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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  AVFrame frame;
46 
47  G723_1_Subframe subframe[4];
48  enum FrameType cur_frame_type;
49  enum FrameType past_frame_type;
50  enum Rate cur_rate;
51  uint8_t lsp_index[LSP_BANDS];
52  int pitch_lag[2];
53  int erased_frames;
54 
55  int16_t prev_lsp[LPC_ORDER];
56  int16_t sid_lsp[LPC_ORDER];
57  int16_t prev_excitation[PITCH_MAX];
58  int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
59  int16_t synth_mem[LPC_ORDER];
60  int16_t fir_mem[LPC_ORDER];
61  int iir_mem[LPC_ORDER];
62 
63  int random_seed;
64  int cng_random_seed;
65  int interp_index;
66  int interp_gain;
67  int sid_gain;
68  int cur_gain;
69  int reflection_coef;
70  int pf_gain; ///< formant postfilter
71  ///< gain scaling unit memory
72  int postfilter;
73 
74  int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
75  int16_t prev_data[HALF_FRAME_LEN];
76  int16_t prev_weight_sig[PITCH_MAX];
77 
78 
79  int16_t hpf_fir_mem; ///< highpass filter fir
80  int hpf_iir_mem; ///< and iir memories
81  int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir
82  int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories
83 
84  int16_t harmonic_mem[PITCH_MAX];
85 } G723_1_Context;
86 
87 static av_cold int g723_1_decode_init(AVCodecContext *avctx)
88 {
89  G723_1_Context *p = avctx->priv_data;
90 
91  avctx->channel_layout = AV_CH_LAYOUT_MONO;
92  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
93  avctx->channels = 1;
94  p->pf_gain = 1 << 12;
95 
96  avcodec_get_frame_defaults(&p->frame);
97  avctx->coded_frame = &p->frame;
98 
99  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
100  memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
101 
102  p->cng_random_seed = CNG_RANDOM_SEED;
103  p->past_frame_type = SID_FRAME;
104 
105  return 0;
106 }
107 
108 /**
109  * Unpack the frame into parameters.
110  *
111  * @param p the context
112  * @param buf pointer to the input buffer
113  * @param buf_size size of the input buffer
114  */
115 static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
116  int buf_size)
117 {
118  GetBitContext gb;
119  int ad_cb_len;
120  int temp, info_bits, i;
121 
122  init_get_bits(&gb, buf, buf_size * 8);
123 
124  /* Extract frame type and rate info */
125  info_bits = get_bits(&gb, 2);
126 
127  if (info_bits == 3) {
128  p->cur_frame_type = UNTRANSMITTED_FRAME;
129  return 0;
130  }
131 
132  /* Extract 24 bit lsp indices, 8 bit for each band */
133  p->lsp_index[2] = get_bits(&gb, 8);
134  p->lsp_index[1] = get_bits(&gb, 8);
135  p->lsp_index[0] = get_bits(&gb, 8);
136 
137  if (info_bits == 2) {
138  p->cur_frame_type = SID_FRAME;
139  p->subframe[0].amp_index = get_bits(&gb, 6);
140  return 0;
141  }
142 
143  /* Extract the info common to both rates */
144  p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
145  p->cur_frame_type = ACTIVE_FRAME;
146 
147  p->pitch_lag[0] = get_bits(&gb, 7);
148  if (p->pitch_lag[0] > 123) /* test if forbidden code */
149  return -1;
150  p->pitch_lag[0] += PITCH_MIN;
151  p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
152 
153  p->pitch_lag[1] = get_bits(&gb, 7);
154  if (p->pitch_lag[1] > 123)
155  return -1;
156  p->pitch_lag[1] += PITCH_MIN;
157  p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
158  p->subframe[0].ad_cb_lag = 1;
159  p->subframe[2].ad_cb_lag = 1;
160 
161  for (i = 0; i < SUBFRAMES; i++) {
162  /* Extract combined gain */
163  temp = get_bits(&gb, 12);
164  ad_cb_len = 170;
165  p->subframe[i].dirac_train = 0;
166  if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
167  p->subframe[i].dirac_train = temp >> 11;
168  temp &= 0x7FF;
169  ad_cb_len = 85;
170  }
171  p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
172  if (p->subframe[i].ad_cb_gain < ad_cb_len) {
173  p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
174  GAIN_LEVELS;
175  } else {
176  return -1;
177  }
178  }
179 
180  p->subframe[0].grid_index = get_bits1(&gb);
181  p->subframe[1].grid_index = get_bits1(&gb);
182  p->subframe[2].grid_index = get_bits1(&gb);
183  p->subframe[3].grid_index = get_bits1(&gb);
184 
185  if (p->cur_rate == RATE_6300) {
186  skip_bits1(&gb); /* skip reserved bit */
187 
188  /* Compute pulse_pos index using the 13-bit combined position index */
189  temp = get_bits(&gb, 13);
190  p->subframe[0].pulse_pos = temp / 810;
191 
192  temp -= p->subframe[0].pulse_pos * 810;
193  p->subframe[1].pulse_pos = FASTDIV(temp, 90);
194 
195  temp -= p->subframe[1].pulse_pos * 90;
196  p->subframe[2].pulse_pos = FASTDIV(temp, 9);
197  p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
198 
199  p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
200  get_bits(&gb, 16);
201  p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
202  get_bits(&gb, 14);
203  p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
204  get_bits(&gb, 16);
205  p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
206  get_bits(&gb, 14);
207 
208  p->subframe[0].pulse_sign = get_bits(&gb, 6);
209  p->subframe[1].pulse_sign = get_bits(&gb, 5);
210  p->subframe[2].pulse_sign = get_bits(&gb, 6);
211  p->subframe[3].pulse_sign = get_bits(&gb, 5);
212  } else { /* 5300 bps */
213  p->subframe[0].pulse_pos = get_bits(&gb, 12);
214  p->subframe[1].pulse_pos = get_bits(&gb, 12);
215  p->subframe[2].pulse_pos = get_bits(&gb, 12);
216  p->subframe[3].pulse_pos = get_bits(&gb, 12);
217 
218  p->subframe[0].pulse_sign = get_bits(&gb, 4);
219  p->subframe[1].pulse_sign = get_bits(&gb, 4);
220  p->subframe[2].pulse_sign = get_bits(&gb, 4);
221  p->subframe[3].pulse_sign = get_bits(&gb, 4);
222  }
223 
224  return 0;
225 }
226 
227 /**
228  * Bitexact implementation of sqrt(val/2).
229  */
230 static int16_t square_root(unsigned val)
231 {
232  av_assert2(!(val & 0x80000000));
233 
234  return (ff_sqrt(val << 1) >> 1) & (~1);
235 }
236 
237 /**
238  * Calculate the number of left-shifts required for normalizing the input.
239  *
240  * @param num input number
241  * @param width width of the input, 15 or 31 bits
242  */
243 static int normalize_bits(int num, int width)
244 {
245  return width - av_log2(num) - 1;
246 }
247 
248 #define normalize_bits_int16(num) normalize_bits(num, 15)
249 #define normalize_bits_int32(num) normalize_bits(num, 31)
250 
251 /**
252  * Scale vector contents based on the largest of their absolutes.
253  */
254 static int scale_vector(int16_t *dst, const int16_t *vector, int length)
255 {
256  int bits, max = 0;
257  int i;
258 
259  for (i = 0; i < length; i++)
260  max |= FFABS(vector[i]);
261 
262  bits= 14 - av_log2_16bit(max);
263  bits= FFMAX(bits, 0);
264 
265  for (i = 0; i < length; i++)
266  dst[i] = vector[i] << bits >> 3;
267 
268  return bits - 3;
269 }
270 
271 /**
272  * Perform inverse quantization of LSP frequencies.
273  *
274  * @param cur_lsp the current LSP vector
275  * @param prev_lsp the previous LSP vector
276  * @param lsp_index VQ indices
277  * @param bad_frame bad frame flag
278  */
279 static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
280  uint8_t *lsp_index, int bad_frame)
281 {
282  int min_dist, pred;
283  int i, j, temp, stable;
284 
285  /* Check for frame erasure */
286  if (!bad_frame) {
287  min_dist = 0x100;
288  pred = 12288;
289  } else {
290  min_dist = 0x200;
291  pred = 23552;
292  lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
293  }
294 
295  /* Get the VQ table entry corresponding to the transmitted index */
296  cur_lsp[0] = lsp_band0[lsp_index[0]][0];
297  cur_lsp[1] = lsp_band0[lsp_index[0]][1];
298  cur_lsp[2] = lsp_band0[lsp_index[0]][2];
299  cur_lsp[3] = lsp_band1[lsp_index[1]][0];
300  cur_lsp[4] = lsp_band1[lsp_index[1]][1];
301  cur_lsp[5] = lsp_band1[lsp_index[1]][2];
302  cur_lsp[6] = lsp_band2[lsp_index[2]][0];
303  cur_lsp[7] = lsp_band2[lsp_index[2]][1];
304  cur_lsp[8] = lsp_band2[lsp_index[2]][2];
305  cur_lsp[9] = lsp_band2[lsp_index[2]][3];
306 
307  /* Add predicted vector & DC component to the previously quantized vector */
308  for (i = 0; i < LPC_ORDER; i++) {
309  temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
310  cur_lsp[i] += dc_lsp[i] + temp;
311  }
312 
313  for (i = 0; i < LPC_ORDER; i++) {
314  cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);
315  cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);
316 
317  /* Stability check */
318  for (j = 1; j < LPC_ORDER; j++) {
319  temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
320  if (temp > 0) {
321  temp >>= 1;
322  cur_lsp[j - 1] -= temp;
323  cur_lsp[j] += temp;
324  }
325  }
326  stable = 1;
327  for (j = 1; j < LPC_ORDER; j++) {
328  temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
329  if (temp > 0) {
330  stable = 0;
331  break;
332  }
333  }
334  if (stable)
335  break;
336  }
337  if (!stable)
338  memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));
339 }
340 
341 /**
342  * Bitexact implementation of 2ab scaled by 1/2^16.
