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authorPaul B Mahol <onemda@gmail.com>2021-03-04 14:01:39 +0100
committerPaul B Mahol <onemda@gmail.com>2021-07-25 12:15:29 +0200
commit274112c88d89d839a27c0766f558f065f9eee0d7 (patch)
tree95bcc08ffbad3c8953e22e15257bde2ef15241c3 /libavfilter/vf_program_opencl.c
parenta2a7547b2f404b8c49f13bfdfb7966445dbe08fc (diff)
downloadffmpeg-274112c88d89d839a27c0766f558f065f9eee0d7.tar.gz
avfilter/f_ebur128: add all sample rates support
The magic constants come from the unofficial "ITU-R BS.1770-1 filter specifications"¹ by Raiden (libebur128) which relies on "Parameter Quantization in Direct-Form Recursive Audio Filters"² by Brian Neunaber. The constants seem to include a quantization bias, for example: - Vb is supposed to be exactly √Vh in a high shelf filter - the Pre-filter Gain should likely be 4dB - Pre Q and RLB Q are respectively very close to √½ and ½ Those are not adjusted to prevent the values from drifting away from the official specifications. An alternative to this approach would be to requantize on the fly as proposed by pbelkner³, where the 48kHz code path would use the exact specifications constants while derivating constants for other frequencies. [1]: https://www.scribd.com/document/49991813/ITU-R-BS-1770-1-filters [2]: https://www.scribd.com/document/6531763/Direct-Form-Filter-Parameter-Quantization [3]: https://hydrogenaud.io/index.php?topic=86116.msg740092#msg740092
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