FFmpeg  4.3.6
g723_1enc.c
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1 /*
2  * G.723.1 compatible encoder
3  * Copyright (c) Mohamed Naufal <naufal22@gmail.com>
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * G.723.1 compatible encoder
25  */
26 
27 #include <stdint.h>
28 #include <string.h>
29 
31 #include "libavutil/common.h"
32 #include "libavutil/mem.h"
33 #include "libavutil/opt.h"
34 
35 #include "avcodec.h"
36 #include "celp_math.h"
37 #include "g723_1.h"
38 #include "internal.h"
39 
40 #define BITSTREAM_WRITER_LE
41 #include "put_bits.h"
42 
44 {
45  G723_1_Context *s = avctx->priv_data;
46  G723_1_ChannelContext *p = &s->ch[0];
47 
48  if (avctx->sample_rate != 8000) {
49  av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
50  return AVERROR(EINVAL);
51  }
52 
53  if (avctx->channels != 1) {
54  av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
55  return AVERROR(EINVAL);
56  }
57 
58  if (avctx->bit_rate == 6300) {
59  p->cur_rate = RATE_6300;
60  } else if (avctx->bit_rate == 5300) {
61  av_log(avctx, AV_LOG_ERROR, "Use bitrate 6300 instead of 5300.\n");
62  avpriv_report_missing_feature(avctx, "Bitrate 5300");
63  return AVERROR_PATCHWELCOME;
64  } else {
65  av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\n");
66  return AVERROR(EINVAL);
67  }
68  avctx->frame_size = 240;
69  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
70 
71  return 0;
72 }
73 
74 /**
75  * Remove DC component from the input signal.
76  *
77  * @param buf input signal
78  * @param fir zero memory
79  * @param iir pole memory
80  */
81 static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
82 {
83  int i;
84  for (i = 0; i < FRAME_LEN; i++) {
85  *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
86  *fir = buf[i];
87  buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
88  }
89 }
90 
91 /**
92  * Estimate autocorrelation of the input vector.
93  *
94  * @param buf input buffer
95  * @param autocorr autocorrelation coefficients vector
96  */
97 static void comp_autocorr(int16_t *buf, int16_t *autocorr)
98 {
99  int i, scale, temp;
100  int16_t vector[LPC_FRAME];
101 
102  ff_g723_1_scale_vector(vector, buf, LPC_FRAME);
103 
104  /* Apply the Hamming window */
105  for (i = 0; i < LPC_FRAME; i++)
106  vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
107 
108  /* Compute the first autocorrelation coefficient */
109  temp = ff_dot_product(vector, vector, LPC_FRAME);
110 
111  /* Apply a white noise correlation factor of (1025/1024) */
112  temp += temp >> 10;
113 
114  /* Normalize */
115  scale = ff_g723_1_normalize_bits(temp, 31);
116  autocorr[0] = av_clipl_int32((int64_t) (temp << scale) +
117  (1 << 15)) >> 16;
118 
119  /* Compute the remaining coefficients */
120  if (!autocorr[0]) {
121  memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
122  } else {
123  for (i = 1; i <= LPC_ORDER; i++) {
124  temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
125  temp = MULL2((temp << scale), binomial_window[i - 1]);
126  autocorr[i] = av_clipl_int32((int64_t) temp + (1 << 15)) >> 16;
127  }
128  }
129 }
130 
131 /**
132  * Use Levinson-Durbin recursion to compute LPC coefficients from
133  * autocorrelation values.
134  *
135  * @param lpc LPC coefficients vector
136  * @param autocorr autocorrelation coefficients vector
137  * @param error prediction error
138  */
139 static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
140 {
141  int16_t vector[LPC_ORDER];
142  int16_t partial_corr;
143  int i, j, temp;
144 
145  memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
146 
147  for (i = 0; i < LPC_ORDER; i++) {
148  /* Compute the partial correlation coefficient */
149  temp = 0;
150  for (j = 0; j < i; j++)
151  temp -= lpc[j] * autocorr[i - j - 1];
152  temp = ((autocorr[i] << 13) + temp) << 3;
153 
154  if (FFABS(temp) >= (error << 16))
155  break;
156 
157  partial_corr = temp / (error << 1);
158 
159  lpc[i] = av_clipl_int32((int64_t) (partial_corr << 14) +
160  (1 << 15)) >> 16;
161 
162  /* Update the prediction error */
163  temp = MULL2(temp, partial_corr);
164  error = av_clipl_int32((int64_t) (error << 16) - temp +
165  (1 << 15)) >> 16;
166 
167  memcpy(vector, lpc, i * sizeof(int16_t));
168  for (j = 0; j < i; j++) {
169  temp = partial_corr * vector[i - j - 1] << 1;
170  lpc[j] = av_clipl_int32((int64_t) (lpc[j] << 16) - temp +
171  (1 << 15)) >> 16;
172  }
173  }
174 }
175 
176 /**
177  * Calculate LPC coefficients for the current frame.
