When converting SACD DSF audio files to FLAC (or any other PCM type format) the conversion process will most likely introduce distortion in the upper frequencies. In order to eliminate this you need to use the lowpass filter during the conversion process. This post is mainly so I won’t forget.
These are the ffmpeg commands I used to convert to regular FLAC. To convert to 24bit FLAC use s32 for the sample format.
for i in *.dsf; do ffmpeg -i "$i" -af "lowpass=24000, volume=6dB" -sample_fmt s16 -ar 48000 "${i%.*}.flac"; done
