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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 *p = avctx->priv_data;
46 
47  if (avctx->sample_rate != 8000) {
48  av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
49  return AVERROR(EINVAL);
50  }
51 
52  if (avctx->channels != 1) {
53  av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
54  return AVERROR(EINVAL);
55  }
56 
57  if (avctx->bit_rate == 6300) {
58  p->cur_rate = RATE_6300;
59  } else if (avctx->bit_rate == 5300) {
60  av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6300\n");
61  return AVERROR_PATCHWELCOME;
62  } else {
63  av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\n");
64  return AVERROR(EINVAL);
65  }
66  avctx->frame_size = 240;
67  memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
68 
69  return 0;
70 }
71 
72 /**
73  * Remove DC component from the input signal.
74  *
75  * @param buf input signal
76  * @param fir zero memory
77  * @param iir pole memory
78  */
79 static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
80 {
81  int i;
82  for (i = 0; i < FRAME_LEN; i++) {
83  *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
84  *fir = buf[i];
85  buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
86  }
87 }
88 
89 /**
90  * Estimate autocorrelation of the input vector.
91  *
92  * @param buf input buffer
93  * @param autocorr autocorrelation coefficients vector
94  */
95 static void comp_autocorr(int16_t *buf, int16_t *autocorr)
96 {
97  int i, scale, temp;
98  int16_t vector[LPC_FRAME];
99 
100  ff_g723_1_scale_vector(vector, buf, LPC_FRAME);
101 
102  /* Apply the Hamming window */
103  for (i = 0; i < LPC_FRAME; i++)
104  vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
105 
106  /* Compute the first autocorrelation coefficient */
107  temp = ff_dot_product(vector, vector, LPC_FRAME);
108 
109  /* Apply a white noise correlation factor of (1025/1024) */
110  temp += temp >> 10;
111 
112  /* Normalize */
113  scale = ff_g723_1_normalize_bits(temp, 31);
114  autocorr[0] = av_clipl_int32((int64_t) (temp << scale) +
115  (1 << 15)) >> 16;
116 
117  /* Compute the remaining coefficients */
118  if (!autocorr[0]) {
119  memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
120  } else {
121  for (i = 1; i <= LPC_ORDER; i++) {
122  temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
123  temp = MULL2((temp << scale), binomial_window[i - 1]);
124  autocorr[i] = av_clipl_int32((int64_t) temp + (1 << 15)) >> 16;
125  }
126  }
127 }
128 
129 /**
130  * Use Levinson-Durbin recursion to compute LPC coefficients from
131  * autocorrelation values.
132  *
133  * @param lpc LPC coefficients vector
134  * @param autocorr autocorrelation coefficients vector
135  * @param error prediction error
136  */
137 static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
138 {
139  int16_t vector[LPC_ORDER];
140  int16_t partial_corr;
141  int i, j, temp;
142 
143  memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
144 
145  for (i = 0; i < LPC_ORDER; i++) {
146  /* Compute the partial correlation coefficient */
147  temp = 0;
148  for (j = 0; j < i; j++)
149  temp -= lpc[j] * autocorr[i - j - 1];
150  temp = ((autocorr[i] << 13) + temp) << 3;
151 
152  if (FFABS(temp) >= (error << 16))
153  break;
154 
155  partial_corr = temp / (error << 1);
156 
157  lpc[i] = av_clipl_int32((int64_t) (partial_corr << 14) +
158  (1 << 15)) >> 16;
159 
160  /* Update the prediction error */
161  temp = MULL2(temp, partial_corr);
162  error = av_clipl_int32((int64_t) (error << 16) - temp +
163  (1 << 15)) >> 16;
164 
165  memcpy(vector, lpc, i * sizeof(int16_t));
166  for (j = 0; j < i; j++) {
167  temp = partial_corr * vector[i - j - 1] << 1;
168  lpc[j] = av_clipl_int32((int64_t) (lpc[j] << 16) - temp +
169  (1 << 15)) >> 16;
170  }
171  }
172 }
173 
174 /**
175  * Calculate LPC coefficients for the current frame.
