When presented with a lossless audio file, what is the proper way to reduce sample rate and bit depth?
When reducing bit depth, I understand that I should apply dither, but I don't know what scale or method to use. (Actually, I read that I should use either a "triangular" or "shaping" dither, but I still don't know which of those to use. Does it come down to personal preference?)
When reducing sample rate, I have no idea if additional processing is required. (Evidently sample rate is far more complicated than bit depth.) At the moment, my conversion process looks like this:
ffmpeg -i <input.file> -compression_level 12 [-ar 44100 -sample_fmt 16] <output.flac>
I'd prefer general answers to application-specific ones, but I'll take what I can get. Thanks in advance!
I'm a consumer, not a producer. No further processing will be applied to these files.
Edit 0: Real world example: I have a 96/24 PCM (WAV) file. I want it to be 44.1/16.
Edit 1: Further research yielded this fantastic comparison of sample rate converters (with graphs)!