145 lines
4.5 KiB
C
145 lines
4.5 KiB
C
/*
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* various filters for ACELP-based codecs
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*
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* Copyright (c) 2008 Vladimir Voroshilov
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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#include <inttypes.h>
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#include "avcodec.h"
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#include "acelp_filters.h"
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const int16_t ff_acelp_interp_filter[61] = { /* (0.15) */
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29443, 28346, 25207, 20449, 14701, 8693,
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3143, -1352, -4402, -5865, -5850, -4673,
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-2783, -672, 1211, 2536, 3130, 2991,
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2259, 1170, 0, -1001, -1652, -1868,
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-1666, -1147, -464, 218, 756, 1060,
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1099, 904, 550, 135, -245, -514,
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-634, -602, -451, -231, 0, 191,
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308, 340, 296, 198, 78, -36,
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-120, -163, -165, -132, -79, -19,
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34, 73, 91, 89, 70, 38,
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0,
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};
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void ff_acelp_interpolate(int16_t* out, const int16_t* in,
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const int16_t* filter_coeffs, int precision,
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int frac_pos, int filter_length, int length)
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{
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int n, i;
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assert(frac_pos >= 0 && frac_pos < precision);
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for (n = 0; n < length; n++) {
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int idx = 0;
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int v = 0x4000;
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for (i = 0; i < filter_length;) {
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/* The reference G.729 and AMR fixed point code performs clipping after
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each of the two following accumulations.
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Since clipping affects only the synthetic OVERFLOW test without
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causing an int type overflow, it was moved outside the loop. */
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/* R(x):=ac_v[-k+x]
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v += R(n-i)*ff_acelp_interp_filter(t+6i)
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v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
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v += in[n + i] * filter_coeffs[idx + frac_pos];
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idx += precision;
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i++;
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v += in[n - i] * filter_coeffs[idx - frac_pos];
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}
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if (av_clip_int16(v >> 15) != (v >> 15))
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av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n");
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out[n] = v >> 15;
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}
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}
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void ff_acelp_interpolatef(float *out, const float *in,
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const float *filter_coeffs, int precision,
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int frac_pos, int filter_length, int length)
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{
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int n, i;
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for (n = 0; n < length; n++) {
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int idx = 0;
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float v = 0;
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for (i = 0; i < filter_length;) {
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v += in[n + i] * filter_coeffs[idx + frac_pos];
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idx += precision;
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i++;
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v += in[n - i] * filter_coeffs[idx - frac_pos];
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}
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out[n] = v;
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}
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}
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void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
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const int16_t* in, int length)
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{
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int i;
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int tmp;
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for (i = 0; i < length; i++) {
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tmp = (hpf_f[0]* 15836LL) >> 13;
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tmp += (hpf_f[1]* -7667LL) >> 13;
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tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);
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/* With "+0x800" rounding, clipping is needed
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for ALGTHM and SPEECH tests. */
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out[i] = av_clip_int16((tmp + 0x800) >> 12);
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hpf_f[1] = hpf_f[0];
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hpf_f[0] = tmp;
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}
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}
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void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
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const float zero_coeffs[2],
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const float pole_coeffs[2],
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float gain, float mem[2], int n)
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{
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int i;
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float tmp;
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for (i = 0; i < n; i++) {
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tmp = gain * in[i] - pole_coeffs[0] * mem[0] - pole_coeffs[1] * mem[1];
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out[i] = tmp + zero_coeffs[0] * mem[0] + zero_coeffs[1] * mem[1];
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mem[1] = mem[0];
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mem[0] = tmp;
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}
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}
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void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
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{
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float new_tilt_mem = samples[size - 1];
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int i;
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for (i = size - 1; i > 0; i--)
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samples[i] -= tilt * samples[i - 1];
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samples[0] -= tilt * *mem;
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*mem = new_tilt_mem;
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}
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