178  *
179  * @param buf current frame
180  * @param prev_data 2 trailing subframes of the previous frame
181  * @param lpc LPC coefficients vector
182  */
183 static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
184 {
185  int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
186  int16_t *autocorr_ptr = autocorr;
187  int16_t *lpc_ptr = lpc;
188  int i, j;
189 
190  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
191  comp_autocorr(buf + i, autocorr_ptr);
192  levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
193 
194  lpc_ptr += LPC_ORDER;
195  autocorr_ptr += LPC_ORDER + 1;
196  }
197 }
198 
199 static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
200 {
201  int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
202  ///< polynomials (F1, F2) ordered as
203  ///< f1[0], f2[0], ...., f1[5], f2[5]
204 
205  int max, shift, cur_val, prev_val, count, p;
206  int i, j;
207  int64_t temp;
208 
209  /* Initialize f1[0] and f2[0] to 1 in Q25 */
210  for (i = 0; i < LPC_ORDER; i++)
211  lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
212 
213  /* Apply bandwidth expansion on the LPC coefficients */
214  f[0] = f[1] = 1 << 25;
215 
216  /* Compute the remaining coefficients */
217  for (i = 0; i < LPC_ORDER / 2; i++) {
218  /* f1 */
219  f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
220  /* f2 */
221  f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
222  }
223 
224  /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
225  f[LPC_ORDER] >>= 1;
226  f[LPC_ORDER + 1] >>= 1;
227 
228  /* Normalize and shorten */
229  max = FFABS(f[0]);
230  for (i = 1; i < LPC_ORDER + 2; i++)
231  max = FFMAX(max, FFABS(f[i]));
232 
233  shift = ff_g723_1_normalize_bits(max, 31);
234 
235  for (i = 0; i < LPC_ORDER + 2; i++)
236  f[i] = av_clipl_int32((int64_t) (f[i] << shift) + (1 << 15)) >> 16;
237 
238  /**
239  * Evaluate F1 and F2 at uniform intervals of pi/256 along the
240  * unit circle and check for zero crossings.
241  */
242  p = 0;
243  temp = 0;
244  for (i = 0; i <= LPC_ORDER / 2; i++)
245  temp += f[2 * i] * cos_tab[0];
246  prev_val = av_clipl_int32(temp << 1);
247  count = 0;
248  for (i = 1; i < COS_TBL_SIZE / 2; i++) {
249  /* Evaluate */
250  temp = 0;
251  for (j = 0; j <= LPC_ORDER / 2; j++)
252  temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
253  cur_val = av_clipl_int32(temp << 1);
254 
255  /* Check for sign change, indicating a zero crossing */
256  if ((cur_val ^ prev_val) < 0) {
257  int abs_cur = FFABS(cur_val);
258  int abs_prev = FFABS(prev_val);
259  int sum = abs_cur + abs_prev;
260 
261  shift = ff_g723_1_normalize_bits(sum, 31);
262  sum <<= shift;
263  abs_prev = abs_prev << shift >> 8;
264  lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
265 
266  if (count == LPC_ORDER)
267  break;
268 
269  /* Switch between sum and difference polynomials */
270  p ^= 1;
271 
272  /* Evaluate */
273  temp = 0;
274  for (j = 0; j <= LPC_ORDER / 2; j++)
275  temp += f[LPC_ORDER - 2 * j + p] *
276  cos_tab[i * j % COS_TBL_SIZE];
277  cur_val = av_clipl_int32(temp << 1);
278  }
279  prev_val = cur_val;
280  }
281 
282  if (count != LPC_ORDER)
283  memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
284 }
285 
286 /**
287  * Quantize the current LSP subvector.
288  *
289  * @param num band number
290  * @param offset offset of the current subvector in an LPC_ORDER vector
291  * @param size size of the current subvector
292  */
293 #define get_index(num, offset, size) \
294 { \
295  int error, max = -1; \
296  int16_t temp[4]; \
297  int i, j; \
298  \
299  for (i = 0; i < LSP_CB_SIZE; i++) { \
300  for (j = 0; j < size; j++){ \
301  temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] + \
302  (1 << 14)) >> 15; \
303  } \
304  error = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1; \
305  error -= ff_g723_1_dot_product(lsp_band##num[i], temp, size); \
306  if (error > max) { \
307  max = error; \
308  lsp_index[num] = i; \
309  } \
310  } \
311 }
312 
313 /**
314  * Vector quantize the LSP frequencies.
315  *
316  * @param lsp the current lsp vector
317  * @param prev_lsp the previous lsp vector
318  */
319 static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
320 {
321  int16_t weight[LPC_ORDER];
322  int16_t min, max;
323  int shift, i;
324 
325  /* Calculate the VQ weighting vector */
326  weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
327  weight[LPC_ORDER - 1] = (1 << 20) /
328  (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
329 
330  for (i = 1; i < LPC_ORDER - 1; i++) {
331  min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
332  if (min > 0x20)
333  weight[i] = (1 << 20) / min;
334  else
335  weight[i] = INT16_MAX;
336  }
337 
338  /* Normalize */
339  max = 0;
340  for (i = 0; i < LPC_ORDER; i++)
341  max = FFMAX(weight[i], max);
342 
343  shift = ff_g723_1_normalize_bits(max, 15);
344  for (i = 0; i < LPC_ORDER; i++) {
345  weight[i] <<= shift;
346  }
347 
348  /* Compute the VQ target vector */
349  for (i = 0; i < LPC_ORDER; i++) {
350  lsp[i] -= dc_lsp[i] +
351  (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
352  }
353 
354  get_index(0, 0, 3);
355  get_index(1, 3, 3);
356  get_index(2, 6, 4);
357 }
358 
359 /**
360  * Perform IIR filtering.
361  *
362  * @param fir_coef FIR coefficients
363  * @param iir_coef IIR coefficients
364  * @param src source vector
365  * @param dest destination vector
366  */
367 static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
368  int16_t *src, int16_t *dest)
369 {
370  int m, n;
371 
372  for (m = 0; m < SUBFRAME_LEN; m++) {
373  int64_t filter = 0;
374  for (n = 1; n <= LPC_ORDER; n++) {
375  filter -= fir_coef[n - 1] * src[m - n] -
376  iir_coef[n - 1] * dest[m - n];
377  }
378 
379  dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) +
380  (1 << 15)) >> 16;
381  }
382 }
383 
384 /**
385  * Apply the formant perceptual weighting filter.
