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g723_1.c
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1 /*
2  * G.723.1 compatible decoder
3  * Copyright (c) 2006 Benjamin Larsson
4  * Copyright (c) 2010 Mohamed Naufal Basheer
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 /**
24  * @file
25  * G.723.1 compatible decoder
26  */
27 
28 #define BITSTREAM_READER_LE
29 #include "libavutil/channel_layout.h"
30 #include "libavutil/mem.h"
31 #include "libavutil/opt.h"
32 #include "avcodec.h"
33 #include "internal.h"
34 #include "get_bits.h"
35 #include "acelp_vectors.h"
36 #include "celp_filters.h"
37 #include "celp_math.h"
38 #include "g723_1_data.h"
39 #include "internal.h"
40 
41 #define CNG_RANDOM_SEED 12345
42 
43 typedef struct g723_1_context {
44  AVClass *class;
45 
46  G723_1_Subframe subframe[4];
47  enum FrameType cur_frame_type;
48  enum FrameType past_frame_type;
49  enum Rate cur_rate;
50  uint8_t lsp_index[LSP_BANDS];
51  int pitch_lag[2];
52  int erased_frames;
53 
54  int16_t prev_lsp[LPC_ORDER];
55  int16_t sid_lsp[LPC_ORDER];
56  int16_t prev_excitation[PITCH_MAX];
57  int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
58  int16_t synth_mem[LPC_ORDER];
59  int16_t fir_mem[LPC_ORDER];
60  int iir_mem[LPC_ORDER];
61 
62  int random_seed;
63  int cng_random_seed;
64  int interp_index;
65  int interp_gain;
66  int sid_gain;
67  int cur_gain;
68  int reflection_coef;
69  int pf_gain; ///< formant postfilter
70  ///< gain scaling unit memory
71  int postfilter;
72 
73  int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
74  int16_t prev_data[HALF_FRAME_LEN];
75  int16_t prev_weight_sig[PITCH_MAX];
76 
77 
78  int16_t hpf_fir_mem; ///< highpass filter fir
79  int hpf_iir_mem; ///< and iir memories
80  int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir
81  int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories
82 
83  int16_t harmonic_mem[PITCH_MAX];
84 } G723_1_Context;
85 
86 static av_cold int g723_1_decode_init(AVCodecContext *avctx)
87 {
88  G723_1_Context *p = avctx->priv_data;
89 
90  avctx->channel_layout = AV_CH_LAYOUT_MONO;
91  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
92  avctx->channels = 1;
93  p->pf_gain = 1 << 12;
94 
95  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
96  memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
97 
98  p->cng_random_seed = CNG_RANDOM_SEED;
99  p->past_frame_type = SID_FRAME;
100 
101  return 0;
102 }
103 
104 /**
105  * Unpack the frame into parameters.
106  *
107  * @param p the context
108  * @param buf pointer to the input buffer
109  * @param buf_size size of the input buffer
110  */
111 static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
112  int buf_size)
113 {
114  GetBitContext gb;
115  int ad_cb_len;
116  int temp, info_bits, i;
117 
118  init_get_bits(&gb, buf, buf_size * 8);
119 
120  /* Extract frame type and rate info */
121  info_bits = get_bits(&gb, 2);
122 
123  if (info_bits == 3) {
124  p->cur_frame_type = UNTRANSMITTED_FRAME;
125  return 0;
126  }
127 
128  /* Extract 24 bit lsp indices, 8 bit for each band */
129  p->lsp_index[2] = get_bits(&gb, 8);
130  p->lsp_index[1] = get_bits(&gb, 8);
131  p->lsp_index[0] = get_bits(&gb, 8);
132 
133  if (info_bits == 2) {
134  p->cur_frame_type = SID_FRAME;
135  p->subframe[0].amp_index = get_bits(&gb, 6);
136  return 0;
137  }
138 
139  /* Extract the info common to both rates */
140  p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
141  p->cur_frame_type = ACTIVE_FRAME;
142 
143  p->pitch_lag[0] = get_bits(&gb, 7);
144  if (p->pitch_lag[0] > 123) /* test if forbidden code */
145  return -1;
146  p->pitch_lag[0] += PITCH_MIN;
147  p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
148 
149  p->pitch_lag[1] = get_bits(&gb, 7);
150  if (p->pitch_lag[1] > 123)
151  return -1;
152  p->pitch_lag[1] += PITCH_MIN;
153  p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
154  p->subframe[0].ad_cb_lag = 1;
155  p->subframe[2].ad_cb_lag = 1;
156 
157  for (i = 0; i < SUBFRAMES; i++) {
158  /* Extract combined gain */
159  temp = get_bits(&gb, 12);
160  ad_cb_len = 170;
161  p->subframe[i].dirac_train = 0;
162  if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
163  p->subframe[i].dirac_train = temp >> 11;
164  temp &= 0x7FF;
165  ad_cb_len = 85;
166  }
167  p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
168  if (p->subframe[i].ad_cb_gain < ad_cb_len) {
169  p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
170  GAIN_LEVELS;
171  } else {
172  return -1;
173  }
174  }
175 
176  p->subframe[0].grid_index = get_bits1(&gb);
177  p->subframe[1].grid_index = get_bits1(&gb);
178  p->subframe[2].grid_index = get_bits1(&gb);
179  p->subframe[3].grid_index = get_bits1(&gb);
180 
181  if (p->cur_rate == RATE_6300) {
182  skip_bits1(&gb); /* skip reserved bit */
183 
184  /* Compute pulse_pos index using the 13-bit combined position index */
185  temp = get_bits(&gb, 13);
186  p->subframe[0].pulse_pos = temp / 810;
187 
188  temp -= p->subframe[0].pulse_pos * 810;
189  p->subframe[1].pulse_pos = FASTDIV(temp, 90);
190 
191  temp -= p->subframe[1].pulse_pos * 90;
192  p->subframe[2].pulse_pos = FASTDIV(temp, 9);
193  p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
194 
195  p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
196  get_bits(&gb, 16);
197  p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
198  get_bits(&gb, 14);
199  p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
200  get_bits(&gb, 16);
201  p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
202  get_bits(&gb, 14);
203 
204  p->subframe[0].pulse_sign = get_bits(&gb, 6);
205  p->subframe[1].pulse_sign = get_bits(&gb, 5);
206  p->subframe[2].pulse_sign = get_bits(&gb, 6);
207  p->subframe[3].pulse_sign = get_bits(&gb, 5);
208  } else { /* 5300 bps */
209  p->subframe[0].pulse_pos = get_bits(&gb, 12);
210  p->subframe[1].pulse_pos = get_bits(&gb, 12);
211  p->subframe[2].pulse_pos = get_bits(&gb, 12);
212  p->subframe[3].pulse_pos = get_bits(&gb, 12);
213 
214  p->subframe[0].pulse_sign = get_bits(&gb, 4);
215  p->subframe[1].pulse_sign = get_bits(&gb, 4);
216  p->subframe[2].pulse_sign = get_bits(&gb, 4);
217  p->subframe[3].pulse_sign = get_bits(&gb, 4);
218  }
219 
220  return 0;
221 }
222 
223 /**
224  * Bitexact implementation of sqrt(val/2).
225  */
226 static int16_t square_root(unsigned val)
227 {
228  av_assert2(!(val & 0x80000000));
229 
230  return (ff_sqrt(val << 1) >> 1) & (~1);
231 }
232 
233 /**
234  * Calculate the number of left-shifts required for normalizing the input.
235  *
236  * @param num input number
237  * @param width width of the input, 15 or 31 bits
238  */
239 static int normalize_bits(int num, int width)
240 {
241  return width - av_log2(num) - 1;
242 }
243 
244 #define normalize_bits_int16(num) normalize_bits(num, 15)
245 #define normalize_bits_int32(num) normalize_bits(num, 31)
246 
247 /**
248  * Scale vector contents based on the largest of their absolutes.
249  */
250 static int scale_vector(int16_t *dst, const int16_t *vector, int length)
251 {
252  int bits, max = 0;
253  int i;
254 
255  for (i = 0; i < length; i++)
256  max |= FFABS(vector[i]);
257 
258  bits= 14 - av_log2_16bit(max);
259  bits= FFMAX(bits, 0);
260 
261  for (i = 0; i < length; i++)
262  dst[i] = vector[i] << bits >> 3;
263 
264  return bits - 3;
265 }
266 
267 /**
268  * Perform inverse quantization of LSP frequencies.
269  *
270  * @param cur_lsp the current LSP vector
271  * @param prev_lsp the previous LSP vector
272  * @param lsp_index VQ indices
273  * @param bad_frame bad frame flag
274  */
275 static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
276  uint8_t *lsp_index, int bad_frame)
277 {
278  int min_dist, pred;
279  int i, j, temp, stable;
280 
281  /* Check for frame erasure */
282  if (!bad_frame) {
283  min_dist = 0x100;
284  pred = 12288;
285  } else {
286  min_dist = 0x200;
287  pred = 23552;
288  lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
289  }
290 
291  /* Get the VQ table entry corresponding to the transmitted index */
292  cur_lsp[0] = lsp_band0[lsp_index[0]][0];
293  cur_lsp[1] = lsp_band0[lsp_index[0]][1];
294  cur_lsp[2] = lsp_band0[lsp_index[0]][2];
295  cur_lsp[3] = lsp_band1[lsp_index[1]][0];
296  cur_lsp[4] = lsp_band1[lsp_index[1]][1];
297  cur_lsp[5] = lsp_band1[lsp_index[1]][2];
298  cur_lsp[6] = lsp_band2[lsp_index[2]][0];
299  cur_lsp[7] = lsp_band2[lsp_index[2]][1];
300  cur_lsp[8] = lsp_band2[lsp_index[2]][2];
301  cur_lsp[9] = lsp_band2[lsp_index[2]][3];
302 
303  /* Add predicted vector & DC component to the previously quantized vector */
304  for (i = 0; i < LPC_ORDER; i++) {
305  temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
306  cur_lsp[i] += dc_lsp[i] + temp;
307  }
308 
309  for (i = 0; i < LPC_ORDER; i++) {
310  cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);
311  cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);
312 
313  /* Stability check */
314  for (j = 1; j < LPC_ORDER; j++) {
315  temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
316  if (temp > 0) {
317  temp >>= 1;
318  cur_lsp[j - 1] -= temp;
319  cur_lsp[j] += temp;
320  }
321  }
322  stable = 1;
323  for (j = 1; j < LPC_ORDER; j++) {
324  temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
325  if (temp > 0) {
326  stable = 0;
327  break;
328  }
329  }
330  if (stable)
331  break;
332  }
333  if (!stable)
334  memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));
335 }
336 
337 /**
338  * Bitexact implementation of 2ab scaled by 1/2^16.
