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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 "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 {
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 
63 {
64  RA288Context *ractx = avctx->priv_data;
65 
66  av_freep(&ractx->fdsp);
67 
68  return 0;
69 }
70 
72 {
73  RA288Context *ractx = avctx->priv_data;
74 
75  avctx->channels = 1;
78 
79  if (avctx->block_align <= 0) {
80  av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
81  return AVERROR_PATCHWELCOME;
82  }
83 
85  if (!ractx->fdsp)
86  return AVERROR(ENOMEM);
87 
88  return 0;
89 }
90 
91 static 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 
98 static 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  */
147 static 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];
157 
158  av_assert2(order>=0);
159 
160  ractx->fdsp->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
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  */
177 static 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 {
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))
187  ractx->fdsp->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
188 
189  memmove(hist, hist + n, move_size*sizeof(*hist));
190 }
191 
192 static 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 */
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 
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),
249  .close = ra288_decode_close,
250  .capabilities = CODEC_CAP_DR1,
251 };