386  *
387  * @param flt_coef filter coefficients
388  * @param unq_lpc unquantized lpc vector
389  */
390 static void perceptual_filter(G723_1_ChannelContext *p, int16_t *flt_coef,
391  int16_t *unq_lpc, int16_t *buf)
392 {
393  int16_t vector[FRAME_LEN + LPC_ORDER];
394  int i, j, k, l = 0;
395 
396  memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
397  memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
398  memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
399 
400  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
401  for (k = 0; k < LPC_ORDER; k++) {
402  flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
403  (1 << 14)) >> 15;
404  flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
405  percept_flt_tbl[1][k] +
406  (1 << 14)) >> 15;
407  }
408  iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER,
409  vector + i, buf + i);
410  l += LPC_ORDER;
411  }
412  memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
413  memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
414 }
415 
416 /**
417  * Estimate the open loop pitch period.
418  *
419  * @param buf perceptually weighted speech
420  * @param start estimation is carried out from this position
421  */
422 static int estimate_pitch(int16_t *buf, int start)
423 {
424  int max_exp = 32;
425  int max_ccr = 0x4000;
426  int max_eng = 0x7fff;
427  int index = PITCH_MIN;
428  int offset = start - PITCH_MIN + 1;
429 
430  int ccr, eng, orig_eng, ccr_eng, exp;
431  int diff, temp;
432 
433  int i;
434 
435  orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
436 
437  for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
438  offset--;
439 
440  /* Update energy and compute correlation */
441  orig_eng += buf[offset] * buf[offset] -
442  buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
443  ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
444  if (ccr <= 0)
445  continue;
446 
447  /* Split into mantissa and exponent to maintain precision */
448  exp = ff_g723_1_normalize_bits(ccr, 31);
449  ccr = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16;
450  exp <<= 1;
451  ccr *= ccr;
452  temp = ff_g723_1_normalize_bits(ccr, 31);
453  ccr = ccr << temp >> 16;
454  exp += temp;
455 
456  temp = ff_g723_1_normalize_bits(orig_eng, 31);
457  eng = av_clipl_int32((int64_t) (orig_eng << temp) + (1 << 15)) >> 16;
458  exp -= temp;
459 
460  if (ccr >= eng) {
461  exp--;
462  ccr >>= 1;
463  }
464  if (exp > max_exp)
465  continue;
466 
467  if (exp + 1 < max_exp)
468  goto update;
469 
470  /* Equalize exponents before comparison */
471  if (exp + 1 == max_exp)
472  temp = max_ccr >> 1;
473  else
474  temp = max_ccr;
475  ccr_eng = ccr * max_eng;
476  diff = ccr_eng - eng * temp;
477  if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
478 update:
479  index = i;
480  max_exp = exp;
481  max_ccr = ccr;
482  max_eng = eng;
483  }
484  }
485  return index;
486 }
487 
488 /**
489  * Compute harmonic noise filter parameters.
490  *
491  * @param buf perceptually weighted speech
492  * @param pitch_lag open loop pitch period
493  * @param hf harmonic filter parameters
494  */
495 static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
496 {
497  int ccr, eng, max_ccr, max_eng;
498  int exp, max, diff;
499  int energy[15];
500  int i, j;
501 
502  for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
503  /* Compute residual energy */
504  energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
505  /* Compute correlation */
506  energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
507  }
508 
509  /* Compute target energy */
510  energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
511 
512  /* Normalize */
513  max = 0;
514  for (i = 0; i < 15; i++)
515  max = FFMAX(max, FFABS(energy[i]));
516 
517  exp = ff_g723_1_normalize_bits(max, 31);
518  for (i = 0; i < 15; i++) {
519  energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
520  (1 << 15)) >> 16;
521  }
522 
523  hf->index = -1;
524  hf->gain = 0;
525  max_ccr = 1;
526  max_eng = 0x7fff;
527 
528  for (i = 0; i <= 6; i++) {
529  eng = energy[i << 1];
530  ccr = energy[(i << 1) + 1];
531 
532  if (ccr <= 0)
533  continue;
534 
535  ccr = (ccr * ccr + (1 << 14)) >> 15;
536  diff = ccr * max_eng - eng * max_ccr;
537  if (diff > 0) {
538  max_ccr = ccr;
539  max_eng = eng;
540  hf->index = i;
541  }
542  }
543 
544  if (hf->index == -1) {
545  hf->index = pitch_lag;
546  return;
547  }
548 
549  eng = energy[14] * max_eng;
550  eng = (eng >> 2) + (eng >> 3);
551  ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
552  if (eng < ccr) {
553  eng = energy[(hf->index << 1) + 1];
554 
555  if (eng >= max_eng)
556  hf->gain = 0x2800;
557  else
558  hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
559  }
560  hf->index += pitch_lag - 3;
561 }
562 
563 /**
564  * Apply the harmonic noise shaping filter.
565  *
566  * @param hf filter parameters
567  */
568 static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
569 {
570  int i;
571 
572  for (i = 0; i < SUBFRAME_LEN; i++) {
573  int64_t temp = hf->gain * src[i - hf->index] << 1;
574  dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
575  }
576 }
577 
578 static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
579 {
580  int i;
581  for (i = 0; i < SUBFRAME_LEN; i++) {
582  int64_t temp = hf->gain * src[i - hf->index] << 1;
583  dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
584  (1 << 15)) >> 16;
585  }
586 }
587 
588 /**
589  * Combined synthesis and formant perceptual weighting filer.
590  *
591  * @param qnt_lpc quantized lpc coefficients
592  * @param perf_lpc perceptual filter coefficients
593  * @param perf_fir perceptual filter fir memory
594  * @param perf_iir perceptual filter iir memory
595  * @param scale the filter output will be scaled by 2^scale
596  */
597 static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
598  int16_t *perf_fir, int16_t *perf_iir,
599  const int16_t *src, int16_t *dest, int scale)
600 {
601  int i, j;
602  int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
603  int64_t buf[SUBFRAME_LEN];
604 
605  int16_t *bptr_16 = buf_16 + LPC_ORDER;
606 
607  memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
608  memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
609 
610  for (i = 0; i < SUBFRAME_LEN; i++) {
611  int64_t temp = 0;
612  for (j = 1; j <= LPC_ORDER; j++)
613  temp -= qnt_lpc[j - 1] * bptr_16[i - j];
614 
615  buf[i] = (src[i] << 15) + (temp << 3);
616  bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
617  }
618 
619  for (i = 0; i < SUBFRAME_LEN; i++) {
620  int64_t fir = 0, iir = 0;
621  for (j = 1; j <= LPC_ORDER; j++) {
622  fir -= perf_lpc[j - 1] * bptr_16[i - j];
623  iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
624  }
625  dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
626  (1 << 15)) >> 16;
627  }
628  memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
629  memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
630  sizeof(int16_t) * LPC_ORDER);
631 }
632 
633 /**
634  * Compute the adaptive codebook contribution.
