Volume loss converting DSF to Wav with Convolution

[simoncn]

(29-07-2019 19:24)richardpl Wrote:  The various converters may apply additional normalization and even compression (like in that 44.1k variant) after conversion. FFmpeg command you use does not do this. Normalization cant be done in single pass, you need to find value with astats/volumedetect and apply it with volume filter. For best output you may use double pcm variant of codec in intermediate steps for best final result.
So when you do conversion of DSD64 to pcm, I'm afraid you will need to first convert to wav and double sample format and than after normalization convert to final output which is s24 FLAC. (or even better keep it in float pcm)

When resampling you may use this additional parameter: -af aresample=44100:isf=flt:osf=dbl:precision=33

This is with my limited listening test.

Thanks very much for this. How much of this (if anything) can be done by a one-shot real-time on-the-fly conversion? This is a constraint of how MinimStreamer works. Would adding isf and osf to the aresample filter help?