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ra288.c
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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 
23 #include "libavutil/float_dsp.h"
24 #include "avcodec.h"
25 #include "internal.h"
26 #define BITSTREAM_READER_LE
27 #include "get_bits.h"
28 #include "ra288.h"
29 #include "lpc.h"
30 #include "celp_filters.h"
31 
32 #define MAX_BACKWARD_FILTER_ORDER 36
33 #define MAX_BACKWARD_FILTER_LEN 40
34 #define MAX_BACKWARD_FILTER_NONREC 35
35 
36 #define RA288_BLOCK_SIZE 5
37 #define RA288_BLOCKS_PER_FRAME 32
38 
39 typedef struct {
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 
64 {
65  RA288Context *ractx = avctx->priv_data;
66 
67  avctx->channels = 1;
70 
71  if (avctx->block_align <= 0) {
72  av_log_ask_for_sample(avctx, "unsupported block align\n");
73  return AVERROR_PATCHWELCOME;
74  }
75 
77 
79  avctx->coded_frame = &ractx->frame;
80 
81  return 0;
82 }
83 
84 static void convolve(float *tgt, const float *src, int len, int n)
85 {
86  for (; n >= 0; n--)
87  tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
88 
89 }
90 
91 static void decode(RA288Context *ractx, float gain, int cb_coef)
92 {
93  int i;
94  double sumsum;
95  float sum, buffer[5];
96  float *block = ractx->sp_hist + 70 + 36; // current block
97  float *gain_block = ractx->gain_hist + 28;
98 
99  memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
100 
101  /* block 46 of G.728 spec */
102  sum = 32.;
103  for (i=0; i < 10; i++)
104  sum -= gain_block[9-i] * ractx->gain_lpc[i];
105 
106  /* block 47 of G.728 spec */
107  sum = av_clipf(sum, 0, 60);
108 
109  /* block 48 of G.728 spec */
110  /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
111  sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
112 
113  for (i=0; i < 5; i++)
114  buffer[i] = codetable[cb_coef][i] * sumsum;
115 
116  sum = ff_scalarproduct_float_c(buffer, buffer, 5);
117 
118  sum = FFMAX(sum, 5. / (1<<24));
119 
120  /* shift and store */
121  memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
122 
123  gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
124 
125  ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
126 }
127 
128 /**
129  * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
130  *
131  * @param order filter order
132  * @param n input length
133  * @param non_rec number of non-recursive samples
134  * @param out filter output
135  * @param hist pointer to the input history of the filter
136  * @param out pointer to the non-recursive part of the output
137  * @param out2 pointer to the recursive part of the output
138  * @param window pointer to the windowing function table
139  */
140 static void do_hybrid_window(RA288Context *ractx,
141  int order, int n, int non_rec, float *out,
142  float *hist, float *out2, const float *window)
143 {
144  int i;
145  float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
146  float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
150 
151  av_assert2(order>=0);
152 
153  ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
154 
155  convolve(buffer1, work + order , n , order);
156  convolve(buffer2, work + order + n, non_rec, order);
157 
158  for (i=0; i <= order; i++) {
159  out2[i] = out2[i] * 0.5625 + buffer1[i];
160  out [i] = out2[i] + buffer2[i];
161  }
162 
163  /* Multiply by the white noise correcting factor (WNCF). */
164  *out *= 257./256.;
165 }
166 
167 /**
168  * Backward synthesis filter, find the LPC coefficients from past speech data.
169  */
170 static void backward_filter(RA288Context *ractx,
171  float *hist, float *rec, const float *window,
172  float *lpc, const float *tab,
173  int order, int n, int non_rec, int move_size)
174 {
176 
177  do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
178 
179  if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
180  ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
181 
182  memmove(hist, hist + n, move_size*sizeof(*hist));
183 }
184 
185 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
186  int *got_frame_ptr, AVPacket *avpkt)
187 {
188  const uint8_t *buf = avpkt->data;
189  int buf_size = avpkt->size;
190  float *out;
191  int i, ret;
192  RA288Context *ractx = avctx->priv_data;
193  GetBitContext gb;
194 
195  if (buf_size < avctx->block_align) {
196  av_log(avctx, AV_LOG_ERROR,
197  "Error! Input buffer is too small [%d<%d]\n",
198  buf_size, avctx->block_align);
199  return AVERROR_INVALIDDATA;
200  }
201 
202  /* get output buffer */
204  if ((ret = ff_get_buffer(avctx, &ractx->frame)) < 0) {
205  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
206  return ret;
207  }
208  out = (float *)ractx->frame.data[0];
209 
210  init_get_bits(&gb, buf, avctx->block_align * 8);
211 
212  for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
213  float gain = amptable[get_bits(&gb, 3)];
214  int cb_coef = get_bits(&gb, 6 + (i&1));
215 
216  decode(ractx, gain, cb_coef);
217 
218  memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
219  out += RA288_BLOCK_SIZE;
220 
221  if ((i & 7) == 3) {
222  backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
223  ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
224 
225  backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
226  ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
227  }
228  }
229 
230  *got_frame_ptr = 1;
231  *(AVFrame *)data = ractx->frame;
232 
233  return avctx->block_align;
234 }
235 
237  .name = "real_288",
238  .type = AVMEDIA_TYPE_AUDIO,
239  .id = AV_CODEC_ID_RA_288,
240  .priv_data_size = sizeof(RA288Context),
243  .capabilities = CODEC_CAP_DR1,
244  .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
245 };