635  *
636  * @param buf input signal
637  * @param index the current subframe index
638  */
639 static void acb_search(G723_1_ChannelContext *p, int16_t *residual,
640  int16_t *impulse_resp, const int16_t *buf,
641  int index)
642 {
643  int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
644 
645  const int16_t *cb_tbl = adaptive_cb_gain85;
646 
647  int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
648 
649  int pitch_lag = p->pitch_lag[index >> 1];
650  int acb_lag = 1;
651  int acb_gain = 0;
652  int odd_frame = index & 1;
653  int iter = 3 + odd_frame;
654  int count = 0;
655  int tbl_size = 85;
656 
657  int i, j, k, l, max;
658  int64_t temp;
659 
660  if (!odd_frame) {
661  if (pitch_lag == PITCH_MIN)
662  pitch_lag++;
663  else
664  pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
665  }
666 
667  for (i = 0; i < iter; i++) {
668  ff_g723_1_get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
669 
670  for (j = 0; j < SUBFRAME_LEN; j++) {
671  temp = 0;
672  for (k = 0; k <= j; k++)
673  temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
674  flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
675  (1 << 15)) >> 16;
676  }
677 
678  for (j = PITCH_ORDER - 2; j >= 0; j--) {
679  flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
680  for (k = 1; k < SUBFRAME_LEN; k++) {
681  temp = (flt_buf[j + 1][k - 1] << 15) +
682  residual[j] * impulse_resp[k];
683  flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
684  }
685  }
686 
687  /* Compute crosscorrelation with the signal */
688  for (j = 0; j < PITCH_ORDER; j++) {
689  temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
690  ccr_buf[count++] = av_clipl_int32(temp << 1);
691  }
692 
693  /* Compute energies */
694  for (j = 0; j < PITCH_ORDER; j++) {
695  ccr_buf[count++] = ff_g723_1_dot_product(flt_buf[j], flt_buf[j],
696  SUBFRAME_LEN);
697  }
698 
699  for (j = 1; j < PITCH_ORDER; j++) {
700  for (k = 0; k < j; k++) {
701  temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
702  ccr_buf[count++] = av_clipl_int32(temp << 2);
703  }
704  }
705  }
706 
707  /* Normalize and shorten */
708  max = 0;
709  for (i = 0; i < 20 * iter; i++)
710  max = FFMAX(max, FFABS(ccr_buf[i]));
711 
712  temp = ff_g723_1_normalize_bits(max, 31);
713 
714  for (i = 0; i < 20 * iter; i++)
715  ccr_buf[i] = av_clipl_int32((int64_t) (ccr_buf[i] << temp) +
716  (1 << 15)) >> 16;
717 
718  max = 0;
719  for (i = 0; i < iter; i++) {
720  /* Select quantization table */
721  if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
722  odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
723  cb_tbl = adaptive_cb_gain170;
724  tbl_size = 170;
725  }
726 
727  for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
728  temp = 0;
729  for (l = 0; l < 20; l++)
730  temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
731  temp = av_clipl_int32(temp);
732 
733  if (temp > max) {
734  max = temp;
735  acb_gain = j;
736  acb_lag = i;
737  }
738  }
739  }
740 
741  if (!odd_frame) {
742  pitch_lag += acb_lag - 1;
743  acb_lag = 1;
744  }
745 
746  p->pitch_lag[index >> 1] = pitch_lag;
747  p->subframe[index].ad_cb_lag = acb_lag;
748  p->subframe[index].ad_cb_gain = acb_gain;
749 }
750 
751 /**
752  * Subtract the adaptive codebook contribution from the input
753  * to obtain the residual.
754  *
755  * @param buf target vector
756  */
757 static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
758  int16_t *buf)
759 {
760  int i, j;
761  /* Subtract adaptive CB contribution to obtain the residual */
762  for (i = 0; i < SUBFRAME_LEN; i++) {
763  int64_t temp = buf[i] << 14;
764  for (j = 0; j <= i; j++)
765  temp -= residual[j] * impulse_resp[i - j];
766 
767  buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
768  }
769 }
770 
771 /**
772  * Quantize the residual signal using the fixed codebook (MP-MLQ).
