# ffmpeg will not change sample rate unless you tell it to 
# (or the output codec does not support it, but then it will most probably fail). 
# So this should be enough:

ffmpeg -i input.flac output.wav

# ffmepg will however not preserve bit depth and default to 16-bit encoding, 
# so if your input is 24 bit, you have to use:

ffmpeg -i input.flac -c:a pcm_s24le output.wav