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author | Rostislav Pehlivanov <atomnuker@gmail.com> | 2015-08-29 06:14:13 +0100 |
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committer | Rostislav Pehlivanov <atomnuker@gmail.com> | 2015-08-29 06:14:13 +0100 |
commit | 20962b567b4d1fe791ca4b537121f13f6e0f5fa0 (patch) | |
tree | be6f384d4251b1415ad4f7cc01c2695f4e49905d /libavcodec/mpegaudiotab.h | |
parent | f55cc579115e4bcfe26eb05ecaa1c823343e9e1e (diff) | |
download | ffmpeg-20962b567b4d1fe791ca4b537121f13f6e0f5fa0.tar.gz |
lpc: create a simplified Levinson-Durbin LPC handling float samples
This commit simply duplicates the functionality of ff_lpc_calc_coefs()
for the case of a Levinson-Durbin LPC with the only difference being
that floating point samples are accepted and the resulting coefficients
are raw and unquantized.
The motivation behind doing this is the fact that the AAC encoder
requires LPC in TNS and LTP and converting non-normalized floating
point coefficients to int32_t using SWR and again back for the LPC
coefficients was very impractical.
The current LPC interfaces were designed for int32_t in mind possibly
because FLAC and ALAC use this type for most internal operations.
The mathematics in case of floats remains of course identical.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Diffstat (limited to 'libavcodec/mpegaudiotab.h')
0 files changed, 0 insertions, 0 deletions