343  *
344  * @param a 32 bit multiplicand
345  * @param b 16 bit multiplier
346  */
347 #define MULL2(a, b) \
348  MULL(a,b,15)
349 
350 /**
351  * Convert LSP frequencies to LPC coefficients.
352  *
353  * @param lpc buffer for LPC coefficients
354  */
355 static void lsp2lpc(int16_t *lpc)
356 {
357  int f1[LPC_ORDER / 2 + 1];
358  int f2[LPC_ORDER / 2 + 1];
359  int i, j;
360 
361  /* Calculate negative cosine */
362  for (j = 0; j < LPC_ORDER; j++) {
363  int index = (lpc[j] >> 7) & 0x1FF;
364  int offset = lpc[j] & 0x7f;
365  int temp1 = cos_tab[index] << 16;
366  int temp2 = (cos_tab[index + 1] - cos_tab[index]) *
367  ((offset << 8) + 0x80) << 1;
368 
369  lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
370  }
371 
372  /*
373  * Compute sum and difference polynomial coefficients
374  * (bitexact alternative to lsp2poly() in lsp.c)
375  */
376  /* Initialize with values in Q28 */
377  f1[0] = 1 << 28;
378  f1[1] = (lpc[0] << 14) + (lpc[2] << 14);
379  f1[2] = lpc[0] * lpc[2] + (2 << 28);
380 
381  f2[0] = 1 << 28;
382  f2[1] = (lpc[1] << 14) + (lpc[3] << 14);
383  f2[2] = lpc[1] * lpc[3] + (2 << 28);
384 
385  /*
386  * Calculate and scale the coefficients by 1/2 in
387  * each iteration for a final scaling factor of Q25
388  */
389  for (i = 2; i < LPC_ORDER / 2; i++) {
390  f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);
391  f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);
392 
393  for (j = i; j >= 2; j--) {
394  f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +
395  (f1[j] >> 1) + (f1[j - 2] >> 1);
396  f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +
397  (f2[j] >> 1) + (f2[j - 2] >> 1);
398  }
399 
400  f1[0] >>= 1;
401  f2[0] >>= 1;
402  f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1;
403  f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
404  }
405 
406  /* Convert polynomial coefficients to LPC coefficients */
407  for (i = 0; i < LPC_ORDER / 2; i++) {
408  int64_t ff1 = f1[i + 1] + f1[i];
409  int64_t ff2 = f2[i + 1] - f2[i];
410 
411  lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;
412  lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +
413  (1 << 15)) >> 16;
414  }
415 }
416 
417 /**
418  * Quantize LSP frequencies by interpolation and convert them to
419  * the corresponding LPC coefficients.
420  *
421  * @param lpc buffer for LPC coefficients
422  * @param cur_lsp the current LSP vector
423  * @param prev_lsp the previous LSP vector
424  */
425 static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
426 {
427  int i;
428  int16_t *lpc_ptr = lpc;
429 
430  /* cur_lsp * 0.25 + prev_lsp * 0.75 */
431  ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,
432  4096, 12288, 1 << 13, 14, LPC_ORDER);
433  ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,
434  8192, 8192, 1 << 13, 14, LPC_ORDER);
435  ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,
436  12288, 4096, 1 << 13, 14, LPC_ORDER);
437  memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc));
438 
439  for (i = 0; i < SUBFRAMES; i++) {
440  lsp2lpc(lpc_ptr);
441  lpc_ptr += LPC_ORDER;
442  }
443 }
444 
445 /**
446  * Generate a train of dirac functions with period as pitch lag.
447  */
448 static void gen_dirac_train(int16_t *buf, int pitch_lag)
449 {
450  int16_t vector[SUBFRAME_LEN];
451  int i, j;
452 
453  memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
454  for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
455  for (j = 0; j < SUBFRAME_LEN - i; j++)
456  buf[i + j] += vector[j];
457  }
458 }
459 
460 /**
461  * Generate fixed codebook excitation vector.
462  *
463  * @param vector decoded excitation vector
464  * @param subfrm current subframe
465  * @param cur_rate current bitrate
466  * @param pitch_lag closed loop pitch lag
467  * @param index current subframe index
468  */
469 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
470  enum Rate cur_rate, int pitch_lag, int index)
471 {
472  int temp, i, j;
473 
474  memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
475 
476  if (cur_rate == RATE_6300) {
477  if (subfrm->pulse_pos >= max_pos[index])
478  return;
479 
480  /* Decode amplitudes and positions */
481  j = PULSE_MAX - pulses[index];
482  temp = subfrm->pulse_pos;
483  for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
484  temp -= combinatorial_table[j][i];
485  if (temp >= 0)
486  continue;
487  temp += combinatorial_table[j++][i];
488  if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
489  vector[subfrm->grid_index + GRID_SIZE * i] =
490  -fixed_cb_gain[subfrm->amp_index];
491  } else {
492  vector[subfrm->grid_index + GRID_SIZE * i] =
493  fixed_cb_gain[subfrm->amp_index];
494  }
495  if (j == PULSE_MAX)
496  break;
497  }
498  if (subfrm->dirac_train == 1)
499  gen_dirac_train(vector, pitch_lag);
500  } else { /* 5300 bps */
501  int cb_gain = fixed_cb_gain[subfrm->amp_index];
502  int cb_shift = subfrm->grid_index;
503  int cb_sign = subfrm->pulse_sign;
504  int cb_pos = subfrm->pulse_pos;
505  int offset, beta, lag;
506 
507  for (i = 0; i < 8; i += 2) {
508  offset = ((cb_pos & 7) << 3) + cb_shift + i;
509  vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
510  cb_pos >>= 3;
511  cb_sign >>= 1;
512  }
513 
514  /* Enhance harmonic components */
515  lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
516  subfrm->ad_cb_lag - 1;
517  beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
518 
519  if (lag < SUBFRAME_LEN - 2) {
520  for (i = lag; i < SUBFRAME_LEN; i++)
521  vector[i] += beta * vector[i - lag] >> 15;
522  }
523  }
524 }
525 
526 /**
527  * Get delayed contribution from the previous excitation vector.
528  */
529 static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
530 {
531  int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
532  int i;
533 
534  residual[0] = prev_excitation[offset];
535  residual[1] = prev_excitation[offset + 1];
536 
537  offset += 2;
538  for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
539  residual[i] = prev_excitation[offset + (i - 2) % lag];
540 }
541 
542 static int dot_product(const int16_t *a, const int16_t *b, int length)
543 {
544  int sum = ff_dot_product(a,b,length);
545  return av_sat_add32(sum, sum);
546 }
547 
548 /**
549  * Generate adaptive codebook excitation.
550  */
551 static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
552  int pitch_lag, G723_1_Subframe *subfrm,
553  enum Rate cur_rate)
554 {
555  int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
556  const int16_t *cb_ptr;
557  int lag = pitch_lag + subfrm->ad_cb_lag - 1;
558 
559  int i;
560  int sum;
561 
562  get_residual(residual, prev_excitation, lag);
563 
564  /* Select quantization table */
565  if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {
566  cb_ptr = adaptive_cb_gain85;
567  } else
568  cb_ptr = adaptive_cb_gain170;
569 
570  /* Calculate adaptive vector */
571  cb_ptr += subfrm->ad_cb_gain * 20;
572  for (i = 0; i < SUBFRAME_LEN; i++) {
573  sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
574  vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
575  }
576 }
577 
578 /**
579  * Estimate maximum auto-correlation around pitch lag.
580  *
581  * @param buf buffer with offset applied
582  * @param offset offset of the excitation vector
583  * @param ccr_max pointer to the maximum auto-correlation
584  * @param pitch_lag decoded pitch lag
585  * @param length length of autocorrelation
586  * @param dir forward lag(1) / backward lag(-1)
587  */
588 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
589  int pitch_lag, int length, int dir)
590 {
591  int limit, ccr, lag = 0;
592  int i;
593 
594  pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
595  if (dir > 0)
596  limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
597  else
598  limit = pitch_lag + 3;
599 
600  for (i = pitch_lag - 3; i <= limit; i++) {
601  ccr = dot_product(buf, buf + dir * i, length);
602 
603  if (ccr > *ccr_max) {
604  *ccr_max = ccr;
605  lag = i;
606  }
607  }
608  return lag;
609 }
610 
611 /**
612  * Calculate pitch postfilter optimal and scaling gains.