339  *
340  * @param a 32 bit multiplicand
341  * @param b 16 bit multiplier
342  */
343 #define MULL2(a, b) \
344  MULL(a,b,15)
345 
346 /**
347  * Convert LSP frequencies to LPC coefficients.
348  *
349  * @param lpc buffer for LPC coefficients
350  */
351 static void lsp2lpc(int16_t *lpc)
352 {
353  int f1[LPC_ORDER / 2 + 1];
354  int f2[LPC_ORDER / 2 + 1];
355  int i, j;
356 
357  /* Calculate negative cosine */
358  for (j = 0; j < LPC_ORDER; j++) {
359  int index = (lpc[j] >> 7) & 0x1FF;
360  int offset = lpc[j] & 0x7f;
361  int temp1 = cos_tab[index] << 16;
362  int temp2 = (cos_tab[index + 1] - cos_tab[index]) *
363  ((offset << 8) + 0x80) << 1;
364 
365  lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
366  }
367 
368  /*
369  * Compute sum and difference polynomial coefficients
370  * (bitexact alternative to lsp2poly() in lsp.c)
371  */
372  /* Initialize with values in Q28 */
373  f1[0] = 1 << 28;
374  f1[1] = (lpc[0] << 14) + (lpc[2] << 14);
375  f1[2] = lpc[0] * lpc[2] + (2 << 28);
376 
377  f2[0] = 1 << 28;
378  f2[1] = (lpc[1] << 14) + (lpc[3] << 14);
379  f2[2] = lpc[1] * lpc[3] + (2 << 28);
380 
381  /*
382  * Calculate and scale the coefficients by 1/2 in
383  * each iteration for a final scaling factor of Q25
384  */
385  for (i = 2; i < LPC_ORDER / 2; i++) {
386  f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);
387  f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);
388 
389  for (j = i; j >= 2; j--) {
390  f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +
391  (f1[j] >> 1) + (f1[j - 2] >> 1);
392  f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +
393  (f2[j] >> 1) + (f2[j - 2] >> 1);
394  }
395 
396  f1[0] >>= 1;
397  f2[0] >>= 1;
398  f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1;
399  f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
400  }
401 
402  /* Convert polynomial coefficients to LPC coefficients */
403  for (i = 0; i < LPC_ORDER / 2; i++) {
404  int64_t ff1 = f1[i + 1] + f1[i];
405  int64_t ff2 = f2[i + 1] - f2[i];
406 
407  lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;
408  lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +
409  (1 << 15)) >> 16;
410  }
411 }
412 
413 /**
414  * Quantize LSP frequencies by interpolation and convert them to
415  * the corresponding LPC coefficients.
416  *
417  * @param lpc buffer for LPC coefficients
418  * @param cur_lsp the current LSP vector
419  * @param prev_lsp the previous LSP vector
420  */
421 static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
422 {
423  int i;
424  int16_t *lpc_ptr = lpc;
425 
426  /* cur_lsp * 0.25 + prev_lsp * 0.75 */
427  ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,
428  4096, 12288, 1 << 13, 14, LPC_ORDER);
429  ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,
430  8192, 8192, 1 << 13, 14, LPC_ORDER);
431  ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,
432  12288, 4096, 1 << 13, 14, LPC_ORDER);
433  memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc));
434 
435  for (i = 0; i < SUBFRAMES; i++) {
436  lsp2lpc(lpc_ptr);
437  lpc_ptr += LPC_ORDER;
438  }
439 }
440 
441 /**
442  * Generate a train of dirac functions with period as pitch lag.
443  */
444 static void gen_dirac_train(int16_t *buf, int pitch_lag)
445 {
446  int16_t vector[SUBFRAME_LEN];
447  int i, j;
448 
449  memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
450  for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
451  for (j = 0; j < SUBFRAME_LEN - i; j++)
452  buf[i + j] += vector[j];
453  }
454 }
455 
456 /**
457  * Generate fixed codebook excitation vector.
458  *
459  * @param vector decoded excitation vector
460  * @param subfrm current subframe
461  * @param cur_rate current bitrate
462  * @param pitch_lag closed loop pitch lag
463  * @param index current subframe index
464  */
465 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
466  enum Rate cur_rate, int pitch_lag, int index)
467 {
468  int temp, i, j;
469 
470  memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
471 
472  if (cur_rate == RATE_6300) {
473  if (subfrm->pulse_pos >= max_pos[index])
474  return;
475 
476  /* Decode amplitudes and positions */
477  j = PULSE_MAX - pulses[index];
478  temp = subfrm->pulse_pos;
479  for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
480  temp -= combinatorial_table[j][i];
481  if (temp >= 0)
482  continue;
483  temp += combinatorial_table[j++][i];
484  if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
485  vector[subfrm->grid_index + GRID_SIZE * i] =
486  -fixed_cb_gain[subfrm->amp_index];
487  } else {
488  vector[subfrm->grid_index + GRID_SIZE * i] =
489  fixed_cb_gain[subfrm->amp_index];
490  }
491  if (j == PULSE_MAX)
492  break;
493  }
494  if (subfrm->dirac_train == 1)
495  gen_dirac_train(vector, pitch_lag);
496  } else { /* 5300 bps */
497  int cb_gain = fixed_cb_gain[subfrm->amp_index];
498  int cb_shift = subfrm->grid_index;
499  int cb_sign = subfrm->pulse_sign;
500  int cb_pos = subfrm->pulse_pos;
501  int offset, beta, lag;
502 
503  for (i = 0; i < 8; i += 2) {
504  offset = ((cb_pos & 7) << 3) + cb_shift + i;
505  vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
506  cb_pos >>= 3;
507  cb_sign >>= 1;
508  }
509 
510  /* Enhance harmonic components */
511  lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
512  subfrm->ad_cb_lag - 1;
513  beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
514 
515  if (lag < SUBFRAME_LEN - 2) {
516  for (i = lag; i < SUBFRAME_LEN; i++)
517  vector[i] += beta * vector[i - lag] >> 15;
518  }
519  }
520 }
521 
522 /**
523  * Get delayed contribution from the previous excitation vector.
524  */
525 static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
526 {
527  int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
528  int i;
529 
530  residual[0] = prev_excitation[offset];
531  residual[1] = prev_excitation[offset + 1];
532 
533  offset += 2;
534  for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
535  residual[i] = prev_excitation[offset + (i - 2) % lag];
536 }
537 
538 static int dot_product(const int16_t *a, const int16_t *b, int length)
539 {
540  int sum = ff_dot_product(a,b,length);
541  return av_sat_add32(sum, sum);
542 }
543 
544 /**
545  * Generate adaptive codebook excitation.
546  */
547 static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
548  int pitch_lag, G723_1_Subframe *subfrm,
549  enum Rate cur_rate)
550 {
551  int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
552  const int16_t *cb_ptr;
553  int lag = pitch_lag + subfrm->ad_cb_lag - 1;
554 
555  int i;
556  int sum;
557 
558  get_residual(residual, prev_excitation, lag);
559 
560  /* Select quantization table */
561  if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {
562  cb_ptr = adaptive_cb_gain85;
563  } else
564  cb_ptr = adaptive_cb_gain170;
565 
566  /* Calculate adaptive vector */
567  cb_ptr += subfrm->ad_cb_gain * 20;
568  for (i = 0; i < SUBFRAME_LEN; i++) {
569  sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
570  vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
571  }
572 }
573 
574 /**
575  * Estimate maximum auto-correlation around pitch lag.
576  *
577  * @param buf buffer with offset applied
578  * @param offset offset of the excitation vector
579  * @param ccr_max pointer to the maximum auto-correlation
580  * @param pitch_lag decoded pitch lag
581  * @param length length of autocorrelation
582  * @param dir forward lag(1) / backward lag(-1)
583  */
584 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
585  int pitch_lag, int length, int dir)
586 {
587  int limit, ccr, lag = 0;
588  int i;
589 
590  pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
591  if (dir > 0)
592  limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
593  else
594  limit = pitch_lag + 3;
595 
596  for (i = pitch_lag - 3; i <= limit; i++) {
597  ccr = dot_product(buf, buf + dir * i, length);
598 
599  if (ccr > *ccr_max) {
600  *ccr_max = ccr;
601  lag = i;
602  }
603  }
604  return lag;
605 }
606 
607 /**
608  * Calculate pitch postfilter optimal and scaling gains.
