libavcodec/acelp_filters.c

   1 /*
   2  * various filters for ACELP-based codecs
   3  *
   4  * Copyright (c) 2008 Vladimir Voroshilov
   5  *
   6  * This file is part of FFmpeg.
   7  *
   8  * FFmpeg is free software; you can redistribute it and/or
   9  * modify it under the terms of the GNU Lesser General Public
  10  * License as published by the Free Software Foundation; either
  11  * version 2.1 of the License, or (at your option) any later version.
  12  *
  13  * FFmpeg is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16  * Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public
  19  * License along with FFmpeg; if not, write to the Free Software
  20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21  */
  22
  23 #include <inttypes.h>
  24
  25 #include "avcodec.h"
  26 #include "acelp_filters.h"
  27
  28 const int16_t ff_acelp_interp_filter[61] = { /* (0.15) */
  29   29443, 28346, 25207, 20449, 14701,  8693,
  30    3143, -1352, -4402, -5865, -5850, -4673,
  31   -2783,  -672,  1211,  2536,  3130,  2991,
  32    2259,  1170,     0, -1001, -1652, -1868,
  33   -1666, -1147,  -464,   218,   756,  1060,
  34    1099,   904,   550,   135,  -245,  -514,
  35    -634,  -602,  -451,  -231,     0,   191,
  36     308,   340,   296,   198,    78,   -36,
  37    -120,  -163,  -165,  -132,   -79,   -19,
  38      34,    73,    91,    89,    70,    38,
  39       0,
  40 };
  41
  42 void ff_acelp_interpolate(int16_t* out, const int16_t* in,
  43                           const int16_t* filter_coeffs, int precision,
  44                           int frac_pos, int filter_length, int length)
  45 {
  46     int n, i;
  47
  48     assert(pitch_delay_frac >= 0 && pitch_delay_frac < precision);
  49
  50     for (n = 0; n < length; n++) {
  51         int idx = 0;
  52         int v = 0x4000;
  53
  54         for (i = 0; i < filter_length;) {
  55
  56             /* The reference G.729 and AMR fixed point code performs clipping after
  57                each of the two following accumulations.
  58                Since clipping affects only the synthetic OVERFLOW test without
  59                causing an int type overflow, it was moved outside the loop. */
  60
  61             /*  R(x):=ac_v[-k+x]
  62                 v += R(n-i)*ff_acelp_interp_filter(t+6i)
  63                 v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
  64
  65             v += in[n + i] * filter_coeffs[idx + frac_pos];
  66             idx += precision;
  67             i++;
  68             v += in[n - i] * filter_coeffs[idx - frac_pos];
  69         }
  70         if (av_clip_int16(v >> 15) != (v >> 15))
  71             av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n");
  72         out[n] = v >> 15;
  73     }
  74 }
  75
  76
  77 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
  78                                const int16_t* in, int length)
  79 {
  80     int i;
  81     int tmp;
  82
  83     for (i = 0; i < length; i++) {
  84         tmp  = (hpf_f[0]* 15836LL) >> 13;
  85         tmp += (hpf_f[1]* -7667LL) >> 13;
  86         tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);
  87
  88         /* With "+0x800" rounding, clipping is needed
  89            for ALGTHM and SPEECH tests. */
  90         out[i] = av_clip_int16((tmp + 0x800) >> 12);
  91
  92         hpf_f[1] = hpf_f[0];
  93         hpf_f[0] = tmp;
  94     }
  95 }
  96
  97 void ff_acelp_apply_order_2_transfer_function(float *buf,
  98                                               const float zero_coeffs[2],
  99                                               const float pole_coeffs[2],
 100                                               float gain, float mem[2], int n)
 101 {
 102     int i;
 103     float tmp;
 104
 105     for (i = 0; i < n; i++) {
 106         tmp = gain * buf[i] - pole_coeffs[0] * mem[0] - pole_coeffs[1] * mem[1];
 107         buf[i] =        tmp + zero_coeffs[0] * mem[0] + zero_coeffs[1] * mem[1];
 108
 109         mem[1] = mem[0];
 110         mem[0] = tmp;
 111     }
 112 }