613  *
614  * @param lag pitch postfilter forward/backward lag
615  * @param ppf pitch postfilter parameters
616  * @param cur_rate current bitrate
617  * @param tgt_eng target energy
618  * @param ccr cross-correlation
619  * @param res_eng residual energy
620  */
621 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
622  int tgt_eng, int ccr, int res_eng)
623 {
624  int pf_residual; /* square of postfiltered residual */
625  int temp1, temp2;
626 
627  ppf->index = lag;
628 
629  temp1 = tgt_eng * res_eng >> 1;
630  temp2 = ccr * ccr << 1;
631 
632  if (temp2 > temp1) {
633  if (ccr >= res_eng) {
634  ppf->opt_gain = ppf_gain_weight[cur_rate];
635  } else {
636  ppf->opt_gain = (ccr << 15) / res_eng *
637  ppf_gain_weight[cur_rate] >> 15;
638  }
639  /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
640  temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
641  temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
642  pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
643 
644  if (tgt_eng >= pf_residual << 1) {
645  temp1 = 0x7fff;
646  } else {
647  temp1 = (tgt_eng << 14) / pf_residual;
648  }
649 
650  /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
651  ppf->sc_gain = square_root(temp1 << 16);
652  } else {
653  ppf->opt_gain = 0;
654  ppf->sc_gain = 0x7fff;
655  }
656 
657  ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
658 }
659 
660 /**
661  * Calculate pitch postfilter parameters.
662  *
663  * @param p the context
664  * @param offset offset of the excitation vector
665  * @param pitch_lag decoded pitch lag
666  * @param ppf pitch postfilter parameters
667  * @param cur_rate current bitrate
668  */
669 static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
670  PPFParam *ppf, enum Rate cur_rate)
671 {
672 
673  int16_t scale;
674  int i;
675  int temp1, temp2;
676 
677  /*
678  * 0 - target energy
679  * 1 - forward cross-correlation
680  * 2 - forward residual energy
681  * 3 - backward cross-correlation
682  * 4 - backward residual energy
683  */
684  int energy[5] = {0, 0, 0, 0, 0};
685  int16_t *buf = p->audio + LPC_ORDER + offset;
686  int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
687  SUBFRAME_LEN, 1);
688  int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
689  SUBFRAME_LEN, -1);
690 
691  ppf->index = 0;
692  ppf->opt_gain = 0;
693  ppf->sc_gain = 0x7fff;
694 
695  /* Case 0, Section 3.6 */
696  if (!back_lag && !fwd_lag)
697  return;
698 
699  /* Compute target energy */
700  energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
701 
702  /* Compute forward residual energy */
703  if (fwd_lag)
704  energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
705 
706  /* Compute backward residual energy */
707  if (back_lag)
708  energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);
709 
710  /* Normalize and shorten */
711  temp1 = 0;
712  for (i = 0; i < 5; i++)
713  temp1 = FFMAX(energy[i], temp1);
714 
715  scale = normalize_bits(temp1, 31);
716  for (i = 0; i < 5; i++)
717  energy[i] = (energy[i] << scale) >> 16;
718 
719  if (fwd_lag && !back_lag) { /* Case 1 */
720  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
721  energy[2]);
722  } else if (!fwd_lag) { /* Case 2 */
723  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
724  energy[4]);
725  } else { /* Case 3 */
726 
727  /*
728  * Select the largest of energy[1]^2/energy[2]
729  * and energy[3]^2/energy[4]
730  */
731  temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
732  temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
733  if (temp1 >= temp2) {
734  comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
735  energy[2]);
736  } else {
737  comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
738  energy[4]);
739  }
740  }
741 }
742 
743 /**
744  * Classify frames as voiced/unvoiced.
745  *
746  * @param p the context
747  * @param pitch_lag decoded pitch_lag
748  * @param exc_eng excitation energy estimation
749  * @param scale scaling factor of exc_eng
750  *
751  * @return residual interpolation index if voiced, 0 otherwise
752  */
753 static int comp_interp_index(G723_1_Context *p, int pitch_lag,
754  int *exc_eng, int *scale)
755 {
756  int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
757  int16_t *buf = p->audio + LPC_ORDER;
758 
759  int index, ccr, tgt_eng, best_eng, temp;
760 
761  *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
762  buf += offset;
763 
764  /* Compute maximum backward cross-correlation */
765  ccr = 0;
766  index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
767  ccr = av_sat_add32(ccr, 1 << 15) >> 16;
768 
769  /* Compute target energy */
770  tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2);
771  *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
772 
773  if (ccr <= 0)
774  return 0;
775 
776  /* Compute best energy */
777  best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);
778  best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
779 
780  temp = best_eng * *exc_eng >> 3;
781 
782  if (temp < ccr * ccr) {
783  return index;
784  } else
785  return 0;
786 }
787 
788 /**
789  * Peform residual interpolation based on frame classification.
790  *
791  * @param buf decoded excitation vector
792  * @param out output vector
793  * @param lag decoded pitch lag
794  * @param gain interpolated gain
795  * @param rseed seed for random number generator
796  */
797 static void residual_interp(int16_t *buf, int16_t *out, int lag,
798  int gain, int *rseed)
799 {
800  int i;
801  if (lag) { /* Voiced */
802  int16_t *vector_ptr = buf + PITCH_MAX;
803  /* Attenuate */
804  for (i = 0; i < lag; i++)
805  out[i] = vector_ptr[i - lag] * 3 >> 2;
806  av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
807  (FRAME_LEN - lag) * sizeof(*out));
808  } else { /* Unvoiced */
809  for (i = 0; i < FRAME_LEN; i++) {
810  *rseed = *rseed * 521 + 259;
811  out[i] = gain * *rseed >> 15;
812  }
813  memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
814  }
815 }
816 
817 /**
818  * Perform IIR filtering.
819  *
820  * @param fir_coef FIR coefficients
821  * @param iir_coef IIR coefficients
822  * @param src source vector
823  * @param dest destination vector
824  * @param width width of the output, 16 bits(0) / 32 bits(1)
825  */
826 #define iir_filter(fir_coef, iir_coef, src, dest, width)\
827 {\
828  int m, n;\
829  int res_shift = 16 & ~-(width);\
830  int in_shift = 16 - res_shift;\
831 \
832  for (m = 0; m < SUBFRAME_LEN; m++) {\
833  int64_t filter = 0;\
834  for (n = 1; n <= LPC_ORDER; n++) {\
835  filter -= (fir_coef)[n - 1] * (src)[m - n] -\
836  (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
837  }\
838 \
839  (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\
840  (1 << 15)) >> res_shift;\
841  }\
842 }
843 
844 /**
845  * Adjust gain of postfiltered signal.
846  *
847  * @param p the context
848  * @param buf postfiltered output vector
849  * @param energy input energy coefficient
850  */
851 static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
852 {
853  int num, denom, gain, bits1, bits2;
854  int i;
855 
856  num = energy;
857  denom = 0;
858  for (i = 0; i < SUBFRAME_LEN; i++) {
859  int temp = buf[i] >> 2;
860  temp *= temp;
861  denom = av_sat_dadd32(denom, temp);
862  }
863 
864  if (num && denom) {
865  bits1 = normalize_bits(num, 31);
866  bits2 = normalize_bits(denom, 31);
867  num = num << bits1 >> 1;
868  denom <<= bits2;
869 
870  bits2 = 5 + bits1 - bits2;
871  bits2 = FFMAX(0, bits2);
872 
873  gain = (num >> 1) / (denom >> 16);
874  gain = square_root(gain << 16 >> bits2);
875  } else {
876  gain = 1 << 12;
877  }
878 
879  for (i = 0; i < SUBFRAME_LEN; i++) {
880  p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
881  buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
882  (1 << 10)) >> 11);
883  }
884 }
885 
886 /**
887  * Perform formant filtering.