609  *
610  * @param lag pitch postfilter forward/backward lag
611  * @param ppf pitch postfilter parameters
612  * @param cur_rate current bitrate
613  * @param tgt_eng target energy
614  * @param ccr cross-correlation
615  * @param res_eng residual energy
616  */
617 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
618  int tgt_eng, int ccr, int res_eng)
619 {
620  int pf_residual; /* square of postfiltered residual */
621  int temp1, temp2;
622 
623  ppf->index = lag;
624 
625  temp1 = tgt_eng * res_eng >> 1;
626  temp2 = ccr * ccr << 1;
627 
628  if (temp2 > temp1) {
629  if (ccr >= res_eng) {
630  ppf->opt_gain = ppf_gain_weight[cur_rate];
631  } else {
632  ppf->opt_gain = (ccr << 15) / res_eng *
633  ppf_gain_weight[cur_rate] >> 15;
634  }
635  /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
636  temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
637  temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
638  pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
639 
640  if (tgt_eng >= pf_residual << 1) {
641  temp1 = 0x7fff;
642  } else {
643  temp1 = (tgt_eng << 14) / pf_residual;
644  }
645 
646  /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
647  ppf->sc_gain = square_root(temp1 << 16);
648  } else {
649  ppf->opt_gain = 0;
650  ppf->sc_gain = 0x7fff;
651  }
652 
653  ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
654 }
655 
656 /**
657  * Calculate pitch postfilter parameters.
658  *
659  * @param p the context
660  * @param offset offset of the excitation vector
661  * @param pitch_lag decoded pitch lag
662  * @param ppf pitch postfilter parameters
663  * @param cur_rate current bitrate
664  */
665 static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
666  PPFParam *ppf, enum Rate cur_rate)
667 {
668 
669  int16_t scale;
670  int i;
671  int temp1, temp2;
672 
673  /*
674  * 0 - target energy
675  * 1 - forward cross-correlation
676  * 2 - forward residual energy
677  * 3 - backward cross-correlation
678  * 4 - backward residual energy
679  */
680  int energy[5] = {0, 0, 0, 0, 0};
681  int16_t *buf = p->audio + LPC_ORDER + offset;
682  int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
683  SUBFRAME_LEN, 1);
684  int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
685  SUBFRAME_LEN, -1);
686 
687  ppf->index = 0;
688  ppf->opt_gain = 0;
689  ppf->sc_gain = 0x7fff;
690 
691  /* Case 0, Section 3.6 */
692  if (!back_lag && !fwd_lag)
693  return;
694 
695  /* Compute target energy */
696  energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
697 
698  /* Compute forward residual energy */
699  if (fwd_lag)
700  energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
701 
702  /* Compute backward residual energy */
703  if (back_lag)
704  energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);
705 
706  /* Normalize and shorten */
707  temp1 = 0;
708  for (i = 0; i < 5; i++)
709  temp1 = FFMAX(energy[i], temp1);
710 
711  scale = normalize_bits(temp1, 31);
712  for (i = 0; i < 5; i++)
713  energy[i] = (energy[i] << scale) >> 16;
714 
715  if (fwd_lag && !back_lag) { /* Case 1 */
716  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
717  energy[2]);
718  } else if (!fwd_lag) { /* Case 2 */
719  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
720  energy[4]);
721  } else { /* Case 3 */
722 
723  /*
724  * Select the largest of energy[1]^2/energy[2]
725  * and energy[3]^2/energy[4]
726  */
727  temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
728  temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
729  if (temp1 >= temp2) {
730  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
731  energy[2]);
732  } else {
733  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
734  energy[4]);
735  }
736  }
737 }
738 
739 /**
740  * Classify frames as voiced/unvoiced.
741  *
742  * @param p the context
743  * @param pitch_lag decoded pitch_lag
744  * @param exc_eng excitation energy estimation
745  * @param scale scaling factor of exc_eng
746  *
747  * @return residual interpolation index if voiced, 0 otherwise
748  */
749 static int comp_interp_index(G723_1_Context *p, int pitch_lag,
750  int *exc_eng, int *scale)
751 {
752  int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
753  int16_t *buf = p->audio + LPC_ORDER;
754 
755  int index, ccr, tgt_eng, best_eng, temp;
756 
757  *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
758  buf += offset;
759 
760  /* Compute maximum backward cross-correlation */
761  ccr = 0;
762  index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
763  ccr = av_sat_add32(ccr, 1 << 15) >> 16;
764 
765  /* Compute target energy */
766  tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2);
767  *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
768 
769  if (ccr <= 0)
770  return 0;
771 
772  /* Compute best energy */
773  best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);
774  best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
775 
776  temp = best_eng * *exc_eng >> 3;
777 
778  if (temp < ccr * ccr) {
779  return index;
780  } else
781  return 0;
782 }
783 
784 /**
785  * Peform residual interpolation based on frame classification.
786  *
787  * @param buf decoded excitation vector
788  * @param out output vector
789  * @param lag decoded pitch lag
790  * @param gain interpolated gain
791  * @param rseed seed for random number generator
792  */
793 static void residual_interp(int16_t *buf, int16_t *out, int lag,
794  int gain, int *rseed)
795 {
796  int i;
797  if (lag) { /* Voiced */
798  int16_t *vector_ptr = buf + PITCH_MAX;
799  /* Attenuate */
800  for (i = 0; i < lag; i++)
801  out[i] = vector_ptr[i - lag] * 3 >> 2;
802  av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
803  (FRAME_LEN - lag) * sizeof(*out));
804  } else { /* Unvoiced */
805  for (i = 0; i < FRAME_LEN; i++) {
806  *rseed = *rseed * 521 + 259;
807  out[i] = gain * *rseed >> 15;
808  }
809  memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
810  }
811 }
812 
813 /**
814  * Perform IIR filtering.
815  *
816  * @param fir_coef FIR coefficients
817  * @param iir_coef IIR coefficients
818  * @param src source vector
819  * @param dest destination vector
820  * @param width width of the output, 16 bits(0) / 32 bits(1)
821  */
822 #define iir_filter(fir_coef, iir_coef, src, dest, width)\
823 {\
824  int m, n;\
825  int res_shift = 16 & ~-(width);\
826  int in_shift = 16 - res_shift;\
827 \
828  for (m = 0; m < SUBFRAME_LEN; m++) {\
829  int64_t filter = 0;\
830  for (n = 1; n <= LPC_ORDER; n++) {\
831  filter -= (fir_coef)[n - 1] * (src)[m - n] -\
832  (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
833  }\
834 \
835  (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\
836  (1 << 15)) >> res_shift;\
837  }\
838 }
839 
840 /**
841  * Adjust gain of postfiltered signal.
842  *
843  * @param p the context
844  * @param buf postfiltered output vector
845  * @param energy input energy coefficient
846  */
847 static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
848 {
849  int num, denom, gain, bits1, bits2;
850  int i;
851 
852  num = energy;
853  denom = 0;
854  for (i = 0; i < SUBFRAME_LEN; i++) {
855  int temp = buf[i] >> 2;
856  temp *= temp;
857  denom = av_sat_dadd32(denom, temp);
858  }
859 
860  if (num && denom) {
861  bits1 = normalize_bits(num, 31);
862  bits2 = normalize_bits(denom, 31);
863  num = num << bits1 >> 1;
864  denom <<= bits2;
865 
866  bits2 = 5 + bits1 - bits2;
867  bits2 = FFMAX(0, bits2);
868 
869  gain = (num >> 1) / (denom >> 16);
870  gain = square_root(gain << 16 >> bits2);
871  } else {
872  gain = 1 << 12;
873  }
874 
875  for (i = 0; i < SUBFRAME_LEN; i++) {
876  p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
877  buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
878  (1 << 10)) >> 11);
879  }
880 }
881 
882 /**
883  * Perform formant filtering.
