Janne Grunau <[email protected]> writes: > There's libavresample for general sample rate conversion, see Anton's > opus decoder patch how to use it in a decoder. That should work as first > pass before figuring out more specifics of the upsampling filter for > bit-exactness.
For a start, I implemented a *very* crude up-sampling, simply repeating each core sample twice. Thanks for the help with decoder flags; I can now run the decoder with ./avconv -i $INFILE -y -filter 'aformat=sample_rates=96000' -c:a pcm_s24be out.au to produce a 24-bit, 96 kHz output file. This gives me audio which is similar to the reference flac file, but with the largest errors around 40000. I can't yet say if the xll processing makes it closer or not, though... E.g., it's likely that I get the channel order wrong when adding the residuals. I just pushed the code to my public repo. Regards, /Niels -- Niels Möller. PGP-encrypted email is preferred. Keyid C0B98E26. Internet email is subject to wholesale government surveillance. _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
