2024-12-22 08:46:32 +11:00
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/*
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* G.728 / RealAudio 2.0 (28.8K) decoder
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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static void convolve(float *tgt, const float *src, int len, int n)
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{
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for (; n >= 0; n--)
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tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
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}
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/**
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* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
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*
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* @param order filter order
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* @param n input length
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* @param non_rec number of non-recursive samples
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* @param out filter output
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* @param hist pointer to the input history of the filter
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* @param out pointer to the non-recursive part of the output
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* @param out2 pointer to the recursive part of the output
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* @param window pointer to the windowing function table
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*/
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static void do_hybrid_window(void (*vector_fmul)(float *dst, const float *src0, const float *src1, int len),
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int order, int n, int non_rec, float *out,
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2024-12-22 09:12:41 +11:00
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const float *hist, float *out2, const float *window)
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2024-12-22 08:46:32 +11:00
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{
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int i;
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float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
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float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
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LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
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MAX_BACKWARD_FILTER_LEN +
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MAX_BACKWARD_FILTER_NONREC, 16)]);
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av_assert2(order>=0);
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vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
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convolve(buffer1, work + order , n , order);
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convolve(buffer2, work + order + n, non_rec, order);
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for (i=0; i <= order; i++) {
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out2[i] = out2[i] * ATTEN + buffer1[i];
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out [i] = out2[i] + buffer2[i];
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}
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/* Multiply by the white noise correcting factor (WNCF). */
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*out *= 257.0 / 256.0;
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}
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