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/*
* various filters for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <inttypes.h>
#include "avcodec.h"
#include "acelp_filters.h"
const int16_t ff_acelp_interp_filter[61] = { /* (0.15) */
29443, 28346, 25207, 20449, 14701, 8693,
3143, -1352, -4402, -5865, -5850, -4673,
-2783, -672, 1211, 2536, 3130, 2991,
2259, 1170, 0, -1001, -1652, -1868,
-1666, -1147, -464, 218, 756, 1060,
1099, 904, 550, 135, -245, -514,
-634, -602, -451, -231, 0, 191,
308, 340, 296, 198, 78, -36,
-120, -163, -165, -132, -79, -19,
34, 73, 91, 89, 70, 38,
0,
};
void ff_acelp_interpolate(int16_t* out, const int16_t* in,
const int16_t* filter_coeffs, int precision,
int frac_pos, int filter_length, int length)
{
int n, i;
assert(frac_pos >= 0 && frac_pos < precision);
for (n = 0; n < length; n++) {
int idx = 0;
int v = 0x4000;
for (i = 0; i < filter_length;) {
/* The reference G.729 and AMR fixed point code performs clipping after
each of the two following accumulations.
Since clipping affects only the synthetic OVERFLOW test without
causing an int type overflow, it was moved outside the loop. */
/* R(x):=ac_v[-k+x]
v += R(n-i)*ff_acelp_interp_filter(t+6i)
v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
v += in[n + i] * filter_coeffs[idx + frac_pos];
idx += precision;
i++;
v += in[n - i] * filter_coeffs[idx - frac_pos];
}
if (av_clip_int16(v >> 15) != (v >> 15))
av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n");
out[n] = v >> 15;
}
}
void ff_acelp_interpolatef(float *out, const float *in,
const float *filter_coeffs, int precision,
int frac_pos, int filter_length, int length)
{
int n, i;
for (n = 0; n < length; n++) {
int idx = 0;
float v = 0;
for (i = 0; i < filter_length;) {
v += in[n + i] * filter_coeffs[idx + frac_pos];
idx += precision;
i++;
v += in[n - i] * filter_coeffs[idx - frac_pos];
}
out[n] = v;
}
}
void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
const int16_t* in, int length)
{
int i;
int tmp;
for (i = 0; i < length; i++) {
tmp = (hpf_f[0]* 15836LL) >> 13;
tmp += (hpf_f[1]* -7667LL) >> 13;
tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);
/* With "+0x800" rounding, clipping is needed
for ALGTHM and SPEECH tests. */
out[i] = av_clip_int16((tmp + 0x800) >> 12);
hpf_f[1] = hpf_f[0];
hpf_f[0] = tmp;
}
}
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
const float zero_coeffs[2],
const float pole_coeffs[2],
float gain, float mem[2], int n)
{
int i;
float tmp;
for (i = 0; i < n; i++) {
tmp = gain * in[i] - pole_coeffs[0] * mem[0] - pole_coeffs[1] * mem[1];
out[i] = tmp + zero_coeffs[0] * mem[0] + zero_coeffs[1] * mem[1];
mem[1] = mem[0];
mem[0] = tmp;
}
}
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
{
float new_tilt_mem = samples[size - 1];
int i;
for (i = size - 1; i > 0; i--)
samples[i] -= tilt * samples[i - 1];
samples[0] -= tilt * *mem;
*mem = new_tilt_mem;
}
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