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/*
* G.728 / RealAudio 2.0 (28.8K) decoder
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
static void convolve(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
}
/**
* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
*
* @param order filter order
* @param n input length
* @param non_rec number of non-recursive samples
* @param out filter output
* @param hist pointer to the input history of the filter
* @param out pointer to the non-recursive part of the output
* @param out2 pointer to the recursive part of the output
* @param window pointer to the windowing function table
*/
static void do_hybrid_window(void (*vector_fmul)(float *dst, const float *src0, const float *src1, int len),
int order, int n, int non_rec, float *out,
const float *hist, float *out2, const float *window)
{
int i;
float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
MAX_BACKWARD_FILTER_LEN +
MAX_BACKWARD_FILTER_NONREC, 16)]);
av_assert2(order>=0);
vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
convolve(buffer1, work + order , n , order);
convolve(buffer2, work + order + n, non_rec, order);
for (i=0; i <= order; i++) {
out2[i] = out2[i] * ATTEN + buffer1[i];
out [i] = out2[i] + buffer2[i];
}
/* Multiply by the white noise correcting factor (WNCF). */
*out *= 257.0 / 256.0;
}
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