884  *
885  * @param p the context
886  * @param lpc quantized lpc coefficients
887  * @param buf input buffer
888  * @param dst output buffer
889  */
890 static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
891  int16_t *buf, int16_t *dst)
892 {
893  int16_t filter_coef[2][LPC_ORDER];
894  int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
895  int i, j, k;
896 
897  memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
898  memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
899 
900  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
901  for (k = 0; k < LPC_ORDER; k++) {
902  filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
903  (1 << 14)) >> 15;
904  filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
905  (1 << 14)) >> 15;
906  }
907  iir_filter(filter_coef[0], filter_coef[1], buf + i,
908  filter_signal + i, 1);
909  lpc += LPC_ORDER;
910  }
911 
912  memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
913  memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
914 
915  buf += LPC_ORDER;
916  signal_ptr = filter_signal + LPC_ORDER;
917  for (i = 0; i < SUBFRAMES; i++) {
918  int temp;
919  int auto_corr[2];
920  int scale, energy;
921 
922  /* Normalize */
923  scale = scale_vector(dst, buf, SUBFRAME_LEN);
924 
925  /* Compute auto correlation coefficients */
926  auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
927  auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN);
928 
929  /* Compute reflection coefficient */
930  temp = auto_corr[1] >> 16;
931  if (temp) {
932  temp = (auto_corr[0] >> 2) / temp;
933  }
934  p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
935  temp = -p->reflection_coef >> 1 & ~3;
936 
937  /* Compensation filter */
938  for (j = 0; j < SUBFRAME_LEN; j++) {
939  dst[j] = av_sat_dadd32(signal_ptr[j],
940  (signal_ptr[j - 1] >> 16) * temp) >> 16;
941  }
942 
943  /* Compute normalized signal energy */
944  temp = 2 * scale + 4;
945  if (temp < 0) {
946  energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
947  } else
948  energy = auto_corr[1] >> temp;
949 
950  gain_scale(p, dst, energy);
951 
952  buf += SUBFRAME_LEN;
953  signal_ptr += SUBFRAME_LEN;
954  dst += SUBFRAME_LEN;
955  }
956 }
957 
958 static int sid_gain_to_lsp_index(int gain)
959 {
960  if (gain < 0x10)
961  return gain << 6;
962  else if (gain < 0x20)
963  return gain - 8 << 7;
964  else
965  return gain - 20 << 8;
966 }
967 
968 static inline int cng_rand(int *state, int base)
969 {
970  *state = (*state * 521 + 259) & 0xFFFF;
971  return (*state & 0x7FFF) * base >> 15;
972 }
973 
974 static int estimate_sid_gain(G723_1_Context *p)
975 {
976  int i, shift, seg, seg2, t, val, val_add, x, y;
977 
978  shift = 16 - p->cur_gain * 2;
979  if (shift > 0)
980  t = p->sid_gain << shift;
981  else
982  t = p->sid_gain >> -shift;
983  x = t * cng_filt[0] >> 16;
984 
985  if (x >= cng_bseg[2])
986  return 0x3F;
987 
988  if (x >= cng_bseg[1]) {
989  shift = 4;
990  seg = 3;
991  } else {
992  shift = 3;
993  seg = (x >= cng_bseg[0]);
994  }
995  seg2 = FFMIN(seg, 3);
996 
997  val = 1 << shift;
998  val_add = val >> 1;
999  for (i = 0; i < shift; i++) {
1000  t = seg * 32 + (val << seg2);
1001  t *= t;
1002  if (x >= t)
1003  val += val_add;
1004  else
1005  val -= val_add;
1006  val_add >>= 1;
1007  }
1008 
1009  t = seg * 32 + (val << seg2);
1010  y = t * t - x;
1011  if (y <= 0) {
1012  t = seg * 32 + (val + 1 << seg2);
1013  t = t * t - x;
1014  val = (seg2 - 1 << 4) + val;
1015  if (t >= y)
1016  val++;
1017  } else {
1018  t = seg * 32 + (val - 1 << seg2);
1019  t = t * t - x;
1020  val = (seg2 - 1 << 4) + val;
1021  if (t >= y)
1022  val--;
1023  }
1024 
1025  return val;
1026 }
1027 
1028 static void generate_noise(G723_1_Context *p)
1029 {
1030  int i, j, idx, t;
1031  int off[SUBFRAMES];
1032  int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
1033  int tmp[SUBFRAME_LEN * 2];
1034  int16_t *vector_ptr;
1035  int64_t sum;
1036  int b0, c, delta, x, shift;
1037 
1038  p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
1039  p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
1040 
1041  for (i = 0; i < SUBFRAMES; i++) {
1042  p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
1043  p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i];
1044  }
1045 
1046  for (i = 0; i < SUBFRAMES / 2; i++) {
1047  t = cng_rand(&p->cng_random_seed, 1 << 13);
1048  off[i * 2] = t & 1;
1049  off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
1050  t >>= 2;
1051  for (j = 0; j < 11; j++) {
1052  signs[i * 11 + j] = (t & 1) * 2 - 1 << 14;
1053  t >>= 1;
1054  }
1055  }
1056 
1057  idx = 0;
1058  for (i = 0; i < SUBFRAMES; i++) {
1059  for (j = 0; j < SUBFRAME_LEN / 2; j++)
1060  tmp[j] = j;
1061  t = SUBFRAME_LEN / 2;
1062  for (j = 0; j < pulses[i]; j++, idx++) {
1063  int idx2 = cng_rand(&p->cng_random_seed, t);
1064 
1065  pos[idx] = tmp[idx2] * 2 + off[i];
1066  tmp[idx2] = tmp[--t];
1067  }
1068  }
1069 
1070  vector_ptr = p->audio + LPC_ORDER;
1071  memcpy(vector_ptr, p->prev_excitation,
1072  PITCH_MAX * sizeof(*p->excitation));
1073  for (i = 0; i < SUBFRAMES; i += 2) {
1074  gen_acb_excitation(vector_ptr, vector_ptr,
1075  p->pitch_lag[i >> 1], &p->subframe[i],
1076  p->cur_rate);
1077  gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
1078  vector_ptr + SUBFRAME_LEN,
1079  p->pitch_lag[i >> 1], &p->subframe[i + 1],
1080  p->cur_rate);
1081 
1082  t = 0;
1083  for (j = 0; j < SUBFRAME_LEN * 2; j++)
1084  t |= FFABS(vector_ptr[j]);
1085  t = FFMIN(t, 0x7FFF);
1086  if (!t) {
1087  shift = 0;
1088  } else {
1089  shift = -10 + av_log2(t);
1090  if (shift < -2)
1091  shift = -2;
1092  }
1093  sum = 0;
1094  if (shift < 0) {
1095  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
1096  t = vector_ptr[j] << -shift;
1097  sum += t * t;
1098  tmp[j] = t;
1099  }
1100  } else {
1101  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
1102  t = vector_ptr[j] >> shift;
1103  sum += t * t;
1104  tmp[j] = t;
1105  }
1106  }
1107 
1108  b0 = 0;
1109  for (j = 0; j < 11; j++)
1110  b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
1111  b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
1112 
1113  c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
1114  if (shift * 2 + 3 >= 0)
1115  c >>= shift * 2 + 3;
1116  else
1117  c <<= -(shift * 2 + 3);
1118  c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
1119 
1120  delta = b0 * b0 * 2 - c;
1121  if (delta <= 0) {
1122  x = -b0;
1123  } else {
1124  delta = square_root(delta);
1125  x = delta - b0;
1126  t = delta + b0;
1127  if (FFABS(t) < FFABS(x))
1128  x = -t;
1129  }
1130  shift++;
1131  if (shift < 0)
1132  x >>= -shift;
1133  else
1134  x <<= shift;
1135  x = av_clip(x, -10000, 10000);
1136 
1137  for (j = 0; j < 11; j++) {
1138  idx = (i / 2) * 11 + j;
1139  vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
1140  (x * signs[idx] >> 15));
1141  }
1142 
1143  /* copy decoded data to serve as a history for the next decoded subframes */
1144  memcpy(vector_ptr + PITCH_MAX, vector_ptr,
1145  sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
1146  vector_ptr += SUBFRAME_LEN * 2;
1147  }
1148  /* Save the excitation for the next frame */
1149  memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
1150  PITCH_MAX * sizeof(*p->excitation));
1151 }
1152 
1153 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
1154  int *got_frame_ptr, AVPacket *avpkt)
1155 {
1156  G723_1_Context *p = avctx->priv_data;
1157  AVFrame *frame = data;
1158  const uint8_t *buf = avpkt->data;
1159  int buf_size = avpkt->size;
1160  int dec_mode = buf[0] & 3;
1161 
1162  PPFParam ppf[SUBFRAMES];
1163  int16_t cur_lsp[LPC_ORDER];
1164  int16_t lpc[SUBFRAMES * LPC_ORDER];
1165  int16_t acb_vector[SUBFRAME_LEN];
1166  int16_t *out;
1167  int bad_frame = 0, i, j, ret;
1168  int16_t *audio = p->audio;
1169 
1170  if (buf_size < frame_size[dec_mode]) {
1171  if (buf_size)
1172  av_log(avctx, AV_LOG_WARNING,
1173  "Expected %d bytes, got %d - skipping packet\n",
1174  frame_size[dec_mode], buf_size);
1175  *got_frame_ptr = 0;
1176  return buf_size;
1177  }
1178 
1179  if (unpack_bitstream(p, buf, buf_size) < 0) {
1180  bad_frame = 1;
1181  if (p->past_frame_type == ACTIVE_FRAME)
1182  p->cur_frame_type = ACTIVE_FRAME;
1183  else
1184  p->cur_frame_type = UNTRANSMITTED_FRAME;
1185  }
1186 
1187  frame->nb_samples = FRAME_LEN;
1188  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1189  return ret;
1190 
1191  out = (int16_t *)frame->data[0];
1192 
1193  if (p->cur_frame_type == ACTIVE_FRAME) {
1194  if (!bad_frame)
1195  p->erased_frames = 0;
1196  else if (p->erased_frames != 3)
1197  p->erased_frames++;
1198 
1199  inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
1200  lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
1201 
1202  /* Save the lsp_vector for the next frame */
1203  memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
1204 
1205  /* Generate the excitation for the frame */
1206  memcpy(p->excitation, p->prev_excitation,
1207  PITCH_MAX * sizeof(*p->excitation));
1208  if (!p->erased_frames) {
1209  int16_t *vector_ptr = p->excitation + PITCH_MAX;
1210 
1211  /* Update interpolation gain memory */
1212  p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
1213  p->subframe[3].amp_index) >> 1];
1214  for (i = 0; i < SUBFRAMES; i++) {
1215  gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
1216  p->pitch_lag[i >> 1], i);
1217  gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
1218  p->pitch_lag[i >> 1], &p->subframe[i],
1219  p->cur_rate);
1220  /* Get the total excitation */
1221  for (j = 0; j < SUBFRAME_LEN; j++) {
1222  int v = av_clip_int16(vector_ptr[j] << 1);
1223  vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
1224  }
1225  vector_ptr += SUBFRAME_LEN;
1226  }
1227 
1228  vector_ptr = p->excitation + PITCH_MAX;
1229 
1230  p->interp_index = comp_interp_index(p, p->pitch_lag[1],
1231  &p->sid_gain, &p->cur_gain);
1232 
