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ffmpeg / libavcodec / acelp_filters.c @ 2912e87a

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/*
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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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