888  *
889  * @param p the context
890  * @param lpc quantized lpc coefficients
891  * @param buf input buffer
892  * @param dst output buffer
893  */
894 static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
895  int16_t *buf, int16_t *dst)
896 {
897  int16_t filter_coef[2][LPC_ORDER];
898  int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
899  int i, j, k;
900 
901  memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
902  memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
903 
904  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
905  for (k = 0; k < LPC_ORDER; k++) {
906  filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
907  (1 << 14)) >> 15;
908  filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
909  (1 << 14)) >> 15;
910  }
911  iir_filter(filter_coef[0], filter_coef[1], buf + i,
912  filter_signal + i, 1);
913  lpc += LPC_ORDER;
914  }
915 
916  memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
917  memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
918 
919  buf += LPC_ORDER;
920  signal_ptr = filter_signal + LPC_ORDER;
921  for (i = 0; i < SUBFRAMES; i++) {
922  int temp;
923  int auto_corr[2];
924  int scale, energy;
925 
926  /* Normalize */
927  scale = scale_vector(dst, buf, SUBFRAME_LEN);
928 
929  /* Compute auto correlation coefficients */
930  auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
931  auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN);
932 
933  /* Compute reflection coefficient */
934  temp = auto_corr[1] >> 16;
935  if (temp) {
936  temp = (auto_corr[0] >> 2) / temp;
937  }
938  p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
939  temp = -p->reflection_coef >> 1 & ~3;
940 
941  /* Compensation filter */
942  for (j = 0; j < SUBFRAME_LEN; j++) {
943  dst[j] = av_sat_dadd32(signal_ptr[j],
944  (signal_ptr[j - 1] >> 16) * temp) >> 16;
945  }
946 
947  /* Compute normalized signal energy */
948  temp = 2 * scale + 4;
949  if (temp < 0) {
950  energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
951  } else
952  energy = auto_corr[1] >> temp;
953 
954  gain_scale(p, dst, energy);
955 
956  buf += SUBFRAME_LEN;
957  signal_ptr += SUBFRAME_LEN;
958  dst += SUBFRAME_LEN;
959  }
960 }
961 
962 static int sid_gain_to_lsp_index(int gain)
963 {
964  if (gain < 0x10)
965  return gain << 6;
966  else if (gain < 0x20)
967  return gain - 8 << 7;
968  else
969  return gain - 20 << 8;
970 }
971 
972 static inline int cng_rand(int *state, int base)
973 {
974  *state = (*state * 521 + 259) & 0xFFFF;
975  return (*state & 0x7FFF) * base >> 15;
976 }
977 
978 static int estimate_sid_gain(G723_1_Context *p)
979 {
980  int i, shift, seg, seg2, t, val, val_add, x, y;
981 
982  shift = 16 - p->cur_gain * 2;
983  if (shift > 0)
984  t = p->sid_gain << shift;
985  else
986  t = p->sid_gain >> -shift;
987  x = t * cng_filt[0] >> 16;
988 
989  if (x >= cng_bseg[2])
990  return 0x3F;
991 
992  if (x >= cng_bseg[1]) {
993  shift = 4;
994  seg = 3;
995  } else {
996  shift = 3;
997  seg = (x >= cng_bseg[0]);
998  }
999  seg2 = FFMIN(seg, 3);
1000 
1001  val = 1 << shift;
1002  val_add = val >> 1;
1003  for (i = 0; i < shift; i++) {
1004  t = seg * 32 + (val << seg2);
1005  t *= t;
1006  if (x >= t)
1007  val += val_add;
1008  else
1009  val -= val_add;
1010  val_add >>= 1;
1011  }
1012 
1013  t = seg * 32 + (val << seg2);
1014  y = t * t - x;
1015  if (y <= 0) {
1016  t = seg * 32 + (val + 1 << seg2);
1017  t = t * t - x;
1018  val = (seg2 - 1 << 4) + val;
1019  if (t >= y)
1020  val++;
1021  } else {
1022  t = seg * 32 + (val - 1 << seg2);
1023  t = t * t - x;
1024  val = (seg2 - 1 << 4) + val;
1025  if (t >= y)
1026  val--;
1027  }
1028 
1029  return val;
1030 }
1031 
1032 static void generate_noise(G723_1_Context *p)
1033 {
1034  int i, j, idx, t;
1035  int off[SUBFRAMES];
1036  int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
1037  int tmp[SUBFRAME_LEN * 2];
1038  int16_t *vector_ptr;
1039  int64_t sum;
1040  int b0, c, delta, x, shift;
1041 
1042  p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
1043  p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
1044 
1045  for (i = 0; i < SUBFRAMES; i++) {
1046  p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
1047  p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i];
1048  }
1049 
1050  for (i = 0; i < SUBFRAMES / 2; i++) {
1051  t = cng_rand(&p->cng_random_seed, 1 << 13);
1052  off[i * 2] = t & 1;
1053  off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
1054  t >>= 2;
1055  for (j = 0; j < 11; j++) {
1056  signs[i * 11 + j] = (t & 1) * 2 - 1 << 14;
1057  t >>= 1;
1058  }
1059  }
1060 
1061  idx = 0;
1062  for (i = 0; i < SUBFRAMES; i++) {
1063  for (j = 0; j < SUBFRAME_LEN / 2; j++)
1064  tmp[j] = j;
1065  t = SUBFRAME_LEN / 2;
1066  for (j = 0; j < pulses[i]; j++, idx++) {
1067  int idx2 = cng_rand(&p->cng_random_seed, t);
1068 
1069  pos[idx] = tmp[idx2] * 2 + off[i];
1070  tmp[idx2] = tmp[--t];
1071  }
1072  }
1073 
1074  vector_ptr = p->audio + LPC_ORDER;
1075  memcpy(vector_ptr, p->prev_excitation,
1076  PITCH_MAX * sizeof(*p->excitation));
1077  for (i = 0; i < SUBFRAMES; i += 2) {
1078  gen_acb_excitation(vector_ptr, vector_ptr,
1079  p->pitch_lag[i >> 1], &p->subframe[i],
1080  p->cur_rate);
1081  gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
1082  vector_ptr + SUBFRAME_LEN,
1083  p->pitch_lag[i >> 1], &p->subframe[i + 1],
1084  p->cur_rate);
1085 
1086  t = 0;
1087  for (j = 0; j < SUBFRAME_LEN * 2; j++)
1088  t |= FFABS(vector_ptr[j]);
1089  t = FFMIN(t, 0x7FFF);
1090  if (!t) {
1091  shift = 0;
1092  } else {
1093  shift = -10 + av_log2(t);
1094  if (shift < -2)
1095  shift = -2;
1096  }
1097  sum = 0;
1098  if (shift < 0) {
1099  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
1100  t = vector_ptr[j] << -shift;
1101  sum += t * t;
1102  tmp[j] = t;
1103  }
1104  } else {
1105  for (j = 0; j < SUBFRAME_LEN * 2; j++) {
1106  t = vector_ptr[j] >> shift;
1107  sum += t * t;
1108  tmp[j] = t;
1109  }
1110  }
1111 
1112  b0 = 0;
1113  for (j = 0; j < 11; j++)
1114  b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
1115  b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
1116 
1117  c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
1118  if (shift * 2 + 3 >= 0)
1119  c >>= shift * 2 + 3;
1120  else
1121  c <<= -(shift * 2 + 3);
1122  c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
1123 
1124  delta = b0 * b0 * 2 - c;
1125  if (delta <= 0) {
1126  x = -b0;
1127  } else {
1128  delta = square_root(delta);
1129  x = delta - b0;
1130  t = delta + b0;
1131  if (FFABS(t) < FFABS(x))
1132  x = -t;
1133  }
1134  shift++;
1135  if (shift < 0)
1136  x >>= -shift;
1137  else
1138  x <<= shift;
1139  x = av_clip(x, -10000, 10000);
1140 
1141  for (j = 0; j < 11; j++) {
1142  idx = (i / 2) * 11 + j;
1143  vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
1144  (x * signs[idx] >> 15));
1145  }
1146 
1147  /* copy decoded data to serve as a history for the next decoded subframes */
1148  memcpy(vector_ptr + PITCH_MAX, vector_ptr,
1149  sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
1150  vector_ptr += SUBFRAME_LEN * 2;
1151  }
1152  /* Save the excitation for the next frame */
1153  memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
1154  PITCH_MAX * sizeof(*p->excitation));
1155 }
1156 
1157 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
1158  int *got_frame_ptr, AVPacket *avpkt)
1159 {
1160  G723_1_Context *p = avctx->priv_data;
1161  const uint8_t *buf = avpkt->data;
1162  int buf_size = avpkt->size;
1163  int dec_mode = buf[0] & 3;
1164 
1165  PPFParam ppf[SUBFRAMES];
1166  int16_t cur_lsp[LPC_ORDER];
1167  int16_t lpc[SUBFRAMES * LPC_ORDER];
1168  int16_t acb_vector[SUBFRAME_LEN];
1169  int16_t *out;
1170  int bad_frame = 0, i, j, ret;
1171  int16_t *audio = p->audio;
1172 
1173  if (buf_size < frame_size[dec_mode]) {
1174  if (buf_size)
1175  av_log(avctx, AV_LOG_WARNING,
1176  "Expected %d bytes, got %d - skipping packet\n",
1177  frame_size[dec_mode], buf_size);
1178  *got_frame_ptr = 0;
1179  return buf_size;
1180  }
1181 
1182  if (unpack_bitstream(p, buf, buf_size) < 0) {
1183  bad_frame = 1;
1184  if (p->past_frame_type == ACTIVE_FRAME)
1185  p->cur_frame_type = ACTIVE_FRAME;
1186  else
1187  p->cur_frame_type = UNTRANSMITTED_FRAME;
1188  }
1189 
1190  p->frame.nb_samples = FRAME_LEN;
1191  if ((ret = ff_get_buffer(avctx, &p->frame)) < 0) {
1192  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1193  return ret;
1194  }
1195 
1196  out = (int16_t *)p->frame.data[0];
1197 
1198  if (p->cur_frame_type == ACTIVE_FRAME) {
1199  if (!bad_frame)
1200  p->erased_frames = 0;
1201  else if (p->erased_frames != 3)
1202  p->erased_frames++;
1203 
1204  inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
1205  lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
1206 
1207  /* Save the lsp_vector for the next frame */
1208  memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
1209 
1210  /* Generate the excitation for the frame */
1211  memcpy(p->excitation, p->prev_excitation,
1212  PITCH_MAX * sizeof(*p->excitation));
1213  if (!p->erased_frames) {
1214  int16_t *vector_ptr = p->excitation + PITCH_MAX;
1215 
1216  /* Update interpolation gain memory */
1217  p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
1218  p->subframe[3].amp_index) >> 1];
1219  for (i = 0; i < SUBFRAMES; i++) {
1220  gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
1221  p->pitch_lag[i >> 1], i);
1222  gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
1223  p->pitch_lag[i >> 1], &p->subframe[i],
1224  p->cur_rate);
1225  /* Get the total excitation */
1226  for (j = 0; j < SUBFRAME_LEN; j++) {
1227  int v = av_clip_int16(vector_ptr[j] << 1);
1228  vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
1229  }
1230  vector_ptr += SUBFRAME_LEN;
1231  }
1232 
1233  vector_ptr = p->excitation + PITCH_MAX;
1234 
1235  p->interp_index = comp_interp_index(p, p->pitch_lag[1],
1236  &p->sid_gain, &p->cur_gain);
1237 
