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] =
  29 { /* (0.15) */
  30   29443, 28346, 25207, 20449, 14701,  8693,
  31    3143, -1352, -4402, -5865, -5850, -4673,
  32   -2783,  -672,  1211,  2536,  3130,  2991,
  33    2259,  1170,     0, -1001, -1652, -1868,
  34   -1666, -1147,  -464,   218,   756,  1060,
  35    1099,   904,   550,   135,  -245,  -514,
  36    -634,  -602,  -451,  -231,     0,   191,
  37     308,   340,   296,   198,    78,   -36,
  38    -120,  -163,  -165,  -132,   -79,   -19,
  39      34,    73,    91,    89,    70,    38,
  40       0,
  41 };
  42
  43 void ff_acelp_interpolate(
  44         int16_t* out,
  45         const int16_t* in,
  46         const int16_t* filter_coeffs,
  47         int precision,
  48         int frac_pos,
  49         int filter_length,
  50         int length)
  51 {
  52     int n, i;
  53
  54     assert(pitch_delay_frac >= 0 && pitch_delay_frac < precision);
  55
  56     for(n=0; n<length; n++)
  57     {
  58         int idx = 0;
  59         int v = 0x4000;
  60
  61         for(i=0; i<filter_length;)
  62         {
  63
  64             /* The reference G.729 and AMR fixed point code performs clipping after
  65                each of the two following accumulations.
  66                Since clipping affects only the synthetic OVERFLOW test without
  67                causing an int type overflow, it was moved outside the loop. */
  68
  69             /*  R(x):=ac_v[-k+x]
  70                 v += R(n-i)*ff_acelp_interp_filter(t+6i)
  71                 v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
  72
  73             v += in[n + i] * filter_coeffs[idx + frac_pos];
  74             idx += precision;
  75             i++;
  76             v += in[n - i] * filter_coeffs[idx - frac_pos];
  77         }
  78         if(av_clip_int16(v>>15) != (v>>15))
  79             av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n");
  80         out[n] = v >> 15;
  81     }
  82 }
  83
  84 void ff_acelp_convolve_circ(
  85         int16_t* fc_out,
  86         const int16_t* fc_in,
  87         const int16_t* filter,
  88         int len)
  89 {
  90     int i, k;
  91
  92     memset(fc_out, 0, len * sizeof(int16_t));
  93
  94     /* Since there are few pulses over an entire subframe (i.e. almost
  95        all fc_in[i] are zero) it is faster to loop over fc_in first. */
  96     for(i=0; i<len; i++)
  97     {
  98         if(fc_in[i])
  99         {
 100             for(k=0; k<i; k++)
 101                 fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15;
 102
 103             for(k=i; k<len; k++)
 104                 fc_out[k] += (fc_in[i] * filter[      k - i]) >> 15;
 105         }
 106     }
 107 }
 108
 109 int ff_acelp_lp_synthesis_filter(
 110         int16_t *out,
 111         const int16_t* filter_coeffs,
 112         const int16_t* in,
 113         int buffer_length,
 114         int filter_length,
 115         int stop_on_overflow,
 116         int rounder)
 117 {
 118     int i,n;
 119
 120     // These two lines are to avoid a -1 subtraction in the main loop
 121     filter_length++;
 122     filter_coeffs--;
 123
 124     for(n=0; n<buffer_length; n++)
 125     {
 126         int sum = rounder;
 127         for(i=1; i<filter_length; i++)
 128             sum -= filter_coeffs[i] * out[n-i];
 129
 130         sum = (sum >> 12) + in[n];
 131
 132         if(sum + 0x8000 > 0xFFFFU)
 133         {
 134             if(stop_on_overflow)
 135                 return 1;
 136             sum = (sum >> 31) ^ 32767;
 137         }
 138         out[n] = sum;
 139     }
 140
 141     return 0;
 142 }
 143
 144 void ff_acelp_high_pass_filter(
 145         int16_t* out,
 146         int hpf_f[2],
 147         const int16_t* in,
 148         int length)
 149 {
 150     int i;
 151     int tmp;
 152
 153     for(i=0; i<length; i++)
 154     {
 155         tmp =  (hpf_f[0]* 15836LL)>>13;
 156         tmp += (hpf_f[1]* -7667LL)>>13;
 157         tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);
 158
 159         /* With "+0x800" rounding, clipping is needed
 160            for ALGTHM and SPEECH tests. */
 161         out[i] = av_clip_int16((tmp + 0x800) >> 12);
 162
 163         hpf_f[1] = hpf_f[0];
 164         hpf_f[0] = tmp;
 165     }
 166 }