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author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-01-27 15:20:43 +0000 |
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committer | Mans Rullgard <mans@mansr.com> | 2011-01-28 00:07:35 +0000 |
commit | 9d06d7bce3babb82ed650c13ed13a57f6f626a71 (patch) | |
tree | 022760430dcd336114b22b841e8a0dcb51dbd62c /libavcodec/binkaudio.c | |
parent | 97b04f5ed3ab3a62e57f0c16841fb8f10d0a174c (diff) | |
download | ffmpeg-9d06d7bce3babb82ed650c13ed13a57f6f626a71.tar.gz |
Remove the add bias hack for the C version of DSPContext.float_to_int16_*().
Diffstat (limited to 'libavcodec/binkaudio.c')
-rw-r--r-- | libavcodec/binkaudio.c | 5 |
1 files changed, 0 insertions, 5 deletions
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index 7f579822db..ae2f6c88b0 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -222,11 +222,6 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct) ff_rdft_calc(&s->trans.rdft, coeffs); } - if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { - for (i = 0; i < s->channels; i++) - for (j = 0; j < s->frame_len; j++) - s->coeffs_ptr[i][j] = 385.0 + s->coeffs_ptr[i][j]*(1.0/32767.0); - } s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels); if (!s->first) { |