1233  /* Peform pitch postfiltering */
1234  if (p->postfilter) {
1235  i = PITCH_MAX;
1236  for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1237  comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
1238  ppf + j, p->cur_rate);
1239 
1240  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1241  ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
1242  vector_ptr + i,
1243  vector_ptr + i + ppf[j].index,
1244  ppf[j].sc_gain,
1245  ppf[j].opt_gain,
1246  1 << 14, 15, SUBFRAME_LEN);
1247  } else {
1248  audio = vector_ptr - LPC_ORDER;
1249  }
1250 
1251  /* Save the excitation for the next frame */
1252  memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
1253  PITCH_MAX * sizeof(*p->excitation));
1254  } else {
1255  p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
1256  if (p->erased_frames == 3) {
1257  /* Mute output */
1258  memset(p->excitation, 0,
1259  (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
1260  memset(p->prev_excitation, 0,
1261  PITCH_MAX * sizeof(*p->excitation));
1262  memset(frame->data[0], 0,
1263  (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
1264  } else {
1265  int16_t *buf = p->audio + LPC_ORDER;
1266 
1267  /* Regenerate frame */
1268  residual_interp(p->excitation, buf, p->interp_index,
1269  p->interp_gain, &p->random_seed);
1270 
1271  /* Save the excitation for the next frame */
1272  memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
1273  PITCH_MAX * sizeof(*p->excitation));
1274  }
1275  }
1276  p->cng_random_seed = CNG_RANDOM_SEED;
1277  } else {
1278  if (p->cur_frame_type == SID_FRAME) {
1279  p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
1280  inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
1281  } else if (p->past_frame_type == ACTIVE_FRAME) {
1282  p->sid_gain = estimate_sid_gain(p);
1283  }
1284 
1285  if (p->past_frame_type == ACTIVE_FRAME)
1286  p->cur_gain = p->sid_gain;
1287  else
1288  p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
1289  generate_noise(p);
1290  lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
1291  /* Save the lsp_vector for the next frame */
1292  memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
1293  }
1294 
1295  p->past_frame_type = p->cur_frame_type;
1296 
1297  memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
1298  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1299  ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
1300  audio + i, SUBFRAME_LEN, LPC_ORDER,
1301  0, 1, 1 << 12);
1302  memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
1303 
1304  if (p->postfilter) {
1305  formant_postfilter(p, lpc, p->audio, out);
1306  } else { // if output is not postfiltered it should be scaled by 2
1307  for (i = 0; i < FRAME_LEN; i++)
1308  out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
1309  }
1310 
1311  *got_frame_ptr = 1;
1312 
1313  return frame_size[dec_mode];
1314 }
1315 
1316 #define OFFSET(x) offsetof(G723_1_Context, x)
1317 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
1318 
1319 static const AVOption options[] = {
1320  { "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,
1321  { .i64 = 1 }, 0, 1, AD },
1322  { NULL }
1323 };
1324 
1325 
1326 static const AVClass g723_1dec_class = {
1327  .class_name = "G.723.1 decoder",
1328  .item_name = av_default_item_name,
1329  .option = options,
1330  .version = LIBAVUTIL_VERSION_INT,
1331 };
1332 
1333 AVCodec ff_g723_1_decoder = {
1334  .name = "g723_1",
1335  .type = AVMEDIA_TYPE_AUDIO,
1336  .id = AV_CODEC_ID_G723_1,
1337  .priv_data_size = sizeof(G723_1_Context),
1338  .init = g723_1_decode_init,
1339  .decode = g723_1_decode_frame,
1340  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1341  .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
1342  .priv_class = &g723_1dec_class,
1343 };
1344 
1345 #if CONFIG_G723_1_ENCODER
1346 #define BITSTREAM_WRITER_LE
1347 #include "put_bits.h"
1348 
1349 static av_cold int g723_1_encode_init(AVCodecContext *avctx)
1350 {
1351  G723_1_Context *p = avctx->priv_data;
1352 
1353  if (avctx->sample_rate != 8000) {
1354  av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
1355  return -1;
1356  }
1357 
1358  if (avctx->channels != 1) {
1359  av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
1360  return AVERROR(EINVAL);
1361  }
1362 
1363  if (avctx->bit_rate == 6300) {
1364  p->cur_rate = RATE_6300;
1365  } else if (avctx->bit_rate == 5300) {
1366  av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");
1367  return AVERROR_PATCHWELCOME;
1368  } else {
1369  av_log(avctx, AV_LOG_ERROR,
1370  "Bitrate not supported, use 6.3k\n");
1371  return AVERROR(EINVAL);
1372  }
1373  avctx->frame_size = 240;
1374  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
1375 
1376  return 0;
1377 }
1378 
1379 /**
1380  * Remove DC component from the input signal.
1381  *
1382  * @param buf input signal
1383  * @param fir zero memory
1384  * @param iir pole memory
1385  */
1386 static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
1387 {
1388  int i;
1389  for (i = 0; i < FRAME_LEN; i++) {
1390  *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
1391  *fir = buf[i];
1392  buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
1393  }
1394 }
1395 
1396 /**
1397  * Estimate autocorrelation of the input vector.
1398  *
1399  * @param buf input buffer
1400  * @param autocorr autocorrelation coefficients vector
1401  */
1402 static void comp_autocorr(int16_t *buf, int16_t *autocorr)
1403 {
1404  int i, scale, temp;
1405  int16_t vector[LPC_FRAME];
1406 
1407  scale_vector(vector, buf, LPC_FRAME);
1408 
1409  /* Apply the Hamming window */
1410  for (i = 0; i < LPC_FRAME; i++)
1411  vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
1412 
1413  /* Compute the first autocorrelation coefficient */
1414  temp = ff_dot_product(vector, vector, LPC_FRAME);
1415 
1416  /* Apply a white noise correlation factor of (1025/1024) */
1417  temp += temp >> 10;
1418 
1419  /* Normalize */
1420  scale = normalize_bits_int32(temp);
1421  autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +
1422  (1 << 15)) >> 16;
1423 
1424  /* Compute the remaining coefficients */
1425  if (!autocorr[0]) {
1426  memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
1427  } else {
1428  for (i = 1; i <= LPC_ORDER; i++) {
1429  temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
1430  temp = MULL2((temp << scale), binomial_window[i - 1]);
1431  autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
1432  }
1433  }
1434 }
1435 
1436 /**
1437  * Use Levinson-Durbin recursion to compute LPC coefficients from
1438  * autocorrelation values.
1439  *
1440  * @param lpc LPC coefficients vector
1441  * @param autocorr autocorrelation coefficients vector
1442  * @param error prediction error
1443  */
1444 static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
1445 {
1446  int16_t vector[LPC_ORDER];
1447  int16_t partial_corr;
1448  int i, j, temp;
1449 
1450  memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
1451 
1452  for (i = 0; i < LPC_ORDER; i++) {
1453  /* Compute the partial correlation coefficient */
1454  temp = 0;
1455  for (j = 0; j < i; j++)
1456  temp -= lpc[j] * autocorr[i - j - 1];
1457  temp = ((autocorr[i] << 13) + temp) << 3;
1458 
1459  if (FFABS(temp) >= (error << 16))
1460  break;
1461 
1462  partial_corr = temp / (error << 1);
1463 
1464  lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +
1465  (1 << 15)) >> 16;
1466 
1467  /* Update the prediction error */
1468  temp = MULL2(temp, partial_corr);
1469  error = av_clipl_int32((int64_t)(error << 16) - temp +
1470  (1 << 15)) >> 16;
1471 
1472  memcpy(vector, lpc, i * sizeof(int16_t));
1473  for (j = 0; j < i; j++) {
1474  temp = partial_corr * vector[i - j - 1] << 1;
1475  lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +
1476  (1 << 15)) >> 16;
1477  }
1478  }
1479 }
1480 
1481 /**
1482  * Calculate LPC coefficients for the current frame.
1483  *
1484  * @param buf current frame
1485  * @param prev_data 2 trailing subframes of the previous frame
1486  * @param lpc LPC coefficients vector
1487  */
1488 static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
1489 {
1490  int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
1491  int16_t *autocorr_ptr = autocorr;
1492  int16_t *lpc_ptr = lpc;
1493  int i, j;
1494 
1495  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
1496  comp_autocorr(buf + i, autocorr_ptr);
1497  levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
1498 
1499  lpc_ptr += LPC_ORDER;
1500  autocorr_ptr += LPC_ORDER + 1;
1501  }
1502 }
1503 
1504 static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
1505 {
1506  int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
1507  ///< polynomials (F1, F2) ordered as
1508  ///< f1[0], f2[0], ...., f1[5], f2[5]
1509 
1510  int max, shift, cur_val, prev_val, count, p;
1511  int i, j;
1512  int64_t temp;
1513 
1514  /* Initialize f1[0] and f2[0] to 1 in Q25 */
1515  for (i = 0; i < LPC_ORDER; i++)
1516  lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
1517 
1518  /* Apply bandwidth expansion on the LPC coefficients */
1519  f[0] = f[1] = 1 << 25;
1520 
1521  /* Compute the remaining coefficients */
1522  for (i = 0; i < LPC_ORDER / 2; i++) {
1523  /* f1 */
1524  f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
1525  /* f2 */
1526  f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
1527  }
1528 
1529  /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
1530  f[LPC_ORDER] >>= 1;
1531  f[LPC_ORDER + 1] >>= 1;
1532 
1533  /* Normalize and shorten */
1534  max = FFABS(f[0]);
1535  for (i = 1; i < LPC_ORDER + 2; i++)
1536  max = FFMAX(max, FFABS(f[i]));
1537 
1538  shift = normalize_bits_int32(max);
1539 
1540  for (i = 0; i < LPC_ORDER + 2; i++)
1541  f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;
1542 
1543  /**
1544  * Evaluate F1 and F2 at uniform intervals of pi/256 along the
1545  * unit circle and check for zero crossings.
