2 * various filters for ACELP-based codecs
4 * Copyright (c) 2008 Vladimir Voroshilov
6 * This file is part of FFmpeg.
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.
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.
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
26 #include "acelp_filters.h"
30 const int16_t ff_acelp_interp_filter[61] =
32 29443, 28346, 25207, 20449, 14701, 8693,
33 3143, -1352, -4402, -5865, -5850, -4673,
34 -2783, -672, 1211, 2536, 3130, 2991,
35 2259, 1170, 0, -1001, -1652, -1868,
36 -1666, -1147, -464, 218, 756, 1060,
37 1099, 904, 550, 135, -245, -514,
38 -634, -602, -451, -231, 0, 191,
39 308, 340, 296, 198, 78, -36,
40 -120, -163, -165, -132, -79, -19,
41 34, 73, 91, 89, 70, 38,
45 void ff_acelp_interpolate(
48 const int16_t* filter_coeffs,
56 assert(pitch_delay_frac >= 0 && pitch_delay_frac < precision);
58 for(n=0; n<length; n++)
63 for(i=0; i<filter_length;)
66 /* The reference G.729 and AMR fixed point code performs clipping after
67 each of the two following accumulations.
68 Since clipping affects only the synthetic OVERFLOW test without
69 causing an int type overflow, it was moved outside the loop. */
72 v += R(n-i)*ff_acelp_interp_filter(t+6i)
73 v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
75 v += in[n + i] * filter_coeffs[idx + pitch_delay_frac];
78 v += in[n - i] * filter_coeffs[idx - pitch_delay_frac];
80 out[n] = av_clip_int16(v >> 15);
84 void ff_acelp_convolve_circ(
87 const int16_t* filter,
92 memset(fc_out, 0, subframe_size * sizeof(int16_t));
94 /* Since there are few pulses over an entire subframe (i.e. almost
95 all fc_in[i] are zero) it is faster to swap two loops and process
96 non-zero samples only. In the case of G.729D the buffer contains
97 two non-zero samples before the call to ff_acelp_enhance_harmonics
98 and, due to pitch_delay being bounded by [20; 143], a maximum
99 of four non-zero samples for a total of 40 after the call. */
100 for(i=0; i<subframe_size; i++)
105 fc_out[k] += (fc_in[i] * filter[subframe_size + k - i]) >> 15;
107 for(k=i; k<subframe_size; k++)
108 fc_out[k] += (fc_in[i] * filter[k - i]) >> 15;
113 int ff_acelp_lp_synthesis_filter(
115 const int16_t* filter_coeffs,
119 int stop_on_overflow,
124 // These two lines are to avoid a -1 subtraction in the main loop
128 for(n=0; n<buffer_length; n++)
131 for(i=1; i<filter_length; i++)
132 sum -= filter_coeffs[i] * out[n-i];
134 sum = (sum >> 12) + in[n];
136 /* Check for overflow */
137 if(sum + 0x8000 > 0xFFFFU)
141 sum = (sum >> 31) ^ 32767;
149 void ff_acelp_weighted_filter(
152 const int16_t *weight_pow,
156 for(n=0; n<filter_length; n++)
157 out[n] = (in[n] * weight_pow[n] + 0x4000) >> 15; /* (3.12) = (0.15) * (3.12) with rounding */
160 void ff_acelp_high_pass_filter(
169 for(i=0; i<length; i++)
171 tmp = MULL(hpf_f[0], 15836); /* (14.13) = (13.13) * (1.13) */
172 tmp += MULL(hpf_f[1], -7667); /* (13.13) = (13.13) * (0.13) */
173 tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]); /* (14.13) = (0.13) * (14.0) */
175 /* Multiplication by 2 with rounding can cause short type
176 overflow, thus clipping is required. */
178 out[i] = av_clip_int16((tmp + 0x800) >> 12); /* (15.0) = 2 * (13.13) = (14.13) */