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avcodec/g728_template: do_hybrid_window() template

intended for use by RealAudio 2.0 (28.8k) and G.728 decoders.

Peter Ross committed 1 year ago

Commit 93368021 parent b1172b82 files · +68 −47Browse files

Unified · Split

2 files
  1. libavcodec/g728_template.c +65 −0
  2. libavcodec/ra288.c +3 −47
libavcodec/g728_template.c+65−0
@@ -0,0 +1,65 @@
1/*
2 * G.728 / RealAudio 2.0 (28.8K) decoder
3 *
4 * This file is part of FFmpeg.
5 *
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21static void convolve(float *tgt, const float *src, int len, int n)
22{
23 for (; n >= 0; n--)
24 tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
25
26}
27
28/**
29 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
30 *
31 * @param order filter order
32 * @param n input length
33 * @param non_rec number of non-recursive samples
34 * @param out filter output
35 * @param hist pointer to the input history of the filter
36 * @param out pointer to the non-recursive part of the output
37 * @param out2 pointer to the recursive part of the output
38 * @param window pointer to the windowing function table
39 */
40static void do_hybrid_window(void (*vector_fmul)(float *dst, const float *src0, const float *src1, int len),
41 int order, int n, int non_rec, float *out,
42 float *hist, float *out2, const float *window)
43{
44 int i;
45 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
46 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
47 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
48 MAX_BACKWARD_FILTER_LEN +
49 MAX_BACKWARD_FILTER_NONREC, 16)]);
50
51 av_assert2(order>=0);
52
53 vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
54
55 convolve(buffer1, work + order , n , order);
56 convolve(buffer2, work + order + n, non_rec, order);
57
58 for (i=0; i <= order; i++) {
59 out2[i] = out2[i] * ATTEN + buffer1[i];
60 out [i] = out2[i] + buffer2[i];
61 }
62
63 /* Multiply by the white noise correcting factor (WNCF). */
64 *out *= 257.0 / 256.0;
65}
libavcodec/ra288.c+3−47
@@ -37,6 +37,8 @@ · 36 unmodified lines
37#define MAX_BACKWARD_FILTER_ORDER 3637#define MAX_BACKWARD_FILTER_ORDER 36
38#define MAX_BACKWARD_FILTER_LEN 4038#define MAX_BACKWARD_FILTER_LEN 40
39#define MAX_BACKWARD_FILTER_NONREC 3539#define MAX_BACKWARD_FILTER_NONREC 35
40#define ATTEN 0.5625
41#include "g728_template.c"
40 42
41#define RA288_BLOCK_SIZE 543#define RA288_BLOCK_SIZE 5
42#define RA288_BLOCKS_PER_FRAME 3244#define RA288_BLOCKS_PER_FRAME 32
@@ -87,13 +89,6 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx) · 44 unmodified lines
87 return 0;89 return 0;
88}90}
89 91
90static void convolve(float *tgt, const float *src, int len, int n)
91{
92 for (; n >= 0; n--)
93 tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
94
95}
96
97static void decode(RA288Context *ractx, float gain, int cb_coef)92static void decode(RA288Context *ractx, float gain, int cb_coef)
98{93{
99 int i;94 int i;
@@ -131,45 +126,6 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) · 31 unmodified lines
131 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);126 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
132}127}
133 128
134/**
135 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
136 *
137 * @param order filter order
138 * @param n input length
139 * @param non_rec number of non-recursive samples
140 * @param out filter output
141 * @param hist pointer to the input history of the filter
142 * @param out pointer to the non-recursive part of the output
143 * @param out2 pointer to the recursive part of the output
144 * @param window pointer to the windowing function table
145 */
146static void do_hybrid_window(RA288Context *ractx,
147 int order, int n, int non_rec, float *out,
148 float *hist, float *out2, const float *window)
149{
150 int i;
151 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
152 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
153 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
154 MAX_BACKWARD_FILTER_LEN +
155 MAX_BACKWARD_FILTER_NONREC, 16)]);
156
157 av_assert2(order>=0);
158
159 ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
160
161 convolve(buffer1, work + order , n , order);
162 convolve(buffer2, work + order + n, non_rec, order);
163
164 for (i=0; i <= order; i++) {
165 out2[i] = out2[i] * 0.5625 + buffer1[i];
166 out [i] = out2[i] + buffer2[i];
167 }
168
169 /* Multiply by the white noise correcting factor (WNCF). */
170 *out *= 257.0 / 256.0;
171}
172
173/**129/**
174 * Backward synthesis filter, find the LPC coefficients from past speech data.130 * Backward synthesis filter, find the LPC coefficients from past speech data.
175 */131 */
@@ -180,7 +136,7 @@ static void backward_filter(RA288Context *ractx, · 4 unmodified lines
180{136{
181 float temp[MAX_BACKWARD_FILTER_ORDER+1];137 float temp[MAX_BACKWARD_FILTER_ORDER+1];
182 138
183 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);139 do_hybrid_window(ractx->vector_fmul, order, n, non_rec, temp, hist, rec, window);
184 140
185 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))141 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
186 ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));142 ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));

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