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path: root/libavcodec/ra288.c (plain)
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1/*
2 * RealAudio 2.0 (28.8K)
3 * Copyright (c) 2003 The FFmpeg project
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22#include "libavutil/channel_layout.h"
23#include "libavutil/float_dsp.h"
24#include "libavutil/internal.h"
25
26#define BITSTREAM_READER_LE
27#include "avcodec.h"
28#include "celp_filters.h"
29#include "get_bits.h"
30#include "internal.h"
31#include "lpc.h"
32#include "ra288.h"
33
34#define MAX_BACKWARD_FILTER_ORDER 36
35#define MAX_BACKWARD_FILTER_LEN 40
36#define MAX_BACKWARD_FILTER_NONREC 35
37
38#define RA288_BLOCK_SIZE 5
39#define RA288_BLOCKS_PER_FRAME 32
40
41typedef struct RA288Context {
42 AVFloatDSPContext *fdsp;
43 DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
44 DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
45
46 /** speech data history (spec: SB).
47 * Its first 70 coefficients are updated only at backward filtering.
48 */
49 float sp_hist[111];
50
51 /// speech part of the gain autocorrelation (spec: REXP)
52 float sp_rec[37];
53
54 /** log-gain history (spec: SBLG).
55 * Its first 28 coefficients are updated only at backward filtering.
56 */
57 float gain_hist[38];
58
59 /// recursive part of the gain autocorrelation (spec: REXPLG)
60 float gain_rec[11];
61} RA288Context;
62
63static av_cold int ra288_decode_close(AVCodecContext *avctx)
64{
65 RA288Context *ractx = avctx->priv_data;
66
67 av_freep(&ractx->fdsp);
68
69 return 0;
70}
71
72static av_cold int ra288_decode_init(AVCodecContext *avctx)
73{
74 RA288Context *ractx = avctx->priv_data;
75
76 avctx->channels = 1;
77 avctx->channel_layout = AV_CH_LAYOUT_MONO;
78 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
79
80 if (avctx->block_align <= 0) {
81 av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
82 return AVERROR_PATCHWELCOME;
83 }
84
85 ractx->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
86 if (!ractx->fdsp)
87 return AVERROR(ENOMEM);
88
89 return 0;
90}
91
92static void convolve(float *tgt, const float *src, int len, int n)
93{
94 for (; n >= 0; n--)
95 tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
96
97}
98
99static void decode(RA288Context *ractx, float gain, int cb_coef)
100{
101 int i;
102 double sumsum;
103 float sum, buffer[5];
104 float *block = ractx->sp_hist + 70 + 36; // current block
105 float *gain_block = ractx->gain_hist + 28;
106
107 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
108
109 /* block 46 of G.728 spec */
110 sum = 32.0;
111 for (i=0; i < 10; i++)
112 sum -= gain_block[9-i] * ractx->gain_lpc[i];
113
114 /* block 47 of G.728 spec */
115 sum = av_clipf(sum, 0, 60);
116
117 /* block 48 of G.728 spec */
118 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
119 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
120
121 for (i=0; i < 5; i++)
122 buffer[i] = codetable[cb_coef][i] * sumsum;
123
124 sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
125
126 sum = FFMAX(sum, 5.0 / (1<<24));
127
128 /* shift and store */
129 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
130
131 gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
132
133 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
134}
135
136/**
137 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
138 *
139 * @param order filter order
140 * @param n input length
141 * @param non_rec number of non-recursive samples
142 * @param out filter output
143 * @param hist pointer to the input history of the filter
144 * @param out pointer to the non-recursive part of the output
145 * @param out2 pointer to the recursive part of the output
146 * @param window pointer to the windowing function table
147 */
148static void do_hybrid_window(RA288Context *ractx,
149 int order, int n, int non_rec, float *out,
150 float *hist, float *out2, const float *window)
151{
152 int i;
153 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
154 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
155 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
156 MAX_BACKWARD_FILTER_LEN +
157 MAX_BACKWARD_FILTER_NONREC, 16)]);
158
159 av_assert2(order>=0);
160
161 ractx->fdsp->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
162
163 convolve(buffer1, work + order , n , order);
164 convolve(buffer2, work + order + n, non_rec, order);
165
166 for (i=0; i <= order; i++) {
167 out2[i] = out2[i] * 0.5625 + buffer1[i];
168 out [i] = out2[i] + buffer2[i];
169 }
170
171 /* Multiply by the white noise correcting factor (WNCF). */
172 *out *= 257.0 / 256.0;
173}
174
175/**
176 * Backward synthesis filter, find the LPC coefficients from past speech data.
177 */
178static void backward_filter(RA288Context *ractx,
179 float *hist, float *rec, const float *window,
180 float *lpc, const float *tab,
181 int order, int n, int non_rec, int move_size)
182{
183 float temp[MAX_BACKWARD_FILTER_ORDER+1];
184
185 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
186
187 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
188 ractx->fdsp->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
189
190 memmove(hist, hist + n, move_size*sizeof(*hist));
191}
192
193static int ra288_decode_frame(AVCodecContext * avctx, void *data,
194 int *got_frame_ptr, AVPacket *avpkt)
195{
196 AVFrame *frame = data;
197 const uint8_t *buf = avpkt->data;
198 int buf_size = avpkt->size;
199 float *out;
200 int i, ret;
201 RA288Context *ractx = avctx->priv_data;
202 GetBitContext gb;
203
204 if (buf_size < avctx->block_align) {
205 av_log(avctx, AV_LOG_ERROR,
206 "Error! Input buffer is too small [%d<%d]\n",
207 buf_size, avctx->block_align);
208 return AVERROR_INVALIDDATA;
209 }
210
211 ret = init_get_bits8(&gb, buf, avctx->block_align);
212 if (ret < 0)
213 return ret;
214
215 /* get output buffer */
216 frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
217 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
218 return ret;
219 out = (float *)frame->data[0];
220
221 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
222 float gain = amptable[get_bits(&gb, 3)];
223 int cb_coef = get_bits(&gb, 6 + (i&1));
224
225 decode(ractx, gain, cb_coef);
226
227 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
228 out += RA288_BLOCK_SIZE;
229
230 if ((i & 7) == 3) {
231 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
232 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
233
234 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
235 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
236 }
237 }
238
239 *got_frame_ptr = 1;
240
241 return avctx->block_align;
242}
243
244AVCodec ff_ra_288_decoder = {
245 .name = "real_288",
246 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
247 .type = AVMEDIA_TYPE_AUDIO,
248 .id = AV_CODEC_ID_RA_288,
249 .priv_data_size = sizeof(RA288Context),
250 .init = ra288_decode_init,
251 .decode = ra288_decode_frame,
252 .close = ra288_decode_close,
253 .capabilities = AV_CODEC_CAP_DR1,
254};
255
generated by cgit at 2024-05-17 03:07:38 (GMT)