773  *
774  * @param optim optimized fixed codebook parameters
775  * @param buf excitation vector
776  */
777 static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
778  int16_t *buf, int pulse_cnt, int pitch_lag)
779 {
780  FCBParam param;
781  int16_t impulse_r[SUBFRAME_LEN];
782  int16_t temp_corr[SUBFRAME_LEN];
783  int16_t impulse_corr[SUBFRAME_LEN];
784 
785  int ccr1[SUBFRAME_LEN];
786  int ccr2[SUBFRAME_LEN];
787  int amp, err, max, max_amp_index, min, scale, i, j, k, l;
788 
789  int64_t temp;
790 
791  /* Update impulse response */
792  memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
793  param.dirac_train = 0;
794  if (pitch_lag < SUBFRAME_LEN - 2) {
795  param.dirac_train = 1;
796  ff_g723_1_gen_dirac_train(impulse_r, pitch_lag);
797  }
798 
799  for (i = 0; i < SUBFRAME_LEN; i++)
800  temp_corr[i] = impulse_r[i] >> 1;
801 
802  /* Compute impulse response autocorrelation */
803  temp = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
804 
805  scale = ff_g723_1_normalize_bits(temp, 31);
806  impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
807 
808  for (i = 1; i < SUBFRAME_LEN; i++) {
809  temp = ff_g723_1_dot_product(temp_corr + i, temp_corr,
810  SUBFRAME_LEN - i);
811  impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
812  }
813 
814  /* Compute crosscorrelation of impulse response with residual signal */
815  scale -= 4;
816  for (i = 0; i < SUBFRAME_LEN; i++) {
817  temp = ff_g723_1_dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
818  if (scale < 0)
819  ccr1[i] = temp >> -scale;
820  else
821  ccr1[i] = av_clipl_int32(temp << scale);
822  }
823 
824  /* Search loop */
825  for (i = 0; i < GRID_SIZE; i++) {
826  /* Maximize the crosscorrelation */
827  max = 0;
828  for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
829  temp = FFABS(ccr1[j]);
830  if (temp >= max) {
831  max = temp;
832  param.pulse_pos[0] = j;
833  }
834  }
835 
836  /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
837  amp = max;
838  min = 1 << 30;
839  max_amp_index = GAIN_LEVELS - 2;
840  for (j = max_amp_index; j >= 2; j--) {
841  temp = av_clipl_int32((int64_t) fixed_cb_gain[j] *
842  impulse_corr[0] << 1);
843  temp = FFABS(temp - amp);
844  if (temp < min) {
845  min = temp;
846  max_amp_index = j;
847  }
848  }
849 
850  max_amp_index--;
851  /* Select additional gain values */
852  for (j = 1; j < 5; j++) {
853  for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
854  temp_corr[k] = 0;
855  ccr2[k] = ccr1[k];
856  }
857  param.amp_index = max_amp_index + j - 2;
858  amp = fixed_cb_gain[param.amp_index];
859 
860  param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
861  temp_corr[param.pulse_pos[0]] = 1;
862 
863  for (k = 1; k < pulse_cnt; k++) {
864  max = INT_MIN;
865  for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
866  if (temp_corr[l])
867  continue;
868  temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
869  temp = av_clipl_int32((int64_t) temp *
870  param.pulse_sign[k - 1] << 1);
871  ccr2[l] -= temp;
872  temp = FFABS(ccr2[l]);
873  if (temp > max) {
874  max = temp;
875  param.pulse_pos[k] = l;
876  }
877  }
878 
879  param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
880  -amp : amp;
881  temp_corr[param.pulse_pos[k]] = 1;
882  }
883 
884  /* Create the error vector */
885  memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
886 
887  for (k = 0; k < pulse_cnt; k++)
888  temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
889 
890  for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
891  temp = 0;
892  for (l = 0; l <= k; l++) {
893  int prod = av_clipl_int32((int64_t) temp_corr[l] *
894  impulse_r[k - l] << 1);
895  temp = av_clipl_int32(temp + prod);
896  }
897  temp_corr[k] = temp << 2 >> 16;
898  }
899 
900  /* Compute square of error */
901  err = 0;
902  for (k = 0; k < SUBFRAME_LEN; k++) {
903  int64_t prod;
904  prod = av_clipl_int32((int64_t) buf[k] * temp_corr[k] << 1);
905  err = av_clipl_int32(err - prod);
906  prod = av_clipl_int32((int64_t) temp_corr[k] * temp_corr[k]);
907  err = av_clipl_int32(err + prod);
908  }
909 
910  /* Minimize */
911  if (err < optim->min_err) {
912  optim->min_err = err;
913  optim->grid_index = i;
914  optim->amp_index = param.amp_index;
915  optim->dirac_train = param.dirac_train;
916 
917  for (k = 0; k < pulse_cnt; k++) {
918  optim->pulse_sign[k] = param.pulse_sign[k];
919  optim->pulse_pos[k] = param.pulse_pos[k];
920  }
921  }
922  }
923  }
924 }
925 
926 /**
927  * Encode the pulse position and gain of the current subframe.
928  *
929  * @param optim optimized fixed CB parameters
930  * @param buf excitation vector
931  */
932 static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
933  int16_t *buf, int pulse_cnt)
934 {
935  int i, j;
936 
937  j = PULSE_MAX - pulse_cnt;
938 
939  subfrm->pulse_sign = 0;
940  subfrm->pulse_pos = 0;
941 
942  for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
943  int val = buf[optim->grid_index + (i << 1)];
944  if (!val) {
945  subfrm->pulse_pos += combinatorial_table[j][i];
946  } else {
947  subfrm->pulse_sign <<= 1;
948  if (val < 0)
949  subfrm->pulse_sign++;
950  j++;
951 
952  if (j == PULSE_MAX)
953  break;
954  }
955  }
956  subfrm->amp_index = optim->amp_index;
957  subfrm->grid_index = optim->grid_index;
958  subfrm->dirac_train = optim->dirac_train;
959 }
960 
961 /**
962  * Compute the fixed codebook excitation.
963  *
964  * @param buf target vector
965  * @param impulse_resp impulse response of the combined filter
966  */
967 static void fcb_search(G723_1_ChannelContext *p, int16_t *impulse_resp,
968  int16_t *buf, int index)
969 {
970  FCBParam optim;
971  int pulse_cnt = pulses[index];
972  int i;
973 
974  optim.min_err = 1 << 30;
975  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
976 
977  if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
978  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
979  p->pitch_lag[index >> 1]);
980  }
981 
982  /* Reconstruct the excitation */
983  memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
984  for (i = 0; i < pulse_cnt; i++)
985  buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
986 
987  pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
988 
989  if (optim.dirac_train)
990  ff_g723_1_gen_dirac_train(buf, p->pitch_lag[index >> 1]);
991 }
992 
993 /**
994  * Pack the frame parameters into output bitstream.
