Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / ra288.c
CommitLineData
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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#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
40typedef struct {
f6fa7814 41 AVFloatDSPContext *fdsp;
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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
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62static av_cold int ra288_decode_close(AVCodecContext *avctx)
63{
64 RA288Context *ractx = avctx->priv_data;
65
66 av_freep(&ractx->fdsp);
67
68 return 0;
69}
70
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71static av_cold int ra288_decode_init(AVCodecContext *avctx)
72{
73 RA288Context *ractx = avctx->priv_data;
74
75 avctx->channels = 1;
76 avctx->channel_layout = AV_CH_LAYOUT_MONO;
77 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
78
79 if (avctx->block_align <= 0) {
80 av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
81 return AVERROR_PATCHWELCOME;
82 }
83
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84 ractx->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
85 if (!ractx->fdsp)
86 return AVERROR(ENOMEM);
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87
88 return 0;
89}
90
91static void convolve(float *tgt, const float *src, int len, int n)
92{
93 for (; n >= 0; n--)
94 tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
95
96}
97
98static void decode(RA288Context *ractx, float gain, int cb_coef)
99{
100 int i;
101 double sumsum;
102 float sum, buffer[5];
103 float *block = ractx->sp_hist + 70 + 36; // current block
104 float *gain_block = ractx->gain_hist + 28;
105
106 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
107
108 /* block 46 of G.728 spec */
109 sum = 32.0;
110 for (i=0; i < 10; i++)
111 sum -= gain_block[9-i] * ractx->gain_lpc[i];
112
113 /* block 47 of G.728 spec */
114 sum = av_clipf(sum, 0, 60);
115
116 /* block 48 of G.728 spec */
117 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
118 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
119
120 for (i=0; i < 5; i++)
121 buffer[i] = codetable[cb_coef][i] * sumsum;
122
123 sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
124
125 sum = FFMAX(sum, 5.0 / (1<<24));
126
127 /* shift and store */
128 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
129
130 gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
131
132 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
133}
134
135/**
136 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
137 *
138 * @param order filter order
139 * @param n input length
140 * @param non_rec number of non-recursive samples
141 * @param out filter output
142 * @param hist pointer to the input history of the filter
143 * @param out pointer to the non-recursive part of the output
144 * @param out2 pointer to the recursive part of the output
145 * @param window pointer to the windowing function table
146 */
147static void do_hybrid_window(RA288Context *ractx,
148 int order, int n, int non_rec, float *out,
149 float *hist, float *out2, const float *window)
150{
151 int i;
152 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
153 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
154 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
155 MAX_BACKWARD_FILTER_LEN +
156 MAX_BACKWARD_FILTER_NONREC, 16)]);
157
158 av_assert2(order>=0);
159
f6fa7814 160 ractx->fdsp->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
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161
162 convolve(buffer1, work + order , n , order);
163 convolve(buffer2, work + order + n, non_rec, order);
164
165 for (i=0; i <= order; i++) {
166 out2[i] = out2[i] * 0.5625 + buffer1[i];
167 out [i] = out2[i] + buffer2[i];
168 }
169
170 /* Multiply by the white noise correcting factor (WNCF). */
171 *out *= 257.0 / 256.0;
172}
173
174/**
175 * Backward synthesis filter, find the LPC coefficients from past speech data.
176 */
177static void backward_filter(RA288Context *ractx,
178 float *hist, float *rec, const float *window,
179 float *lpc, const float *tab,
180 int order, int n, int non_rec, int move_size)
181{
182 float temp[MAX_BACKWARD_FILTER_ORDER+1];
183
184 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
185
186 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
f6fa7814 187 ractx->fdsp->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
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188
189 memmove(hist, hist + n, move_size*sizeof(*hist));
190}
191
192static int ra288_decode_frame(AVCodecContext * avctx, void *data,
193 int *got_frame_ptr, AVPacket *avpkt)
194{
195 AVFrame *frame = data;
196 const uint8_t *buf = avpkt->data;
197 int buf_size = avpkt->size;
198 float *out;
199 int i, ret;
200 RA288Context *ractx = avctx->priv_data;
201 GetBitContext gb;
202
203 if (buf_size < avctx->block_align) {
204 av_log(avctx, AV_LOG_ERROR,
205 "Error! Input buffer is too small [%d<%d]\n",
206 buf_size, avctx->block_align);
207 return AVERROR_INVALIDDATA;
208 }
209
210 /* get output buffer */
211 frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
212 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
213 return ret;
214 out = (float *)frame->data[0];
215
216 init_get_bits8(&gb, buf, avctx->block_align);
217
218 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
219 float gain = amptable[get_bits(&gb, 3)];
220 int cb_coef = get_bits(&gb, 6 + (i&1));
221
222 decode(ractx, gain, cb_coef);
223
224 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
225 out += RA288_BLOCK_SIZE;
226
227 if ((i & 7) == 3) {
228 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
229 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
230
231 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
232 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
233 }
234 }
235
236 *got_frame_ptr = 1;
237
238 return avctx->block_align;
239}
240
241AVCodec ff_ra_288_decoder = {
242 .name = "real_288",
243 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
244 .type = AVMEDIA_TYPE_AUDIO,
245 .id = AV_CODEC_ID_RA_288,
246 .priv_data_size = sizeof(RA288Context),
247 .init = ra288_decode_init,
248 .decode = ra288_decode_frame,
f6fa7814 249 .close = ra288_decode_close,
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250 .capabilities = CODEC_CAP_DR1,
251};