How to set sample format when using sox with ffmpeg?

Question

I am trying to convert a 44.1k 16bit flac file into 48k 32 bit (float) wav file.

This is the command I use:

'ffmpeg -i in.flac -af aresample=resampler=soxr:precision=28:out_sample_fmt=fltp:out_sample_rate=48000 out.wav'

No matter which value I use for out_sample_fmt like s32, flt, fltp the output out.wav is only 16 bit.

What am I doing wrong here? How to get the highest quality (as in resampling) 32 bit floating point wav file with ffmpeg using soxr?

Answer 1

The issue isn't with soxr or aresample. Typically, after media data is filtered, it is encoded before being written to output. For each output format, there is a default encoder designated for each type of stream (audio, video..). In case of WAV, it's pcm_s16le for audio.

Add -c:a pcm_f32le for 32-bit floating point PCM, in little-endian order. Change le to be for big-endian.