[FFmpeg-user] help: resampling audio settings

Moritz Barsnick barsnick at gmx.net
Thu Jan 14 21:18:20 CET 2016

On Wed, Jan 13, 2016 at 20:14:29 -0800, G A wrote:
> i’m resampling some audio with soxr with flac.  do these seem reasonable?

I know nothing about the optimal resampling and FLAC encoding options.
You don't say what you are trying to achieve.

That said, this I can comment:

> -codec:a flac -prediction_order_method 4 -lpc_passes 12 -exact_rice_parameters 1 -lpc_type 3 -ch_mode indep -sample_fmt s16 -ar 44100 -b:a 253588 -filter:a "volume=.7dB" -filter:a aresample=resampler=soxr=cheby=1:precision=28, dynaudnorm=f=24=g=64=p=.96=m=20=r=1=n=1=c=1=b=0 -filter:a shibata -ac 2

- You cannot specify multiple filters with "-vf" or "-filter:v". You
  need to read about how to chain filters.

- The syntax for handing options to your filters is incorrect. It is
  but rather

- The whitespace in your (first) filter chain (", dynaudionorm") needs
  to be protected from the shell.

- "shibata" is not a filter. It is an argument to the resampler's
  "-dither_method" option.


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