1238  /* Peform pitch postfiltering */
1239  if (p->postfilter) {
1240  i = PITCH_MAX;
1241  for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1242  comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
1243  ppf + j, p->cur_rate);
1244 
1245  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1246  ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
1247  vector_ptr + i,
1248  vector_ptr + i + ppf[j].index,
1249  ppf[j].sc_gain,
1250  ppf[j].opt_gain,
1251  1 << 14, 15, SUBFRAME_LEN);
1252  } else {
1253  audio = vector_ptr - LPC_ORDER;
1254  }
1255 
1256  /* Save the excitation for the next frame */
1257  memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
1258  PITCH_MAX * sizeof(*p->excitation));
1259  } else {
1260  p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
1261  if (p->erased_frames == 3) {
1262  /* Mute output */
1263  memset(p->excitation, 0,
1264  (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
1265  memset(p->prev_excitation, 0,
1266  PITCH_MAX * sizeof(*p->excitation));
1267  memset(p->frame.data[0], 0,
1268  (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
1269  } else {
1270  int16_t *buf = p->audio + LPC_ORDER;
1271 
1272  /* Regenerate frame */
1273  residual_interp(p->excitation, buf, p->interp_index,
1274  p->interp_gain, &p->random_seed);
1275 
1276  /* Save the excitation for the next frame */
1277  memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
1278  PITCH_MAX * sizeof(*p->excitation));
1279  }
1280  }
1281  p->cng_random_seed = CNG_RANDOM_SEED;
1282  } else {
1283  if (p->cur_frame_type == SID_FRAME) {
1284  p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
1285  inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
1286  } else if (p->past_frame_type == ACTIVE_FRAME) {
1287  p->sid_gain = estimate_sid_gain(p);
1288  }
1289 
1290  if (p->past_frame_type == ACTIVE_FRAME)
1291  p->cur_gain = p->sid_gain;
1292  else
1293  p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
1294  generate_noise(p);
1295  lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
1296  /* Save the lsp_vector for the next frame */
1297  memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
1298  }
1299 
1300  p->past_frame_type = p->cur_frame_type;
1301 
1302  memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
1303  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1304  ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
1305  audio + i, SUBFRAME_LEN, LPC_ORDER,
1306  0, 1, 1 << 12);
1307  memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
1308 
1309  if (p->postfilter) {
1310  formant_postfilter(p, lpc, p->audio, out);
1311  } else { // if output is not postfiltered it should be scaled by 2
1312  for (i = 0; i < FRAME_LEN; i++)
1313  out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
1314  }
1315 
1316  *got_frame_ptr = 1;
1317  *(AVFrame *)data = p->frame;
1318 
1319  return frame_size[dec_mode];
1320 }
1321 
1322 #define OFFSET(x) offsetof(G723_1_Context, x)
1323 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
1324 
1325 static const AVOption options[] = {
1326  { "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,
1327  { .i64 = 1 }, 0, 1, AD },
1328  { NULL }
1329 };
1330 
1331 
1332 static const AVClass g723_1dec_class = {
1333  .class_name = "G.723.1 decoder",
1334  .item_name = av_default_item_name,
1335  .option = options,
1336  .version = LIBAVUTIL_VERSION_INT,
1337 };
1338 
1339 AVCodec ff_g723_1_decoder = {
1340  .name = "g723_1",
1341  .type = AVMEDIA_TYPE_AUDIO,
1342  .id = AV_CODEC_ID_G723_1,
1343  .priv_data_size = sizeof(G723_1_Context),
1344  .init = g723_1_decode_init,
1345  .decode = g723_1_decode_frame,
1346  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1347  .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
1348  .priv_class = &g723_1dec_class,
1349 };
1350 
1351 #if CONFIG_G723_1_ENCODER
1352 #define BITSTREAM_WRITER_LE
1353 #include "put_bits.h"
1354 
1355 static av_cold int g723_1_encode_init(AVCodecContext *avctx)
1356 {
1357  G723_1_Context *p = avctx->priv_data;
1358 
1359  if (avctx->sample_rate != 8000) {
1360  av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
1361  return -1;
1362  }
1363 
1364  if (avctx->channels != 1) {
1365  av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
1366  return AVERROR(EINVAL);
1367  }
1368 
1369  if (avctx->bit_rate == 6300) {
1370  p->cur_rate = RATE_6300;
1371  } else if (avctx->bit_rate == 5300) {
1372  av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");
1373  return AVERROR_PATCHWELCOME;
1374  } else {
1375  av_log(avctx, AV_LOG_ERROR,
1376  "Bitrate not supported, use 6.3k\n");
1377  return AVERROR(EINVAL);
1378  }
1379  avctx->frame_size = 240;
1380  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
1381 
1382  return 0;
1383 }
1384 
1385 /**
1386  * Remove DC component from the input signal.
1387  *
1388  * @param buf input signal
1389  * @param fir zero memory
1390  * @param iir pole memory
1391  */
1392 static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
1393 {
1394  int i;
1395  for (i = 0; i < FRAME_LEN; i++) {
1396  *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
1397  *fir = buf[i];
1398  buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
1399  }
1400 }
1401 
1402 /**
1403  * Estimate autocorrelation of the input vector.
1404  *
1405  * @param buf input buffer
1406  * @param autocorr autocorrelation coefficients vector
1407  */
1408 static void comp_autocorr(int16_t *buf, int16_t *autocorr)
1409 {
1410  int i, scale, temp;
1411  int16_t vector[LPC_FRAME];
1412 
1413  scale_vector(vector, buf, LPC_FRAME);
1414 
1415  /* Apply the Hamming window */
1416  for (i = 0; i < LPC_FRAME; i++)
1417  vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
1418 
1419  /* Compute the first autocorrelation coefficient */
1420  temp = ff_dot_product(vector, vector, LPC_FRAME);
1421 
1422  /* Apply a white noise correlation factor of (1025/1024) */
1423  temp += temp >> 10;
1424 
1425  /* Normalize */
1426  scale = normalize_bits_int32(temp);
1427  autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +
1428  (1 << 15)) >> 16;
1429 
1430  /* Compute the remaining coefficients */
1431  if (!autocorr[0]) {
1432  memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
1433  } else {
1434  for (i = 1; i <= LPC_ORDER; i++) {
1435  temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
1436  temp = MULL2((temp << scale), binomial_window[i - 1]);
1437  autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
1438  }
1439  }
1440 }
1441 
1442 /**
1443  * Use Levinson-Durbin recursion to compute LPC coefficients from
1444  * autocorrelation values.
1445  *
1446  * @param lpc LPC coefficients vector
1447  * @param autocorr autocorrelation coefficients vector
1448  * @param error prediction error
1449  */
1450 static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
1451 {
1452  int16_t vector[LPC_ORDER];
1453  int16_t partial_corr;
1454  int i, j, temp;
1455 
1456  memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
1457 
1458  for (i = 0; i < LPC_ORDER; i++) {
1459  /* Compute the partial correlation coefficient */
1460  temp = 0;
1461  for (j = 0; j < i; j++)
1462  temp -= lpc[j] * autocorr[i - j - 1];
1463  temp = ((autocorr[i] << 13) + temp) << 3;
1464 
1465  if (FFABS(temp) >= (error << 16))
1466  break;
1467 
1468  partial_corr = temp / (error << 1);
1469 
1470  lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +
1471  (1 << 15)) >> 16;
1472 
1473  /* Update the prediction error */
1474  temp = MULL2(temp, partial_corr);
1475  error = av_clipl_int32((int64_t)(error << 16) - temp +
1476  (1 << 15)) >> 16;
1477 
1478  memcpy(vector, lpc, i * sizeof(int16_t));
1479  for (j = 0; j < i; j++) {
1480  temp = partial_corr * vector[i - j - 1] << 1;
1481  lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +
1482  (1 << 15)) >> 16;
1483  }
1484  }
1485 }
1486 
1487 /**
1488  * Calculate LPC coefficients for the current frame.
1489  *
1490  * @param buf current frame
1491  * @param prev_data 2 trailing subframes of the previous frame
1492  * @param lpc LPC coefficients vector
1493  */
1494 static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
1495 {
1496  int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
1497  int16_t *autocorr_ptr = autocorr;
1498  int16_t *lpc_ptr = lpc;
1499  int i, j;
1500 
1501  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
1502  comp_autocorr(buf + i, autocorr_ptr);
1503  levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
1504 
1505  lpc_ptr += LPC_ORDER;
1506  autocorr_ptr += LPC_ORDER + 1;
1507  }
1508 }
1509 
1510 static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
1511 {
1512  int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
1513  ///< polynomials (F1, F2) ordered as
1514  ///< f1[0], f2[0], ...., f1[5], f2[5]
1515 
1516  int max, shift, cur_val, prev_val, count, p;
1517  int i, j;
1518  int64_t temp;
1519 
1520  /* Initialize f1[0] and f2[0] to 1 in Q25 */
1521  for (i = 0; i < LPC_ORDER; i++)
1522  lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
1523 
1524  /* Apply bandwidth expansion on the LPC coefficients */
1525  f[0] = f[1] = 1 << 25;
1526 
1527  /* Compute the remaining coefficients */
1528  for (i = 0; i < LPC_ORDER / 2; i++) {
1529  /* f1 */
1530  f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
1531  /* f2 */
1532  f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
1533  }
1534 
1535  /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
1536  f[LPC_ORDER] >>= 1;
1537  f[LPC_ORDER + 1] >>= 1;
1538 
1539  /* Normalize and shorten */
1540  max = FFABS(f[0]);
1541  for (i = 1; i < LPC_ORDER + 2; i++)
1542  max = FFMAX(max, FFABS(f[i]));
1543 
1544  shift = normalize_bits_int32(max);
1545 
1546  for (i = 0; i < LPC_ORDER + 2; i++)
1547  f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;
1548 
1549  /**
1550  * Evaluate F1 and F2 at uniform intervals of pi/256 along the
1551  * unit circle and check for zero crossings.