1546  */
1547  p = 0;
1548  temp = 0;
1549  for (i = 0; i <= LPC_ORDER / 2; i++)
1550  temp += f[2 * i] * cos_tab[0];
1551  prev_val = av_clipl_int32(temp << 1);
1552  count = 0;
1553  for ( i = 1; i < COS_TBL_SIZE / 2; i++) {
1554  /* Evaluate */
1555  temp = 0;
1556  for (j = 0; j <= LPC_ORDER / 2; j++)
1557  temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
1558  cur_val = av_clipl_int32(temp << 1);
1559 
1560  /* Check for sign change, indicating a zero crossing */
1561  if ((cur_val ^ prev_val) < 0) {
1562  int abs_cur = FFABS(cur_val);
1563  int abs_prev = FFABS(prev_val);
1564  int sum = abs_cur + abs_prev;
1565 
1566  shift = normalize_bits_int32(sum);
1567  sum <<= shift;
1568  abs_prev = abs_prev << shift >> 8;
1569  lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
1570 
1571  if (count == LPC_ORDER)
1572  break;
1573 
1574  /* Switch between sum and difference polynomials */
1575  p ^= 1;
1576 
1577  /* Evaluate */
1578  temp = 0;
1579  for (j = 0; j <= LPC_ORDER / 2; j++){
1580  temp += f[LPC_ORDER - 2 * j + p] *
1581  cos_tab[i * j % COS_TBL_SIZE];
1582  }
1583  cur_val = av_clipl_int32(temp<<1);
1584  }
1585  prev_val = cur_val;
1586  }
1587 
1588  if (count != LPC_ORDER)
1589  memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
1590 }
1591 
1592 /**
1593  * Quantize the current LSP subvector.
1594  *
1595  * @param num band number
1596  * @param offset offset of the current subvector in an LPC_ORDER vector
1597  * @param size size of the current subvector
1598  */
1599 #define get_index(num, offset, size) \
1600 {\
1601  int error, max = -1;\
1602  int16_t temp[4];\
1603  int i, j;\
1604  for (i = 0; i < LSP_CB_SIZE; i++) {\
1605  for (j = 0; j < size; j++){\
1606  temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
1607  (1 << 14)) >> 15;\
1608  }\
1609  error = dot_product(lsp + (offset), temp, size) << 1;\
1610  error -= dot_product(lsp_band##num[i], temp, size);\
1611  if (error > max) {\
1612  max = error;\
1613  lsp_index[num] = i;\
1614  }\
1615  }\
1616 }
1617 
1618 /**
1619  * Vector quantize the LSP frequencies.
1620  *
1621  * @param lsp the current lsp vector
1622  * @param prev_lsp the previous lsp vector
1623  */
1624 static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
1625 {
1626  int16_t weight[LPC_ORDER];
1627  int16_t min, max;
1628  int shift, i;
1629 
1630  /* Calculate the VQ weighting vector */
1631  weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
1632  weight[LPC_ORDER - 1] = (1 << 20) /
1633  (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
1634 
1635  for (i = 1; i < LPC_ORDER - 1; i++) {
1636  min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
1637  if (min > 0x20)
1638  weight[i] = (1 << 20) / min;
1639  else
1640  weight[i] = INT16_MAX;
1641  }
1642 
1643  /* Normalize */
1644  max = 0;
1645  for (i = 0; i < LPC_ORDER; i++)
1646  max = FFMAX(weight[i], max);
1647 
1648  shift = normalize_bits_int16(max);
1649  for (i = 0; i < LPC_ORDER; i++) {
1650  weight[i] <<= shift;
1651  }
1652 
1653  /* Compute the VQ target vector */
1654  for (i = 0; i < LPC_ORDER; i++) {
1655  lsp[i] -= dc_lsp[i] +
1656  (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
1657  }
1658 
1659  get_index(0, 0, 3);
1660  get_index(1, 3, 3);
1661  get_index(2, 6, 4);
1662 }
1663 
1664 /**
1665  * Apply the formant perceptual weighting filter.
1666  *
1667  * @param flt_coef filter coefficients
1668  * @param unq_lpc unquantized lpc vector
1669  */
1670 static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
1671  int16_t *unq_lpc, int16_t *buf)
1672 {
1673  int16_t vector[FRAME_LEN + LPC_ORDER];
1674  int i, j, k, l = 0;
1675 
1676  memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
1677  memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
1678  memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
1679 
1680  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
1681  for (k = 0; k < LPC_ORDER; k++) {
1682  flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
1683  (1 << 14)) >> 15;
1684  flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
1685  percept_flt_tbl[1][k] +
1686  (1 << 14)) >> 15;
1687  }
1688  iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,
1689  buf + i, 0);
1690  l += LPC_ORDER;
1691  }
1692  memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
1693  memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
1694 }
1695 
1696 /**
1697  * Estimate the open loop pitch period.
1698  *
1699  * @param buf perceptually weighted speech
1700  * @param start estimation is carried out from this position
1701  */
1702 static int estimate_pitch(int16_t *buf, int start)
1703 {
1704  int max_exp = 32;
1705  int max_ccr = 0x4000;
1706  int max_eng = 0x7fff;
1707  int index = PITCH_MIN;
1708  int offset = start - PITCH_MIN + 1;
1709 
1710  int ccr, eng, orig_eng, ccr_eng, exp;
1711  int diff, temp;
1712 
1713  int i;
1714 
1715  orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
1716 
1717  for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
1718  offset--;
1719 
1720  /* Update energy and compute correlation */
1721  orig_eng += buf[offset] * buf[offset] -
1722  buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
1723  ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
1724  if (ccr <= 0)
1725  continue;
1726 
1727  /* Split into mantissa and exponent to maintain precision */
1728  exp = normalize_bits_int32(ccr);
1729  ccr = av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;
1730  exp <<= 1;
1731  ccr *= ccr;
1732  temp = normalize_bits_int32(ccr);
1733  ccr = ccr << temp >> 16;
1734  exp += temp;
1735 
1736  temp = normalize_bits_int32(orig_eng);
1737  eng = av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;
1738  exp -= temp;
1739 
1740  if (ccr >= eng) {
1741  exp--;
1742  ccr >>= 1;
1743  }
1744  if (exp > max_exp)
1745  continue;
1746 
1747  if (exp + 1 < max_exp)
1748  goto update;
1749 
1750  /* Equalize exponents before comparison */
1751  if (exp + 1 == max_exp)
1752  temp = max_ccr >> 1;
1753  else
1754  temp = max_ccr;
1755  ccr_eng = ccr * max_eng;
1756  diff = ccr_eng - eng * temp;
1757  if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
1758 update:
1759  index = i;
1760  max_exp = exp;
1761  max_ccr = ccr;
1762  max_eng = eng;
1763  }
1764  }
1765  return index;
1766 }
1767 
1768 /**
1769  * Compute harmonic noise filter parameters.
1770  *
1771  * @param buf perceptually weighted speech
1772  * @param pitch_lag open loop pitch period
1773  * @param hf harmonic filter parameters
1774  */
1775 static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
1776 {
1777  int ccr, eng, max_ccr, max_eng;
1778  int exp, max, diff;
1779  int energy[15];
1780  int i, j;
1781 
1782  for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
1783  /* Compute residual energy */
1784  energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
1785  /* Compute correlation */
1786  energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
1787  }
1788 
1789  /* Compute target energy */
1790  energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
1791 
1792  /* Normalize */
1793  max = 0;
1794  for (i = 0; i < 15; i++)
1795  max = FFMAX(max, FFABS(energy[i]));
1796 
1797  exp = normalize_bits_int32(max);
1798  for (i = 0; i < 15; i++) {
1799  energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
1800  (1 << 15)) >> 16;
1801  }
1802 
1803  hf->index = -1;
1804  hf->gain = 0;
1805  max_ccr = 1;
1806  max_eng = 0x7fff;
1807 
1808  for (i = 0; i <= 6; i++) {
1809  eng = energy[i << 1];
1810  ccr = energy[(i << 1) + 1];
1811 
1812  if (ccr <= 0)
1813  continue;
1814 
1815  ccr = (ccr * ccr + (1 << 14)) >> 15;
1816  diff = ccr * max_eng - eng * max_ccr;
1817  if (diff > 0) {
1818  max_ccr = ccr;
1819  max_eng = eng;
1820  hf->index = i;
1821  }
1822  }
1823 
1824  if (hf->index == -1) {
1825  hf->index = pitch_lag;
1826  return;
1827  }
1828 
1829  eng = energy[14] * max_eng;
1830  eng = (eng >> 2) + (eng >> 3);
1831  ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
1832  if (eng < ccr) {
1833  eng = energy[(hf->index << 1) + 1];
1834 
1835  if (eng >= max_eng)
1836  hf->gain = 0x2800;
1837  else
1838  hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
1839  }
1840  hf->index += pitch_lag - 3;
1841 }
1842 
1843 /**
1844  * Apply the harmonic noise shaping filter.
1845  *
1846  * @param hf filter parameters
1847  */
1848 static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
1849 {
1850  int i;
1851 
1852  for (i = 0; i < SUBFRAME_LEN; i++) {
1853  int64_t temp = hf->gain * src[i - hf->index] << 1;
1854  dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
1855  }
1856 }
1857 
1858 static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
1859 {
1860  int i;
1861  for (i = 0; i < SUBFRAME_LEN; i++) {
1862  int64_t temp = hf->gain * src[i - hf->index] << 1;
1863  dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
1864  (1 << 15)) >> 16;
1865 
1866  }
1867 }
1868 
1869 /**
1870  * Combined synthesis and formant perceptual weighting filer.