995  *
996  * @param frame output buffer
997  * @param size size of the buffer
998  */
1000 {
1001  PutBitContext pb;
1002  int info_bits = 0;
1003  int i, temp;
1004 
1005  init_put_bits(&pb, avpkt->data, avpkt->size);
1006 
1007  put_bits(&pb, 2, info_bits);
1008 
1009  put_bits(&pb, 8, p->lsp_index[2]);
1010  put_bits(&pb, 8, p->lsp_index[1]);
1011  put_bits(&pb, 8, p->lsp_index[0]);
1012 
1013  put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
1014  put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
1015  put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
1016  put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
1017 
1018  /* Write 12 bit combined gain */
1019  for (i = 0; i < SUBFRAMES; i++) {
1020  temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
1021  p->subframe[i].amp_index;
1022  if (p->cur_rate == RATE_6300)
1023  temp += p->subframe[i].dirac_train << 11;
1024  put_bits(&pb, 12, temp);
1025  }
1026 
1027  put_bits(&pb, 1, p->subframe[0].grid_index);
1028  put_bits(&pb, 1, p->subframe[1].grid_index);
1029  put_bits(&pb, 1, p->subframe[2].grid_index);
1030  put_bits(&pb, 1, p->subframe[3].grid_index);
1031 
1032  if (p->cur_rate == RATE_6300) {
1033  skip_put_bits(&pb, 1); /* reserved bit */
1034 
1035  /* Write 13 bit combined position index */
1036  temp = (p->subframe[0].pulse_pos >> 16) * 810 +
1037  (p->subframe[1].pulse_pos >> 14) * 90 +
1038  (p->subframe[2].pulse_pos >> 16) * 9 +
1039  (p->subframe[3].pulse_pos >> 14);
1040  put_bits(&pb, 13, temp);
1041 
1042  put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
1043  put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
1044  put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
1045  put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
1046 
1047  put_bits(&pb, 6, p->subframe[0].pulse_sign);
1048  put_bits(&pb, 5, p->subframe[1].pulse_sign);
1049  put_bits(&pb, 6, p->subframe[2].pulse_sign);
1050  put_bits(&pb, 5, p->subframe[3].pulse_sign);
1051  }
1052 
1053  flush_put_bits(&pb);
1054  return frame_size[info_bits];
1055 }
1056 
1057 static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
1058  const AVFrame *frame, int *got_packet_ptr)
1059 {
1060  G723_1_Context *s = avctx->priv_data;
1061  G723_1_ChannelContext *p = &s->ch[0];
1062  int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
1063  int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
1064  int16_t cur_lsp[LPC_ORDER];
1065  int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
1066  int16_t vector[FRAME_LEN + PITCH_MAX];
1067  int offset, ret, i, j;
1068  int16_t *in, *start;
1069  HFParam hf[4];
1070 
1071  /* duplicate input */
1072  start = in = av_malloc(frame->nb_samples * sizeof(int16_t));
1073  if (!in)
1074  return AVERROR(ENOMEM);
1075  memcpy(in, frame->data[0], frame->nb_samples * sizeof(int16_t));
1076 
1077  highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
1078 
1079  memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
1080  memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
1081 
1082  comp_lpc_coeff(vector, unq_lpc);
1083  lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
1084  lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
1085 
1086  /* Update memory */
1087  memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
1088  sizeof(int16_t) * SUBFRAME_LEN);
1089  memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
1090  sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
1091  memcpy(p->prev_data, in + HALF_FRAME_LEN,
1092  sizeof(int16_t) * HALF_FRAME_LEN);
1093  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
1094 
1095  perceptual_filter(p, weighted_lpc, unq_lpc, vector);
1096 
1097  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
1098  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
1099  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
1100 
1101  ff_g723_1_scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
1102 
1103  p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
1104  p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
1105 
1106  for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1107  comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
1108 
1109  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
1110  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
1111  memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
1112 
1113  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1114  harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
1115 
1116  ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
1117  ff_g723_1_lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
1118 
1119  memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
1120 
1121  offset = 0;
1122  for (i = 0; i < SUBFRAMES; i++) {
1123  int16_t impulse_resp[SUBFRAME_LEN];
1124  int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
1125  int16_t flt_in[SUBFRAME_LEN];
1126  int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
1127 
1128  /**
1129  * Compute the combined impulse response of the synthesis filter,
1130  * formant perceptual weighting filter and harmonic noise shaping filter
1131  */
1132  memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
1133  memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
1134  memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
1135 
1136  flt_in[0] = 1 << 13; /* Unit impulse */
1137  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
1138  zero, zero, flt_in, vector + PITCH_MAX, 1);
1139  harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
1140 
1141  /* Compute the combined zero input response */
1142  flt_in[0] = 0;
1143  memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
1144  memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
1145 
1146  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
1147  fir, iir, flt_in, vector + PITCH_MAX, 0);
1148  memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
1149  harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
1150 
1151  acb_search(p, residual, impulse_resp, in, i);
1153  p->pitch_lag[i >> 1], &p->subframe[i],
1154  p->cur_rate);
1155  sub_acb_contrib(residual, impulse_resp, in);
1156 
1157  fcb_search(p, impulse_resp, in, i);
1158 
1159  /* Reconstruct the excitation */
1161  p->pitch_lag[i >> 1], &p->subframe[i],
1162  RATE_6300);
1163 
1164  memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
1165  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
1166  for (j = 0; j < SUBFRAME_LEN; j++)
1167  in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
1168  memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
1169  sizeof(int16_t) * SUBFRAME_LEN);
1170 
1171  /* Update filter memories */
1172  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
1173  p->perf_fir_mem, p->perf_iir_mem,
1174  in, vector + PITCH_MAX, 0);
1175  memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
1176  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
1177  memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
1178  sizeof(int16_t) * SUBFRAME_LEN);
1179 
1180  in += SUBFRAME_LEN;
1181  offset += LPC_ORDER;
1182  }
1183 
1184  av_free(start);
1185 
1186  if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0)
1187  return ret;
1188 
1189  *got_packet_ptr = 1;
1190  avpkt->size = pack_bitstream(p, avpkt);
1191  return 0;
1192 }
1193 
1194 static const AVCodecDefault defaults[] = {
1195  { "b", "6300" },
1196  { NULL },
1197 };
1198 
1200  .name = "g723_1",
1201  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1202  .type = AVMEDIA_TYPE_AUDIO,
1203  .id = AV_CODEC_ID_G723_1,
1204  .priv_data_size = sizeof(G723_1_Context),
1206  .encode2 = g723_1_encode_frame,
1207  .defaults = defaults,
1208  .sample_fmts = (const enum AVSampleFormat[]) {
1210  },
1211 };
#define NULL
Definition: coverity.c:32
static void perceptual_filter(G723_1_ChannelContext *p, int16_t *flt_coef, int16_t *unq_lpc, int16_t *buf)
Apply the formant perceptual weighting filter.