1552  */
1553  p = 0;
1554  temp = 0;
1555  for (i = 0; i <= LPC_ORDER / 2; i++)
1556  temp += f[2 * i] * cos_tab[0];
1557  prev_val = av_clipl_int32(temp << 1);
1558  count = 0;
1559  for ( i = 1; i < COS_TBL_SIZE / 2; i++) {
1560  /* Evaluate */
1561  temp = 0;
1562  for (j = 0; j <= LPC_ORDER / 2; j++)
1563  temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
1564  cur_val = av_clipl_int32(temp << 1);
1565 
1566  /* Check for sign change, indicating a zero crossing */
1567  if ((cur_val ^ prev_val) < 0) {
1568  int abs_cur = FFABS(cur_val);
1569  int abs_prev = FFABS(prev_val);
1570  int sum = abs_cur + abs_prev;
1571 
1572  shift = normalize_bits_int32(sum);
1573  sum <<= shift;
1574  abs_prev = abs_prev << shift >> 8;
1575  lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
1576 
1577  if (count == LPC_ORDER)
1578  break;
1579 
1580  /* Switch between sum and difference polynomials */
1581  p ^= 1;
1582 
1583  /* Evaluate */
1584  temp = 0;
1585  for (j = 0; j <= LPC_ORDER / 2; j++){
1586  temp += f[LPC_ORDER - 2 * j + p] *
1587  cos_tab[i * j % COS_TBL_SIZE];
1588  }
1589  cur_val = av_clipl_int32(temp<<1);
1590  }
1591  prev_val = cur_val;
1592  }
1593 
1594  if (count != LPC_ORDER)
1595  memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
1596 }
1597 
1598 /**
1599  * Quantize the current LSP subvector.
1600  *
1601  * @param num band number
1602  * @param offset offset of the current subvector in an LPC_ORDER vector
1603  * @param size size of the current subvector
1604  */
1605 #define get_index(num, offset, size) \
1606 {\
1607  int error, max = -1;\
1608  int16_t temp[4];\
1609  int i, j;\
1610  for (i = 0; i < LSP_CB_SIZE; i++) {\
1611  for (j = 0; j < size; j++){\
1612  temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
1613  (1 << 14)) >> 15;\
1614  }\
1615  error = dot_product(lsp + (offset), temp, size) << 1;\
1616  error -= dot_product(lsp_band##num[i], temp, size);\
1617  if (error > max) {\
1618  max = error;\
1619  lsp_index[num] = i;\
1620  }\
1621  }\
1622 }
1623 
1624 /**
1625  * Vector quantize the LSP frequencies.
1626  *
1627  * @param lsp the current lsp vector
1628  * @param prev_lsp the previous lsp vector
1629  */
1630 static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
1631 {
1632  int16_t weight[LPC_ORDER];
1633  int16_t min, max;
1634  int shift, i;
1635 
1636  /* Calculate the VQ weighting vector */
1637  weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
1638  weight[LPC_ORDER - 1] = (1 << 20) /
1639  (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
1640 
1641  for (i = 1; i < LPC_ORDER - 1; i++) {
1642  min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
1643  if (min > 0x20)
1644  weight[i] = (1 << 20) / min;
1645  else
1646  weight[i] = INT16_MAX;
1647  }
1648 
1649  /* Normalize */
1650  max = 0;
1651  for (i = 0; i < LPC_ORDER; i++)
1652  max = FFMAX(weight[i], max);
1653 
1654  shift = normalize_bits_int16(max);
1655  for (i = 0; i < LPC_ORDER; i++) {
1656  weight[i] <<= shift;
1657  }
1658 
1659  /* Compute the VQ target vector */
1660  for (i = 0; i < LPC_ORDER; i++) {
1661  lsp[i] -= dc_lsp[i] +
1662  (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
1663  }
1664 
1665  get_index(0, 0, 3);
1666  get_index(1, 3, 3);
1667  get_index(2, 6, 4);
1668 }
1669 
1670 /**
1671  * Apply the formant perceptual weighting filter.
1672  *
1673  * @param flt_coef filter coefficients
1674  * @param unq_lpc unquantized lpc vector
1675  */
1676 static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
1677  int16_t *unq_lpc, int16_t *buf)
1678 {
1679  int16_t vector[FRAME_LEN + LPC_ORDER];
1680  int i, j, k, l = 0;
1681 
1682  memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
1683  memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
1684  memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
1685 
1686  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
1687  for (k = 0; k < LPC_ORDER; k++) {
1688  flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
1689  (1 << 14)) >> 15;
1690  flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
1691  percept_flt_tbl[1][k] +
1692  (1 << 14)) >> 15;
1693  }
1694  iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,
1695  buf + i, 0);
1696  l += LPC_ORDER;
1697  }
1698  memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
1699  memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
1700 }
1701 
1702 /**
1703  * Estimate the open loop pitch period.
1704  *
1705  * @param buf perceptually weighted speech
1706  * @param start estimation is carried out from this position
1707  */
1708 static int estimate_pitch(int16_t *buf, int start)
1709 {
1710  int max_exp = 32;
1711  int max_ccr = 0x4000;
1712  int max_eng = 0x7fff;
1713  int index = PITCH_MIN;
1714  int offset = start - PITCH_MIN + 1;
1715 
1716  int ccr, eng, orig_eng, ccr_eng, exp;
1717  int diff, temp;
1718 
1719  int i;
1720 
1721  orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
1722 
1723  for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
1724  offset--;
1725 
1726  /* Update energy and compute correlation */
1727  orig_eng += buf[offset] * buf[offset] -
1728  buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
1729  ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
1730  if (ccr <= 0)
1731  continue;
1732 
1733  /* Split into mantissa and exponent to maintain precision */
1734  exp = normalize_bits_int32(ccr);
1735  ccr = av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;
1736  exp <<= 1;
1737  ccr *= ccr;
1738  temp = normalize_bits_int32(ccr);
1739  ccr = ccr << temp >> 16;
1740  exp += temp;
1741 
1742  temp = normalize_bits_int32(orig_eng);
1743  eng = av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;
1744  exp -= temp;
1745 
1746  if (ccr >= eng) {
1747  exp--;
1748  ccr >>= 1;
1749  }
1750  if (exp > max_exp)
1751  continue;
1752 
1753  if (exp + 1 < max_exp)
1754  goto update;
1755 
1756  /* Equalize exponents before comparison */
1757  if (exp + 1 == max_exp)
1758  temp = max_ccr >> 1;
1759  else
1760  temp = max_ccr;
1761  ccr_eng = ccr * max_eng;
1762  diff = ccr_eng - eng * temp;
1763  if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
1764 update:
1765  index = i;
1766  max_exp = exp;
1767  max_ccr = ccr;
1768  max_eng = eng;
1769  }
1770  }
1771  return index;
1772 }
1773 
1774 /**
1775  * Compute harmonic noise filter parameters.
1776  *
1777  * @param buf perceptually weighted speech
1778  * @param pitch_lag open loop pitch period
1779  * @param hf harmonic filter parameters
1780  */
1781 static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
1782 {
1783  int ccr, eng, max_ccr, max_eng;
1784  int exp, max, diff;
1785  int energy[15];
1786  int i, j;
1787 
1788  for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
1789  /* Compute residual energy */
1790  energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
1791  /* Compute correlation */
1792  energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
1793  }
1794 
1795  /* Compute target energy */
1796  energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
1797 
1798  /* Normalize */
1799  max = 0;
1800  for (i = 0; i < 15; i++)
1801  max = FFMAX(max, FFABS(energy[i]));
1802 
1803  exp = normalize_bits_int32(max);
1804  for (i = 0; i < 15; i++) {
1805  energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
1806  (1 << 15)) >> 16;
1807  }
1808 
1809  hf->index = -1;
1810  hf->gain = 0;
1811  max_ccr = 1;
1812  max_eng = 0x7fff;
1813 
1814  for (i = 0; i <= 6; i++) {
1815  eng = energy[i << 1];
1816  ccr = energy[(i << 1) + 1];
1817 
1818  if (ccr <= 0)
1819  continue;
1820 
1821  ccr = (ccr * ccr + (1 << 14)) >> 15;
1822  diff = ccr * max_eng - eng * max_ccr;
1823  if (diff > 0) {
1824  max_ccr = ccr;
1825  max_eng = eng;
1826  hf->index = i;
1827  }
1828  }
1829 
1830  if (hf->index == -1) {
1831  hf->index = pitch_lag;
1832  return;
1833  }
1834 
1835  eng = energy[14] * max_eng;
1836  eng = (eng >> 2) + (eng >> 3);
1837  ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
1838  if (eng < ccr) {
1839  eng = energy[(hf->index << 1) + 1];
1840 
1841  if (eng >= max_eng)
1842  hf->gain = 0x2800;
1843  else
1844  hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
1845  }
1846  hf->index += pitch_lag - 3;
1847 }
1848 
1849 /**
1850  * Apply the harmonic noise shaping filter.
1851  *
1852  * @param hf filter parameters
1853  */
1854 static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
1855 {
1856  int i;
1857 
1858  for (i = 0; i < SUBFRAME_LEN; i++) {
1859  int64_t temp = hf->gain * src[i - hf->index] << 1;
1860  dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
1861  }
1862 }
1863 
1864 static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
1865 {
1866  int i;
1867  for (i = 0; i < SUBFRAME_LEN; i++) {
1868  int64_t temp = hf->gain * src[i - hf->index] << 1;
1869  dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
1870  (1 << 15)) >> 16;
1871 
1872  }
1873 }
1874 
1875 /**
1876  * Combined synthesis and formant perceptual weighting filer.