1871  *
1872  * @param qnt_lpc quantized lpc coefficients
1873  * @param perf_lpc perceptual filter coefficients
1874  * @param perf_fir perceptual filter fir memory
1875  * @param perf_iir perceptual filter iir memory
1876  * @param scale the filter output will be scaled by 2^scale
1877  */
1878 static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
1879  int16_t *perf_fir, int16_t *perf_iir,
1880  const int16_t *src, int16_t *dest, int scale)
1881 {
1882  int i, j;
1883  int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
1884  int64_t buf[SUBFRAME_LEN];
1885 
1886  int16_t *bptr_16 = buf_16 + LPC_ORDER;
1887 
1888  memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
1889  memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
1890 
1891  for (i = 0; i < SUBFRAME_LEN; i++) {
1892  int64_t temp = 0;
1893  for (j = 1; j <= LPC_ORDER; j++)
1894  temp -= qnt_lpc[j - 1] * bptr_16[i - j];
1895 
1896  buf[i] = (src[i] << 15) + (temp << 3);
1897  bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
1898  }
1899 
1900  for (i = 0; i < SUBFRAME_LEN; i++) {
1901  int64_t fir = 0, iir = 0;
1902  for (j = 1; j <= LPC_ORDER; j++) {
1903  fir -= perf_lpc[j - 1] * bptr_16[i - j];
1904  iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
1905  }
1906  dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
1907  (1 << 15)) >> 16;
1908  }
1909  memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
1910  memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
1911  sizeof(int16_t) * LPC_ORDER);
1912 }
1913 
1914 /**
1915  * Compute the adaptive codebook contribution.
1916  *
1917  * @param buf input signal
1918  * @param index the current subframe index
1919  */
1920 static void acb_search(G723_1_Context *p, int16_t *residual,
1921  int16_t *impulse_resp, const int16_t *buf,
1922  int index)
1923 {
1924 
1925  int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
1926 
1927  const int16_t *cb_tbl = adaptive_cb_gain85;
1928 
1929  int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
1930 
1931  int pitch_lag = p->pitch_lag[index >> 1];
1932  int acb_lag = 1;
1933  int acb_gain = 0;
1934  int odd_frame = index & 1;
1935  int iter = 3 + odd_frame;
1936  int count = 0;
1937  int tbl_size = 85;
1938 
1939  int i, j, k, l, max;
1940  int64_t temp;
1941 
1942  if (!odd_frame) {
1943  if (pitch_lag == PITCH_MIN)
1944  pitch_lag++;
1945  else
1946  pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
1947  }
1948 
1949  for (i = 0; i < iter; i++) {
1950  get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
1951 
1952  for (j = 0; j < SUBFRAME_LEN; j++) {
1953  temp = 0;
1954  for (k = 0; k <= j; k++)
1955  temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
1956  flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
1957  (1 << 15)) >> 16;
1958  }
1959 
1960  for (j = PITCH_ORDER - 2; j >= 0; j--) {
1961  flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
1962  for (k = 1; k < SUBFRAME_LEN; k++) {
1963  temp = (flt_buf[j + 1][k - 1] << 15) +
1964  residual[j] * impulse_resp[k];
1965  flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
1966  }
1967  }
1968 
1969  /* Compute crosscorrelation with the signal */
1970  for (j = 0; j < PITCH_ORDER; j++) {
1971  temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
1972  ccr_buf[count++] = av_clipl_int32(temp << 1);
1973  }
1974 
1975  /* Compute energies */
1976  for (j = 0; j < PITCH_ORDER; j++) {
1977  ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
1978  SUBFRAME_LEN);
1979  }
1980 
1981  for (j = 1; j < PITCH_ORDER; j++) {
1982  for (k = 0; k < j; k++) {
1983  temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
1984  ccr_buf[count++] = av_clipl_int32(temp<<2);
1985  }
1986  }
1987  }
1988 
1989  /* Normalize and shorten */
1990  max = 0;
1991  for (i = 0; i < 20 * iter; i++)
1992  max = FFMAX(max, FFABS(ccr_buf[i]));
1993 
1994  temp = normalize_bits_int32(max);
1995 
1996  for (i = 0; i < 20 * iter; i++){
1997  ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +
1998  (1 << 15)) >> 16;
1999  }
2000 
2001  max = 0;
2002  for (i = 0; i < iter; i++) {
2003  /* Select quantization table */
2004  if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
2005  odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
2006  cb_tbl = adaptive_cb_gain170;
2007  tbl_size = 170;
2008  }
2009 
2010  for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
2011  temp = 0;
2012  for (l = 0; l < 20; l++)
2013  temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
2014  temp = av_clipl_int32(temp);
2015 
2016  if (temp > max) {
2017  max = temp;
2018  acb_gain = j;
2019  acb_lag = i;
2020  }
2021  }
2022  }
2023 
2024  if (!odd_frame) {
2025  pitch_lag += acb_lag - 1;
2026  acb_lag = 1;
2027  }
2028 
2029  p->pitch_lag[index >> 1] = pitch_lag;
2030  p->subframe[index].ad_cb_lag = acb_lag;
2031  p->subframe[index].ad_cb_gain = acb_gain;
2032 }
2033 
2034 /**
2035  * Subtract the adaptive codebook contribution from the input
2036  * to obtain the residual.
2037  *
2038  * @param buf target vector
2039  */
2040 static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
2041  int16_t *buf)
2042 {
2043  int i, j;
2044  /* Subtract adaptive CB contribution to obtain the residual */
2045  for (i = 0; i < SUBFRAME_LEN; i++) {
2046  int64_t temp = buf[i] << 14;
2047  for (j = 0; j <= i; j++)
2048  temp -= residual[j] * impulse_resp[i - j];
2049 
2050  buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
2051  }
2052 }
2053 
2054 /**
2055  * Quantize the residual signal using the fixed codebook (MP-MLQ).
2056  *
2057  * @param optim optimized fixed codebook parameters
2058  * @param buf excitation vector
2059  */
2060 static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
2061  int16_t *buf, int pulse_cnt, int pitch_lag)
2062 {
2063  FCBParam param;
2064  int16_t impulse_r[SUBFRAME_LEN];
2065  int16_t temp_corr[SUBFRAME_LEN];
2066  int16_t impulse_corr[SUBFRAME_LEN];
2067 
2068  int ccr1[SUBFRAME_LEN];
2069  int ccr2[SUBFRAME_LEN];
2070  int amp, err, max, max_amp_index, min, scale, i, j, k, l;
2071 
2072  int64_t temp;
2073 
2074  /* Update impulse response */
2075  memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
2076  param.dirac_train = 0;
2077  if (pitch_lag < SUBFRAME_LEN - 2) {
2078  param.dirac_train = 1;
2079  gen_dirac_train(impulse_r, pitch_lag);
2080  }
2081 
2082  for (i = 0; i < SUBFRAME_LEN; i++)
2083  temp_corr[i] = impulse_r[i] >> 1;
2084 
2085  /* Compute impulse response autocorrelation */
2086  temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
2087 
2088  scale = normalize_bits_int32(temp);
2089  impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
2090 
2091  for (i = 1; i < SUBFRAME_LEN; i++) {
2092  temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
2093  impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
2094  }
2095 
2096  /* Compute crosscorrelation of impulse response with residual signal */
2097  scale -= 4;
2098  for (i = 0; i < SUBFRAME_LEN; i++){
2099  temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
2100  if (scale < 0)
2101  ccr1[i] = temp >> -scale;
2102  else
2103  ccr1[i] = av_clipl_int32(temp << scale);
2104  }
2105 
2106  /* Search loop */
2107  for (i = 0; i < GRID_SIZE; i++) {
2108  /* Maximize the crosscorrelation */
2109  max = 0;
2110  for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
2111  temp = FFABS(ccr1[j]);
2112  if (temp >= max) {
2113  max = temp;
2114  param.pulse_pos[0] = j;
2115  }
2116  }
2117 
2118  /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
2119  amp = max;
2120  min = 1 << 30;
2121  max_amp_index = GAIN_LEVELS - 2;
2122  for (j = max_amp_index; j >= 2; j--) {
2123  temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *
2124  impulse_corr[0] << 1);
2125  temp = FFABS(temp - amp);
2126  if (temp < min) {
2127  min = temp;
2128  max_amp_index = j;
2129  }
2130  }
2131 
2132  max_amp_index--;
2133  /* Select additional gain values */
2134  for (j = 1; j < 5; j++) {
2135  for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
2136  temp_corr[k] = 0;
2137  ccr2[k] = ccr1[k];
2138  }
2139  param.amp_index = max_amp_index + j - 2;
2140  amp = fixed_cb_gain[param.amp_index];
2141 
2142  param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
2143  temp_corr[param.pulse_pos[0]] = 1;
2144 
2145  for (k = 1; k < pulse_cnt; k++) {
2146  max = -1 << 30;
2147  for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
2148  if (temp_corr[l])
2149  continue;
2150  temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
2151  temp = av_clipl_int32((int64_t)temp *
2152  param.pulse_sign[k - 1] << 1);
2153  ccr2[l] -= temp;
2154  temp = FFABS(ccr2[l]);
2155  if (temp > max) {
2156  max = temp;
2157  param.pulse_pos[k] = l;
2158  }
2159  }
2160 
2161  param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
2162  -amp : amp;
2163  temp_corr[param.pulse_pos[k]] = 1;
2164  }
2165 
2166  /* Create the error vector */
2167  memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
2168 
2169  for (k = 0; k < pulse_cnt; k++)
2170  temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
2171 
2172  for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
2173  temp = 0;
2174  for (l = 0; l <= k; l++) {
2175  int prod = av_clipl_int32((int64_t)temp_corr[l] *
2176  impulse_r[k - l] << 1);
2177  temp = av_clipl_int32(temp + prod);
2178  }
2179  temp_corr[k] = temp << 2 >> 16;
2180  }
2181 
2182  /* Compute square of error */
2183  err = 0;
2184  for (k = 0; k < SUBFRAME_LEN; k++) {
2185  int64_t prod;
2186  prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);
2187  err = av_clipl_int32(err - prod);
2188  prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);
2189  err = av_clipl_int32(err + prod);
2190  }
2191 
2192  /* Minimize */
2193  if (err < optim->min_err) {
2194  optim->min_err = err;
2195  optim->grid_index = i;
2196  optim->amp_index = param.amp_index;
2197  optim->dirac_train = param.dirac_train;
2198 
2199  for (k = 0; k < pulse_cnt; k++) {
2200  optim->pulse_sign[k] = param.pulse_sign[k];
2201  optim->pulse_pos[k] = param.pulse_pos[k];
2202  }
2203  }
2204  }
2205  }
2206 }
2207 
2208 /**
2209  * Encode the pulse position and gain of the current subframe.