Definition: g723_1enc.c:390
static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim, int16_t *buf, int pulse_cnt)
Encode the pulse position and gain of the current subframe.
Definition: g723_1enc.c:932
#define COS_TBL_SIZE
Definition: g723_1.h:49
static void comp_autocorr(int16_t *buf, int16_t *autocorr)
Estimate autocorrelation of the input vector.
Definition: g723_1enc.c:97
static int shift(int a, int b)
Definition: sonic.c:82
int grid_index
Definition: g723_1.h:113
int dirac_train
Definition: g723_1.h:83
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
int ad_cb_gain
Definition: g723_1.h:82
int amp_index
Definition: g723_1.h:112
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:208
int64_t bit_rate
the average bitrate
Definition: avcodec.h:576
Memory handling functions.
else temp
Definition: vf_mcdeint.c:256
G723_1_Subframe subframe[4]
Definition: g723_1.h:120
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
G723.1 unpacked data subframe.
Definition: g723_1.h:80
static float cos_tab[256]
Definition: dca_lbr.c:123
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition: g723_1.h:723
int size
Definition: packet.h:356
static void skip_put_bits(PutBitContext *s, int n)
Skip the given number of bits.
Definition: put_bits.h:346
#define PITCH_ORDER
Definition: g723_1.h:45
int min_err
Definition: g723_1.h:111
int index
Definition: g723_1.h:103
static void error(const char *err)
AVCodec.
Definition: codec.h:190
#define PITCH_MIN
Definition: g723_1.h:43
int pulse_pos[PULSE_MAX]
Definition: g723_1.h:115
#define FRAME_LEN
Definition: g723_1.h:37
uint8_t lsp_index[LSP_BANDS]
Definition: g723_1.h:124
int dirac_train
Definition: g723_1.h:114
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:32
void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, uint8_t *lsp_index, int bad_frame)
Perform inverse quantization of LSP frequencies.
Definition: g723_1.c:201
static void filter(int16_t *output, ptrdiff_t out_stride, int16_t *low, ptrdiff_t low_stride, int16_t *high, ptrdiff_t high_stride, int len, int clip)
Definition: cfhd.c:196
static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp, int16_t *buf)
Subtract the adaptive codebook contribution from the input to obtain the residual.
Definition: g723_1enc.c:757
uint8_t
#define av_cold
Definition: attributes.h:88
#define av_malloc(s)
Optimized fixed codebook excitation parameters.
Definition: g723_1.h:110
AVOptions.
#define f(width, name)
Definition: cbs_vp9.c:255
#define LPC_ORDER
Definition: g723_1.h:40
int hpf_iir_mem
and iir memories
Definition: g723_1.h:152
static const int16_t adaptive_cb_gain85[85 *20]
Definition: g723_1.h:736
static AVFrame * frame
int pulse_sign
Definition: g723_1.h:84
uint8_t * data
Definition: packet.h:355
static const int16_t percept_flt_tbl[2][LPC_ORDER]
0.5^i scaled by 2^15
Definition: g723_1.h:1431
static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
Use Levinson-Durbin recursion to compute LPC coefficients from autocorrelation values.
Definition: g723_1enc.c:139
int16_t prev_data[HALF_FRAME_LEN]
Definition: g723_1.h:148
void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
Quantize LSP frequencies by interpolation and convert them to the corresponding LPC coefficients...
Definition: g723_1.c:180
#define max(a, b)
Definition: cuda_runtime.h:33
#define GRID_SIZE
Definition: g723_1.h:46
#define av_log(a,...)
static int pack_bitstream(G723_1_ChannelContext *p, AVPacket *avpkt)
Pack the frame parameters into output bitstream.
Definition: g723_1enc.c:999
static const int16_t adaptive_cb_gain170[170 *20]
Definition: g723_1.h:952
#define src
Definition: vp8dsp.c:254
static const int32_t combinatorial_table[PULSE_MAX][SUBFRAME_LEN/GRID_SIZE]
Used for the coding/decoding of the pulses positions for the MP-MLQ codebook.
Definition: g723_1.h:630
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVCodec ff_g723_1_encoder
Definition: g723_1enc.c:1199
static av_always_inline void update(SilenceDetectContext *s, AVFrame *insamples, int is_silence, int current_sample, int64_t nb_samples_notify, AVRational time_base)
int ff_g723_1_normalize_bits(int num, int width)
Calculate the number of left-shifts required for normalizing the input.
Definition: g723_1.c:49
#define AVERROR(e)
Definition: error.h:43
static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: g723_1enc.c:1057
int amp_index
Definition: g723_1.h:86
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:188
#define LPC_FRAME
Definition: g723_1.h:39
void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag)
Generate a train of dirac functions with period as pitch lag.
Definition: g723_1.c:74
int pulse_sign[PULSE_MAX]
Definition: g723_1.h:116
#define zero
Definition: regdef.h:64
int grid_index
Definition: g723_1.h:85
const char * name
Name of the codec implementation.