1877  *
1878  * @param qnt_lpc quantized lpc coefficients
1879  * @param perf_lpc perceptual filter coefficients
1880  * @param perf_fir perceptual filter fir memory
1881  * @param perf_iir perceptual filter iir memory
1882  * @param scale the filter output will be scaled by 2^scale
1883  */
1884 static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
1885  int16_t *perf_fir, int16_t *perf_iir,
1886  const int16_t *src, int16_t *dest, int scale)
1887 {
1888  int i, j;
1889  int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
1890  int64_t buf[SUBFRAME_LEN];
1891 
1892  int16_t *bptr_16 = buf_16 + LPC_ORDER;
1893 
1894  memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
1895  memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
1896 
1897  for (i = 0; i < SUBFRAME_LEN; i++) {
1898  int64_t temp = 0;
1899  for (j = 1; j <= LPC_ORDER; j++)
1900  temp -= qnt_lpc[j - 1] * bptr_16[i - j];
1901 
1902  buf[i] = (src[i] << 15) + (temp << 3);
1903  bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
1904  }
1905 
1906  for (i = 0; i < SUBFRAME_LEN; i++) {
1907  int64_t fir = 0, iir = 0;
1908  for (j = 1; j <= LPC_ORDER; j++) {
1909  fir -= perf_lpc[j - 1] * bptr_16[i - j];
1910  iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
1911  }
1912  dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
1913  (1 << 15)) >> 16;
1914  }
1915  memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
1916  memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
1917  sizeof(int16_t) * LPC_ORDER);
1918 }
1919 
1920 /**
1921  * Compute the adaptive codebook contribution.
1922  *
1923  * @param buf input signal
1924  * @param index the current subframe index
1925  */
1926 static void acb_search(G723_1_Context *p, int16_t *residual,
1927  int16_t *impulse_resp, const int16_t *buf,
1928  int index)
1929 {
1930 
1931  int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
1932 
1933  const int16_t *cb_tbl = adaptive_cb_gain85;
1934 
1935  int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
1936 
1937  int pitch_lag = p->pitch_lag[index >> 1];
1938  int acb_lag = 1;
1939  int acb_gain = 0;
1940  int odd_frame = index & 1;
1941  int iter = 3 + odd_frame;
1942  int count = 0;
1943  int tbl_size = 85;
1944 
1945  int i, j, k, l, max;
1946  int64_t temp;
1947 
1948  if (!odd_frame) {
1949  if (pitch_lag == PITCH_MIN)
1950  pitch_lag++;
1951  else
1952  pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
1953  }
1954 
1955  for (i = 0; i < iter; i++) {
1956  get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
1957 
1958  for (j = 0; j < SUBFRAME_LEN; j++) {
1959  temp = 0;
1960  for (k = 0; k <= j; k++)
1961  temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
1962  flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
1963  (1 << 15)) >> 16;
1964  }
1965 
1966  for (j = PITCH_ORDER - 2; j >= 0; j--) {
1967  flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
1968  for (k = 1; k < SUBFRAME_LEN; k++) {
1969  temp = (flt_buf[j + 1][k - 1] << 15) +
1970  residual[j] * impulse_resp[k];
1971  flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
1972  }
1973  }
1974 
1975  /* Compute crosscorrelation with the signal */
1976  for (j = 0; j < PITCH_ORDER; j++) {
1977  temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
1978  ccr_buf[count++] = av_clipl_int32(temp << 1);
1979  }
1980 
1981  /* Compute energies */
1982  for (j = 0; j < PITCH_ORDER; j++) {
1983  ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
1984  SUBFRAME_LEN);
1985  }
1986 
1987  for (j = 1; j < PITCH_ORDER; j++) {
1988  for (k = 0; k < j; k++) {
1989  temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
1990  ccr_buf[count++] = av_clipl_int32(temp<<2);
1991  }
1992  }
1993  }
1994 
1995  /* Normalize and shorten */
1996  max = 0;
1997  for (i = 0; i < 20 * iter; i++)
1998  max = FFMAX(max, FFABS(ccr_buf[i]));
1999 
2000  temp = normalize_bits_int32(max);
2001 
2002  for (i = 0; i < 20 * iter; i++){
2003  ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +
2004  (1 << 15)) >> 16;
2005  }
2006 
2007  max = 0;
2008  for (i = 0; i < iter; i++) {
2009  /* Select quantization table */
2010  if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
2011  odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
2012  cb_tbl = adaptive_cb_gain170;
2013  tbl_size = 170;
2014  }
2015 
2016  for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
2017  temp = 0;
2018  for (l = 0; l < 20; l++)
2019  temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
2020  temp = av_clipl_int32(temp);
2021 
2022  if (temp > max) {
2023  max = temp;
2024  acb_gain = j;
2025  acb_lag = i;
2026  }
2027  }
2028  }
2029 
2030  if (!odd_frame) {
2031  pitch_lag += acb_lag - 1;
2032  acb_lag = 1;
2033  }
2034 
2035  p->pitch_lag[index >> 1] = pitch_lag;
2036  p->subframe[index].ad_cb_lag = acb_lag;
2037  p->subframe[index].ad_cb_gain = acb_gain;
2038 }
2039 
2040 /**
2041  * Subtract the adaptive codebook contribution from the input
2042  * to obtain the residual.
2043  *
2044  * @param buf target vector
2045  */
2046 static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
2047  int16_t *buf)
2048 {
2049  int i, j;
2050  /* Subtract adaptive CB contribution to obtain the residual */
2051  for (i = 0; i < SUBFRAME_LEN; i++) {
2052  int64_t temp = buf[i] << 14;
2053  for (j = 0; j <= i; j++)
2054  temp -= residual[j] * impulse_resp[i - j];
2055 
2056  buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
2057  }
2058 }
2059 
2060 /**
2061  * Quantize the residual signal using the fixed codebook (MP-MLQ).
2062  *
2063  * @param optim optimized fixed codebook parameters
2064  * @param buf excitation vector
2065  */
2066 static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
2067  int16_t *buf, int pulse_cnt, int pitch_lag)
2068 {
2069  FCBParam param;
2070  int16_t impulse_r[SUBFRAME_LEN];
2071  int16_t temp_corr[SUBFRAME_LEN];
2072  int16_t impulse_corr[SUBFRAME_LEN];
2073 
2074  int ccr1[SUBFRAME_LEN];
2075  int ccr2[SUBFRAME_LEN];
2076  int amp, err, max, max_amp_index, min, scale, i, j, k, l;
2077 
2078  int64_t temp;
2079 
2080  /* Update impulse response */
2081  memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
2082  param.dirac_train = 0;
2083  if (pitch_lag < SUBFRAME_LEN - 2) {
2084  param.dirac_train = 1;
2085  gen_dirac_train(impulse_r, pitch_lag);
2086  }
2087 
2088  for (i = 0; i < SUBFRAME_LEN; i++)
2089  temp_corr[i] = impulse_r[i] >> 1;
2090 
2091  /* Compute impulse response autocorrelation */
2092  temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
2093 
2094  scale = normalize_bits_int32(temp);
2095  impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
2096 
2097  for (i = 1; i < SUBFRAME_LEN; i++) {
2098  temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
2099  impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
2100  }
2101 
2102  /* Compute crosscorrelation of impulse response with residual signal */
2103  scale -= 4;
2104  for (i = 0; i < SUBFRAME_LEN; i++){
2105  temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
2106  if (scale < 0)
2107  ccr1[i] = temp >> -scale;
2108  else
2109  ccr1[i] = av_clipl_int32(temp << scale);
2110  }
2111 
2112  /* Search loop */
2113  for (i = 0; i < GRID_SIZE; i++) {
2114  /* Maximize the crosscorrelation */
2115  max = 0;
2116  for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
2117  temp = FFABS(ccr1[j]);
2118  if (temp >= max) {
2119  max = temp;
2120  param.pulse_pos[0] = j;
2121  }
2122  }
2123 
2124  /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
2125  amp = max;
2126  min = 1 << 30;
2127  max_amp_index = GAIN_LEVELS - 2;
2128  for (j = max_amp_index; j >= 2; j--) {
2129  temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *
2130  impulse_corr[0] << 1);
2131  temp = FFABS(temp - amp);
2132  if (temp < min) {
2133  min = temp;
2134  max_amp_index = j;
2135  }
2136  }
2137 
2138  max_amp_index--;
2139  /* Select additional gain values */
2140  for (j = 1; j < 5; j++) {
2141  for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
2142  temp_corr[k] = 0;
2143  ccr2[k] = ccr1[k];
2144  }
2145  param.amp_index = max_amp_index + j - 2;
2146  amp = fixed_cb_gain[param.amp_index];
2147 
2148  param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
2149  temp_corr[param.pulse_pos[0]] = 1;
2150 
2151  for (k = 1; k < pulse_cnt; k++) {
2152  max = -1 << 30;
2153  for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
2154  if (temp_corr[l])
2155  continue;
2156  temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
2157  temp = av_clipl_int32((int64_t)temp *
2158  param.pulse_sign[k - 1] << 1);
2159  ccr2[l] -= temp;
2160  temp = FFABS(ccr2[l]);
2161  if (temp > max) {
2162  max = temp;
2163  param.pulse_pos[k] = l;
2164  }
2165  }
2166 
2167  param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
2168  -amp : amp;
2169  temp_corr[param.pulse_pos[k]] = 1;
2170  }
2171 
2172  /* Create the error vector */
2173  memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
2174 
2175  for (k = 0; k < pulse_cnt; k++)
2176  temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
2177 
2178  for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
2179  temp = 0;
2180  for (l = 0; l <= k; l++) {
2181  int prod = av_clipl_int32((int64_t)temp_corr[l] *
2182  impulse_r[k - l] << 1);
2183  temp = av_clipl_int32(temp + prod);
2184  }
2185  temp_corr[k] = temp << 2 >> 16;
2186  }
2187 
2188  /* Compute square of error */
2189  err = 0;
2190  for (k = 0; k < SUBFRAME_LEN; k++) {
2191  int64_t prod;
2192  prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);
2193  err = av_clipl_int32(err - prod);
2194  prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);
2195  err = av_clipl_int32(err + prod);
2196  }
2197 
2198  /* Minimize */
2199  if (err < optim->min_err) {
2200  optim->min_err = err;
2201  optim->grid_index = i;
2202  optim->amp_index = param.amp_index;
2203  optim->dirac_train = param.dirac_train;
2204 
2205  for (k = 0; k < pulse_cnt; k++) {
2206  optim->pulse_sign[k] = param.pulse_sign[k];
2207  optim->pulse_pos[k] = param.pulse_pos[k];
2208  }
2209  }
2210  }
2211  }
2212 }
2213 
2214 /**
2215  * Encode the pulse position and gain of the current subframe.