2210  *
2211  * @param optim optimized fixed CB parameters
2212  * @param buf excitation vector
2213  */
2214 static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
2215  int16_t *buf, int pulse_cnt)
2216 {
2217  int i, j;
2218 
2219  j = PULSE_MAX - pulse_cnt;
2220 
2221  subfrm->pulse_sign = 0;
2222  subfrm->pulse_pos = 0;
2223 
2224  for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
2225  int val = buf[optim->grid_index + (i << 1)];
2226  if (!val) {
2227  subfrm->pulse_pos += combinatorial_table[j][i];
2228  } else {
2229  subfrm->pulse_sign <<= 1;
2230  if (val < 0) subfrm->pulse_sign++;
2231  j++;
2232 
2233  if (j == PULSE_MAX) break;
2234  }
2235  }
2236  subfrm->amp_index = optim->amp_index;
2237  subfrm->grid_index = optim->grid_index;
2238  subfrm->dirac_train = optim->dirac_train;
2239 }
2240 
2241 /**
2242  * Compute the fixed codebook excitation.
2243  *
2244  * @param buf target vector
2245  * @param impulse_resp impulse response of the combined filter
2246  */
2247 static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
2248  int16_t *buf, int index)
2249 {
2250  FCBParam optim;
2251  int pulse_cnt = pulses[index];
2252  int i;
2253 
2254  optim.min_err = 1 << 30;
2255  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
2256 
2257  if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
2258  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
2259  p->pitch_lag[index >> 1]);
2260  }
2261 
2262  /* Reconstruct the excitation */
2263  memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
2264  for (i = 0; i < pulse_cnt; i++)
2265  buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
2266 
2267  pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
2268 
2269  if (optim.dirac_train)
2270  gen_dirac_train(buf, p->pitch_lag[index >> 1]);
2271 }
2272 
2273 /**
2274  * Pack the frame parameters into output bitstream.
2275  *
2276  * @param frame output buffer
2277  * @param size size of the buffer
2278  */
2279 static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)
2280 {
2281  PutBitContext pb;
2282  int info_bits, i, temp;
2283 
2284  init_put_bits(&pb, frame, size);
2285 
2286  if (p->cur_rate == RATE_6300) {
2287  info_bits = 0;
2288  put_bits(&pb, 2, info_bits);
2289  }
2290 
2291  put_bits(&pb, 8, p->lsp_index[2]);
2292  put_bits(&pb, 8, p->lsp_index[1]);
2293  put_bits(&pb, 8, p->lsp_index[0]);
2294 
2295  put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
2296  put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
2297  put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
2298  put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
2299 
2300  /* Write 12 bit combined gain */
2301  for (i = 0; i < SUBFRAMES; i++) {
2302  temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
2303  p->subframe[i].amp_index;
2304  if (p->cur_rate == RATE_6300)
2305  temp += p->subframe[i].dirac_train << 11;
2306  put_bits(&pb, 12, temp);
2307  }
2308 
2309  put_bits(&pb, 1, p->subframe[0].grid_index);
2310  put_bits(&pb, 1, p->subframe[1].grid_index);
2311  put_bits(&pb, 1, p->subframe[2].grid_index);
2312  put_bits(&pb, 1, p->subframe[3].grid_index);
2313 
2314  if (p->cur_rate == RATE_6300) {
2315  skip_put_bits(&pb, 1); /* reserved bit */
2316 
2317  /* Write 13 bit combined position index */
2318  temp = (p->subframe[0].pulse_pos >> 16) * 810 +
2319  (p->subframe[1].pulse_pos >> 14) * 90 +
2320  (p->subframe[2].pulse_pos >> 16) * 9 +
2321  (p->subframe[3].pulse_pos >> 14);
2322  put_bits(&pb, 13, temp);
2323 
2324  put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
2325  put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
2326  put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
2327  put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
2328 
2329  put_bits(&pb, 6, p->subframe[0].pulse_sign);
2330  put_bits(&pb, 5, p->subframe[1].pulse_sign);
2331  put_bits(&pb, 6, p->subframe[2].pulse_sign);
2332  put_bits(&pb, 5, p->subframe[3].pulse_sign);
2333  }
2334 
2335  flush_put_bits(&pb);
2336  return frame_size[info_bits];
2337 }
2338 
2339 static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
2340  const AVFrame *frame, int *got_packet_ptr)
2341 {
2342  G723_1_Context *p = avctx->priv_data;
2343  int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
2344  int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
2345  int16_t cur_lsp[LPC_ORDER];
2346  int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
2347  int16_t vector[FRAME_LEN + PITCH_MAX];
2348  int offset, ret;
2349  int16_t *in = (const int16_t *)frame->data[0];
2350 
2351  HFParam hf[4];
2352  int i, j;
2353 
2354  highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
2355 
2356  memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
2357  memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
2358 
2359  comp_lpc_coeff(vector, unq_lpc);
2360  lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
2361  lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
2362 
2363  /* Update memory */
2364  memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
2365  sizeof(int16_t) * SUBFRAME_LEN);
2366  memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
2367  sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
2368  memcpy(p->prev_data, in + HALF_FRAME_LEN,
2369  sizeof(int16_t) * HALF_FRAME_LEN);
2370  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
2371 
2372  perceptual_filter(p, weighted_lpc, unq_lpc, vector);
2373 
2374  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
2375  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
2376  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
2377 
2378  scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
2379 
2380  p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
2381  p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
2382 
2383  for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
2384  comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
2385 
2386  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
2387  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
2388  memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
2389 
2390  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
2391  harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
2392 
2393  inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
2394  lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
2395 
2396  memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
2397 
2398  offset = 0;
2399  for (i = 0; i < SUBFRAMES; i++) {
2400  int16_t impulse_resp[SUBFRAME_LEN];
2401  int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
2402  int16_t flt_in[SUBFRAME_LEN];
2403  int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
2404 
2405  /**
2406  * Compute the combined impulse response of the synthesis filter,
2407  * formant perceptual weighting filter and harmonic noise shaping filter
2408  */
2409  memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
2410  memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
2411  memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
2412 
2413  flt_in[0] = 1 << 13; /* Unit impulse */
2414  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
2415  zero, zero, flt_in, vector + PITCH_MAX, 1);
2416  harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
2417 
2418  /* Compute the combined zero input response */
2419  flt_in[0] = 0;
2420  memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
2421  memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
2422 
2423  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
2424  fir, iir, flt_in, vector + PITCH_MAX, 0);
2425  memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
2426  harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
2427 
2428  acb_search(p, residual, impulse_resp, in, i);
2429  gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
2430  &p->subframe[i], p->cur_rate);
2431  sub_acb_contrib(residual, impulse_resp, in);
2432 
2433  fcb_search(p, impulse_resp, in, i);
2434 
2435  /* Reconstruct the excitation */
2436  gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
2437  &p->subframe[i], RATE_6300);
2438 
2439  memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
2440  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
2441  for (j = 0; j < SUBFRAME_LEN; j++)
2442  in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
2443  memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
2444  sizeof(int16_t) * SUBFRAME_LEN);
2445 
2446  /* Update filter memories */
2447  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
2448  p->perf_fir_mem, p->perf_iir_mem,
2449  in, vector + PITCH_MAX, 0);
2450  memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
2451  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
2452  memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
2453  sizeof(int16_t) * SUBFRAME_LEN);
2454 
2455  in += SUBFRAME_LEN;
2456  offset += LPC_ORDER;
2457  }
2458 
2459  if ((ret = ff_alloc_packet2(avctx, avpkt, 24)) < 0)
2460  return ret;
2461 
2462  *got_packet_ptr = 1;
2463  avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);
2464  return 0;
2465 }
2466 
2467 AVCodec ff_g723_1_encoder = {
2468  .name = "g723_1",
2469  .type = AVMEDIA_TYPE_AUDIO,
2470  .id = AV_CODEC_ID_G723_1,
2471  .priv_data_size = sizeof(G723_1_Context),
2472  .init = g723_1_encode_init,
2473  .encode2 = g723_1_encode_frame,
2474  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
2475  .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
2476  AV_SAMPLE_FMT_NONE},
2477 };
2478 #endif
Generated on Wed Jul 10 2013 23:47:58 for FFmpeg by   doxygen 1.8.2