Am I doing this right? (SRC 32-bit floating-point intermediate file) 2019-03-10 16:19:41 I've got an idea from 'Mastered for iTunes' droplet:QuoteApple’s latest encoding methodology is a two-step process. The first step in the encoding path is to use state-of-the-art, mastering-quality Sample Rate Conversion (SRC) to resample the master file to a sample rate of 44.1kHz.Because this SRC outputs a 32-bit floating-point file, it can preserve values that might otherwise fall outside of the permitted amplitude range. This critical intermediary step prevents any aliasing or clipping that could otherwise occur in SRC. It is this 32-bit floating file that’s used as the input to the encoder and is one key reason for such stunning results.Our encoders then use every bit of resolution available, preserving all the dynamic range of a 24-bit source file and eliminating the need for dithering. The advantage of this is twofold. Not only does it obviate the need of adding dither noise, it also lets the encoders work more efficiently as they don’t need to waste resources encoding this unwanted and unnecessary noise.By using this highly accurate file directly from our SRC and taking advantage of its clean signal, our encoder can deliver the final product exactly as the artist and sound engineers intended it to sound.So I'm doing this:Code: [Select]afconvert <in.aif> -d LEF32@48000 -f WAVE --src-complexity bats -r 127 <out.wav>opusenc --bitrate 128 <in.wav> <out.opus>Source is 96/24 from HDtracks.The result is fantastic and indistinguishable from the source, I'm just wondering is this intermediate step/method sane or if it's even doing anything to improve the input for opusenc to work with?