cosmetics: Add '0' to float constants ending in '.'.
[libav.git] / libavcodec / ra288.c
1 /*
2 * RealAudio 2.0 (28.8K)
3 * Copyright (c) 2003 the ffmpeg project
4 *
5 * This file is part of Libav.
6 *
7 * Libav 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 * Libav 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 Libav; 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 #include "avcodec.h"
26 #include "internal.h"
27 #define BITSTREAM_READER_LE
28 #include "get_bits.h"
29 #include "ra288.h"
30 #include "lpc.h"
31 #include "celp_filters.h"
32
33 #define MAX_BACKWARD_FILTER_ORDER 36
34 #define MAX_BACKWARD_FILTER_LEN 40
35 #define MAX_BACKWARD_FILTER_NONREC 35
36
37 #define RA288_BLOCK_SIZE 5
38 #define RA288_BLOCKS_PER_FRAME 32
39
40 typedef struct {
41 AVFloatDSPContext fdsp;
42 DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
43 DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
44
45 /** speech data history (spec: SB).
46 * Its first 70 coefficients are updated only at backward filtering.
47 */
48 float sp_hist[111];
49
50 /// speech part of the gain autocorrelation (spec: REXP)
51 float sp_rec[37];
52
53 /** log-gain history (spec: SBLG).
54 * Its first 28 coefficients are updated only at backward filtering.
55 */
56 float gain_hist[38];
57
58 /// recursive part of the gain autocorrelation (spec: REXPLG)
59 float gain_rec[11];
60 } RA288Context;
61
62 static av_cold int ra288_decode_init(AVCodecContext *avctx)
63 {
64 RA288Context *ractx = avctx->priv_data;
65
66 avctx->channels = 1;
67 avctx->channel_layout = AV_CH_LAYOUT_MONO;
68 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
69
70 avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
71
72 return 0;
73 }
74
75 static void convolve(float *tgt, const float *src, int len, int n)
76 {
77 for (; n >= 0; n--)
78 tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
79
80 }
81
82 static void decode(RA288Context *ractx, float gain, int cb_coef)
83 {
84 int i;
85 double sumsum;
86 float sum, buffer[5];
87 float *block = ractx->sp_hist + 70 + 36; // current block
88 float *gain_block = ractx->gain_hist + 28;
89
90 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
91
92 /* block 46 of G.728 spec */
93 sum = 32.0;
94 for (i=0; i < 10; i++)
95 sum -= gain_block[9-i] * ractx->gain_lpc[i];
96
97 /* block 47 of G.728 spec */
98 sum = av_clipf(sum, 0, 60);
99
100 /* block 48 of G.728 spec */
101 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
102 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
103
104 for (i=0; i < 5; i++)
105 buffer[i] = codetable[cb_coef][i] * sumsum;
106
107 sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.0);
108
109 sum = FFMAX(sum, 1);
110
111 /* shift and store */
112 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
113
114 gain_block[9] = 10 * log10(sum) - 32;
115
116 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
117 }
118
119 /**
120 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
121 *
122 * @param order filter order
123 * @param n input length
124 * @param non_rec number of non-recursive samples
125 * @param out filter output
126 * @param hist pointer to the input history of the filter
127 * @param out pointer to the non-recursive part of the output
128 * @param out2 pointer to the recursive part of the output
129 * @param window pointer to the windowing function table
130 */
131 static void do_hybrid_window(RA288Context *ractx,
132 int order, int n, int non_rec, float *out,
133 float *hist, float *out2, const float *window)
134 {
135 int i;
136 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
137 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
138 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
139 MAX_BACKWARD_FILTER_LEN +
140 MAX_BACKWARD_FILTER_NONREC, 16)]);
141
142 ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
143
144 convolve(buffer1, work + order , n , order);
145 convolve(buffer2, work + order + n, non_rec, order);
146
147 for (i=0; i <= order; i++) {
148 out2[i] = out2[i] * 0.5625 + buffer1[i];
149 out [i] = out2[i] + buffer2[i];
150 }
151
152 /* Multiply by the white noise correcting factor (WNCF). */
153 *out *= 257.0 / 256.0;
154 }
155
156 /**
157 * Backward synthesis filter, find the LPC coefficients from past speech data.
158 */
159 static void backward_filter(RA288Context *ractx,
160 float *hist, float *rec, const float *window,
161 float *lpc, const float *tab,
162 int order, int n, int non_rec, int move_size)
163 {
164 float temp[MAX_BACKWARD_FILTER_ORDER+1];
165
166 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
167
168 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
169 ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
170
171 memmove(hist, hist + n, move_size*sizeof(*hist));
172 }
173
174 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
175 int *got_frame_ptr, AVPacket *avpkt)
176 {
177 AVFrame *frame = data;
178 const uint8_t *buf = avpkt->data;
179 int buf_size = avpkt->size;
180 float *out;
181 int i, ret;
182 RA288Context *ractx = avctx->priv_data;
183 GetBitContext gb;
184
185 if (buf_size < avctx->block_align) {
186 av_log(avctx, AV_LOG_ERROR,
187 "Error! Input buffer is too small [%d<%d]\n",
188 buf_size, avctx->block_align);
189 return AVERROR_INVALIDDATA;
190 }
191
192 /* get output buffer */
193 frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
194 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
195 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
196 return ret;
197 }
198 out = (float *)frame->data[0];
199
200 init_get_bits(&gb, buf, avctx->block_align * 8);
201
202 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
203 float gain = amptable[get_bits(&gb, 3)];
204 int cb_coef = get_bits(&gb, 6 + (i&1));
205
206 decode(ractx, gain, cb_coef);
207
208 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
209 out += RA288_BLOCK_SIZE;
210
211 if ((i & 7) == 3) {
212 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
213 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
214
215 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
216 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
217 }
218 }
219
220 *got_frame_ptr = 1;
221
222 return avctx->block_align;
223 }
224
225 AVCodec ff_ra_288_decoder = {
226 .name = "real_288",
227 .type = AVMEDIA_TYPE_AUDIO,
228 .id = AV_CODEC_ID_RA_288,
229 .priv_data_size = sizeof(RA288Context),
230 .init = ra288_decode_init,
231 .decode = ra288_decode_frame,
232 .capabilities = CODEC_CAP_DR1,
233 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
234 };