2216  *
2217  * @param optim optimized fixed CB parameters
2218  * @param buf excitation vector
2219  */
2220 static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
2221  int16_t *buf, int pulse_cnt)
2222 {
2223  int i, j;
2224 
2225  j = PULSE_MAX - pulse_cnt;
2226 
2227  subfrm->pulse_sign = 0;
2228  subfrm->pulse_pos = 0;
2229 
2230  for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
2231  int val = buf[optim->grid_index + (i << 1)];
2232  if (!val) {
2233  subfrm->pulse_pos += combinatorial_table[j][i];
2234  } else {
2235  subfrm->pulse_sign <<= 1;
2236  if (val < 0) subfrm->pulse_sign++;
2237  j++;
2238 
2239  if (j == PULSE_MAX) break;
2240  }
2241  }
2242  subfrm->amp_index = optim->amp_index;
2243  subfrm->grid_index = optim->grid_index;
2244  subfrm->dirac_train = optim->dirac_train;
2245 }
2246 
2247 /**
2248  * Compute the fixed codebook excitation.
2249  *
2250  * @param buf target vector
2251  * @param impulse_resp impulse response of the combined filter
2252  */
2253 static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
2254  int16_t *buf, int index)
2255 {
2256  FCBParam optim;
2257  int pulse_cnt = pulses[index];
2258  int i;
2259 
2260  optim.min_err = 1 << 30;
2261  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
2262 
2263  if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
2264  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
2265  p->pitch_lag[index >> 1]);
2266  }
2267 
2268  /* Reconstruct the excitation */
2269  memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
2270  for (i = 0; i < pulse_cnt; i++)
2271  buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
2272 
2273  pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
2274 
2275  if (optim.dirac_train)
2276  gen_dirac_train(buf, p->pitch_lag[index >> 1]);
2277 }
2278 
2279 /**
2280  * Pack the frame parameters into output bitstream.
2281  *
2282  * @param frame output buffer
2283  * @param size size of the buffer
2284  */
2285 static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)
2286 {
2287  PutBitContext pb;
2288  int info_bits, i, temp;
2289 
2290  init_put_bits(&pb, frame, size);
2291 
2292  if (p->cur_rate == RATE_6300) {
2293  info_bits = 0;
2294  put_bits(&pb, 2, info_bits);
2295  }
2296 
2297  put_bits(&pb, 8, p->lsp_index[2]);
2298  put_bits(&pb, 8, p->lsp_index[1]);
2299  put_bits(&pb, 8, p->lsp_index[0]);
2300 
2301  put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
2302  put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
2303  put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
2304  put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
2305 
2306  /* Write 12 bit combined gain */
2307  for (i = 0; i < SUBFRAMES; i++) {
2308  temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
2309  p->subframe[i].amp_index;
2310  if (p->cur_rate == RATE_6300)
2311  temp += p->subframe[i].dirac_train << 11;
2312  put_bits(&pb, 12, temp);
2313  }
2314 
2315  put_bits(&pb, 1, p->subframe[0].grid_index);
2316  put_bits(&pb, 1, p->subframe[1].grid_index);
2317  put_bits(&pb, 1, p->subframe[2].grid_index);
2318  put_bits(&pb, 1, p->subframe[3].grid_index);
2319 
2320  if (p->cur_rate == RATE_6300) {
2321  skip_put_bits(&pb, 1); /* reserved bit */
2322 
2323  /* Write 13 bit combined position index */
2324  temp = (p->subframe[0].pulse_pos >> 16) * 810 +
2325  (p->subframe[1].pulse_pos >> 14) * 90 +
2326  (p->subframe[2].pulse_pos >> 16) * 9 +
2327  (p->subframe[3].pulse_pos >> 14);
2328  put_bits(&pb, 13, temp);
2329 
2330  put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
2331  put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
2332  put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
2333  put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
2334 
2335  put_bits(&pb, 6, p->subframe[0].pulse_sign);
2336  put_bits(&pb, 5, p->subframe[1].pulse_sign);
2337  put_bits(&pb, 6, p->subframe[2].pulse_sign);
2338  put_bits(&pb, 5, p->subframe[3].pulse_sign);
2339  }
2340 
2341  flush_put_bits(&pb);
2342  return frame_size[info_bits];
2343 }
2344 
2345 static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
2346  const AVFrame *frame, int *got_packet_ptr)
2347 {
2348  G723_1_Context *p = avctx->priv_data;
2349  int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
2350  int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
2351  int16_t cur_lsp[LPC_ORDER];
2352  int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
2353  int16_t vector[FRAME_LEN + PITCH_MAX];
2354  int offset, ret;
2355  int16_t *in = (const int16_t *)frame->data[0];
2356 
2357  HFParam hf[4];
2358  int i, j;
2359 
2360  highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
2361 
2362  memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
2363  memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
2364 
2365  comp_lpc_coeff(vector, unq_lpc);
2366  lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
2367  lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
2368 
2369  /* Update memory */
2370  memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
2371  sizeof(int16_t) * SUBFRAME_LEN);
2372  memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
2373  sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
2374  memcpy(p->prev_data, in + HALF_FRAME_LEN,
2375  sizeof(int16_t) * HALF_FRAME_LEN);
2376  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
2377 
2378  perceptual_filter(p, weighted_lpc, unq_lpc, vector);
2379 
2380  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
2381  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
2382  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
2383 
2384  scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
2385 
2386  p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
2387  p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
2388 
2389  for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
2390  comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
2391 
2392  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
2393  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
2394  memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
2395 
2396  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
2397  harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
2398 
2399  inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
2400  lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
2401 
2402  memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
2403 
2404  offset = 0;
2405  for (i = 0; i < SUBFRAMES; i++) {
2406  int16_t impulse_resp[SUBFRAME_LEN];
2407  int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
2408  int16_t flt_in[SUBFRAME_LEN];
2409  int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
2410 
2411  /**
2412  * Compute the combined impulse response of the synthesis filter,
2413  * formant perceptual weighting filter and harmonic noise shaping filter
2414  */
2415  memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
2416  memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
2417  memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
2418 
2419  flt_in[0] = 1 << 13; /* Unit impulse */
2420  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
2421  zero, zero, flt_in, vector + PITCH_MAX, 1);
2422  harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
2423 
2424  /* Compute the combined zero input response */
2425  flt_in[0] = 0;
2426  memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
2427  memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
2428 
2429  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
2430  fir, iir, flt_in, vector + PITCH_MAX, 0);
2431  memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
2432  harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
2433 
2434  acb_search(p, residual, impulse_resp, in, i);
2435  gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
2436  &p->subframe[i], p->cur_rate);
2437  sub_acb_contrib(residual, impulse_resp, in);
2438 
2439  fcb_search(p, impulse_resp, in, i);
2440 
2441  /* Reconstruct the excitation */
2442  gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
2443  &p->subframe[i], RATE_6300);
2444 
2445  memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
2446  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
2447  for (j = 0; j < SUBFRAME_LEN; j++)
2448  in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
2449  memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
2450  sizeof(int16_t) * SUBFRAME_LEN);
2451 
2452  /* Update filter memories */
2453  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
2454  p->perf_fir_mem, p->perf_iir_mem,
2455  in, vector + PITCH_MAX, 0);
2456  memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
2457  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
2458  memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
2459  sizeof(int16_t) * SUBFRAME_LEN);
2460 
2461  in += SUBFRAME_LEN;
2462  offset += LPC_ORDER;
2463  }
2464 
2465  if ((ret = ff_alloc_packet2(avctx, avpkt, 24)))
2466  return ret;
2467 
2468  *got_packet_ptr = 1;
2469  avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);
2470  return 0;
2471 }
2472 
2473 AVCodec ff_g723_1_encoder = {
2474  .name = "g723_1",
2475  .type = AVMEDIA_TYPE_AUDIO,
2476  .id = AV_CODEC_ID_G723_1,
2477  .priv_data_size = sizeof(G723_1_Context),
2478  .init = g723_1_encode_init,
2479  .encode2 = g723_1_encode_frame,
2480  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
2481  .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
2482  AV_SAMPLE_FMT_NONE},
2483 };
2484 #endif
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