Trying to figure out ffmpeg, currently working on getting 24bit/96khz FLAC files into 16bit/48khz.
As slhck says, ffmpeg appears unable to recognise flac.
The bit depth can be changed with the sample_fmt option, e.g. Note that not all formats are supported by every encoder. See the chapter Audio Options in the FFmpeg command line documentation. @xjcl The codec you are using only supports 4 bit.
Tl;DR If the source was recorded mastered and released with 24 bit depth and a 96kHz sample rate. The answer is absolutely, yes.
Reading and Writing Raw Audio The pcm_s16le tells you what format your audio is in. And that happens to be a common format.
ffmpeg -i input.flac -sample_fmt s16 -ar 48000 output.flac
ffmpeg -sample_fmts
ffmpeg -h encoder=flac
ffmpeg -i input.flac -af aresample=out_sample_fmt=s16:out_sample_rate=48000 output.flac
Either example will result in the same output: you can verify with the hash muxer.
See the -dither_method
option for a list of available dithering methods and additional resampling options. Example:
ffmpeg -i input.flac -dither_method triangular_hp -sample_fmt s16 -ar 48000 output.flac
FFmpeg supports two resamplers: the default swresample library, and the external SoX resampler (soxr).
To use soxr your ffmpeg
must be compiled with --enable-libsoxr
. Then choose it with the -resampler
option:
ffmpeg -i input.flac -resampler soxr -sample_fmt s16 -ar 48000 output.flac
Or use the aresample filter to do it all:
ffmpeg -i input.flac -af aresample=resampler=soxr:out_sample_fmt=s16:out_sample_rate=48000 output.flac
As a bash script, that produces new files with -16 appended to their names; one could rename then delete the original files easily in the script but I'm a little too paranoid for that.
#!/bin/sh
# requires: ffmpeg
for f in *.flac;
do
echo "Processing $f"
ffmpeg -i "$f" -sample_fmt s16 -ar 48000 "${f%.flac}-16.flac"
done
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