ffmpeg-normalize however it seem to cause artifacts in the
mp3 files over the last 3 seconds where the file is repetitively clipped in 0.1
second intervals.
Normalizing a file to the maximum recorded level seems like a trivial thing to
do with ffmpeg... am I missing something obvious. Any ideas or assistance are
welcome.
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Normalization of audio, please refer to
https://k.ylo.ph/2016/04/04/loudnorm.html/. /I use for a lot of files
single-pass mode for single movies dual-pass mode. Both with acceptable
results.
This script is also usefull, to read source file parameter (bit=) and
set the same value as input for encoding (-ab ${bit}k) :
#!/bin/sh
for f in ../*.mp3; do
bit=`ffmpeg -i "$f" 2>&1 | grep Audio | awk -F", " '{print $5}' |
cut -d' ' -f1`
ffmpeg -i "$f" -af loudnorm=I=-16:TP=-1.5:LRA=11 -ab ${bit}k -ar 48k
"$(basename "$f" .mp3).mp3";
done
This script : mp3 to mp3, same file name, therefore the sources are in
the directory above. The script will read the bitrate of the sources and
set the same bitrate for output.
Best regards
Richard
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