Definition: codec.h:197
int64_t ff_dot_product(const int16_t *a, const int16_t *b, int length)
Calculate the dot product of 2 int16_t vectors.
Definition: celp_math.c:98
static void acb_search(G723_1_ChannelContext *p, int16_t *residual, int16_t *impulse_resp, const int16_t *buf, int index)
Compute the adaptive codebook contribution.
Definition: g723_1enc.c:639
static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
Apply the harmonic noise shaping filter.
Definition: g723_1enc.c:568
static const uint8_t offset[127][2]
Definition: vf_spp.c:93
static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
Vector quantize the LSP frequencies.
Definition: g723_1enc.c:319
#define FFMAX(a, b)
Definition: common.h:94
int8_t exp
Definition: eval.c:72
void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, int pitch_lag, G723_1_Subframe *subfrm, enum Rate cur_rate)
Generate adaptive codebook excitation.
Definition: g723_1.c:86
#define PITCH_MAX
Definition: g723_1.h:44
static const int16_t fixed_cb_gain[GAIN_LEVELS]
Definition: g723_1.h:730
static av_cold int g723_1_encode_init(AVCodecContext *avctx)
Definition: g723_1enc.c:43
static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
Definition: g723_1enc.c:578
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
static void iir_filter(int16_t *fir_coef, int16_t *iir_coef, int16_t *src, int16_t *dest)
Perform IIR filtering.
Definition: g723_1enc.c:367
static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
Calculate LPC coefficients for the current frame.
Definition: g723_1enc.c:183
void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
Get delayed contribution from the previous excitation vector.
Definition: g723_1.c:60
int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length)
Definition: g723_1.c:54
#define HALF_FRAME_LEN
Definition: g723_1.h:38
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define s(width, name)
Definition: cbs_vp9.c:257
int iir_mem[LPC_ORDER]
Definition: g723_1.h:134
#define GAIN_LEVELS
Definition: g723_1.h:48
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1206
int16_t harmonic_mem[PITCH_MAX]
Definition: g723_1.h:156
int frame_size
Definition: mxfenc.c:2137
int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length)
Scale vector contents based on the largest of their absolutes.
Definition: g723_1.c:32
Libavcodec external API header.
static const int16_t dc_lsp[LPC_ORDER]
LSP DC component.
Definition: g723_1.h:232
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int sample_rate
samples per second
Definition: avcodec.h:1186
main external API structure.
Definition: avcodec.h:526
static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
Remove DC component from the input signal.
Definition: g723_1enc.c:81
G.723.1 types, functions and data tables.
int16_t fir_mem[LPC_ORDER]
Definition: g723_1.h:133
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
#define PULSE_MAX
Definition: dss_sp.c:32
static const int16_t hamming_window[LPC_FRAME]
Hamming window coefficients scaled by 2^15.
Definition: g723_1.h:1393
int index
Definition: gxfenc.c:89
G723_1_ChannelContext ch[2]
Definition: g723_1.h:163
int16_t prev_lsp[LPC_ORDER]
Definition: g723_1.h:128
static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc, int16_t *perf_fir, int16_t *perf_iir, const int16_t *src, int16_t *dest, int scale)
Combined synthesis and formant perceptual weighting filer.
Definition: g723_1enc.c:597
static int weight(int i, int blen, int offset)
Definition: diracdec.c:1560
#define SUBFRAME_LEN
Definition: g723_1.h:36
static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp, int16_t *buf, int pulse_cnt, int pitch_lag)
Quantize the residual signal using the fixed codebook (MP-MLQ).
Definition: g723_1enc.c:777
#define get_index(num, offset, size)
Quantize the current LSP subvector.
Definition: g723_1enc.c:293
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:314
static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
Definition: g723_1enc.c:199
#define SUBFRAMES
Definition: dcaenc.c:50
int16_t perf_fir_mem[LPC_ORDER]
perceptual filter fir
Definition: g723_1.h:153
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
common internal and external API header
signed 16 bits
Definition: samplefmt.h:61
static int estimate_pitch(int16_t *buf, int start)
Estimate the open loop pitch period.
Definition: g723_1enc.c:422
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
int16_t hpf_fir_mem
highpass filter fir
Definition: g723_1.h:151
int16_t prev_weight_sig[PITCH_MAX]
Definition: g723_1.h:149
Harmonic filter parameters.
Definition: g723_1.h:102
void * priv_data
Definition: avcodec.h:553
static void fcb_search(G723_1_ChannelContext *p, int16_t *impulse_resp, int16_t *buf, int index)
Compute the fixed codebook excitation.
Definition: g723_1enc.c:967
static av_always_inline int diff(const uint32_t a, const uint32_t b)
#define av_free(p)
static const AVCodecDefault defaults[]
Definition: g723_1enc.c:1194
#define MULL2(a, b)
Bitexact implementation of 2ab scaled by 1/2^16.
Definition: g723_1.h:57
int channels
number of audio channels
Definition: avcodec.h:1187
enum Rate cur_rate
Definition: g723_1.h:123
static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
Compute harmonic noise filter parameters.
Definition: g723_1enc.c:495
int pulse_pos
Definition: g723_1.h:87
static const int16_t bandwidth_expand[LPC_ORDER]
0.994^i scaled by 2^15
Definition: g723_1.h:1424
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:731
int16_t perf_iir_mem[LPC_ORDER]
and iir memories
Definition: g723_1.h:154
int16_t prev_excitation[PITCH_MAX]
Definition: g723_1.h:130
float min
static double val(void *priv, double ch)
Definition: aeval.c:76
This structure stores compressed data.
Definition: packet.h:332
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
for(j=16;j >0;--j)
static const int16_t binomial_window[LPC_ORDER]
Binomial window coefficients scaled by 2^15.
Definition: g723_1.h:1417
int ad_cb_lag
adaptive codebook lag
Definition: g723_1.h:81
int gain
Definition: g723_1.h:104
bitstream writer API