I'm using the command below to convert from wav to flac:
ffmpeg -i "input.wav" -af aformat=s16:44100 "output.flac"
The bit-depth and sample_rate are ok, but is the command converting the
audio from CBR to VBR.
What should I add in the above command, or what command should I use for
the audio to keep the
CBR format in the FLAC output?
Flac is by definition VBR, anything else would just be waste of space.
It only just takes the amount of bytes needed for the specified
compression level. This can be set between 1 and 12. All levels are
lossless, the encoder just uses better and more computational expensive
algorithms for encoding. With modern processors level 12 is relatively
fast while in the early days of the standard (20 years ago) that level
would exceed the playing time by several magnitudes. It did however not
matter for decoding time. Default compression value in ffmpeg is 5
So for maximum lossless compression use
ffmpeg -i "input.wav" -af aformat=s16:44100 -c:a flac -compression_level 12
"output.flac"
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