176  *
177  * @param buf current frame
178  * @param prev_data 2 trailing subframes of the previous frame
179  * @param lpc LPC coefficients vector
180  */
181 static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
182 {
183  int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
184  int16_t *autocorr_ptr = autocorr;
185  int16_t *lpc_ptr = lpc;
186  int i, j;
187 
188  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
189  comp_autocorr(buf + i, autocorr_ptr);
190  levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
191 
192  lpc_ptr += LPC_ORDER;
193  autocorr_ptr += LPC_ORDER + 1;
194  }
195 }
196 
197 static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
198 {
199  int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
200  ///< polynomials (F1, F2) ordered as
201  ///< f1[0], f2[0], ...., f1[5], f2[5]
202 
203  int max, shift, cur_val, prev_val, count, p;
204  int i, j;
205  int64_t temp;
206 
207  /* Initialize f1[0] and f2[0] to 1 in Q25 */
208  for (i = 0; i < LPC_ORDER; i++)
209  lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
210 
211  /* Apply bandwidth expansion on the LPC coefficients */
212  f[0] = f[1] = 1 << 25;
213 
214  /* Compute the remaining coefficients */
215  for (i = 0; i < LPC_ORDER / 2; i++) {
216  /* f1 */
217  f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
218  /* f2 */
219  f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
220  }
221 
222  /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
223  f[LPC_ORDER] >>= 1;
224  f[LPC_ORDER + 1] >>= 1;
225 
226  /* Normalize and shorten */
227  max = FFABS(f[0]);
228  for (i = 1; i < LPC_ORDER + 2; i++)
229  max = FFMAX(max, FFABS(f[i]));
230 
231  shift = ff_g723_1_normalize_bits(max, 31);
232 
233  for (i = 0; i < LPC_ORDER + 2; i++)
234  f[i] = av_clipl_int32((int64_t) (f[i] << shift) + (1 << 15)) >> 16;
235 
236  /**
237  * Evaluate F1 and F2 at uniform intervals of pi/256 along the
238  * unit circle and check for zero crossings.
239  */
240  p = 0;
241  temp = 0;
242  for (i = 0; i <= LPC_ORDER / 2; i++)
243  temp += f[2 * i] * cos_tab[0];
244  prev_val = av_clipl_int32(temp << 1);
245  count = 0;
246  for (i = 1; i < COS_TBL_SIZE / 2; i++) {
247  /* Evaluate */
248  temp = 0;
249  for (j = 0; j <= LPC_ORDER / 2; j++)
250  temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
251  cur_val = av_clipl_int32(temp << 1);
252 
253  /* Check for sign change, indicating a zero crossing */
254  if ((cur_val ^ prev_val) < 0) {
255  int abs_cur = FFABS(cur_val);
256  int abs_prev = FFABS(prev_val);
257  int sum = abs_cur + abs_prev;
258 
259  shift = ff_g723_1_normalize_bits(sum, 31);
260  sum <<= shift;
261  abs_prev = abs_prev << shift >> 8;
262  lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
263 
264  if (count == LPC_ORDER)
265  break;
266 
267  /* Switch between sum and difference polynomials */
268  p ^= 1;
269 
270  /* Evaluate */
271  temp = 0;
272  for (j = 0; j <= LPC_ORDER / 2; j++)
273  temp += f[LPC_ORDER - 2 * j + p] *
274  cos_tab[i * j % COS_TBL_SIZE];
275  cur_val = av_clipl_int32(temp << 1);
276  }
277  prev_val = cur_val;
278  }
279 
280  if (count != LPC_ORDER)
281  memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
282 }
283 
284 /**
285  * Quantize the current LSP subvector.
286  *
287  * @param num band number
288  * @param offset offset of the current subvector in an LPC_ORDER vector
289  * @param size size of the current subvector
290  */
291 #define get_index(num, offset, size) \
292 { \
293  int error, max = -1; \
294  int16_t temp[4]; \
295  int i, j; \
296  \
297  for (i = 0; i < LSP_CB_SIZE; i++) { \
298  for (j = 0; j < size; j++){ \
299  temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] + \
300  (1 << 14)) >> 15; \
301  } \
302  error = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1; \
303  error -= ff_g723_1_dot_product(lsp_band##num[i], temp, size); \
304  if (error > max) { \
305  max = error; \
306  lsp_index[num] = i; \
307  } \
308  } \
309 }
310 
311 /**
312  * Vector quantize the LSP frequencies.
313  *
314  * @param lsp the current lsp vector
315  * @param prev_lsp the previous lsp vector
316  */
317 static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
318 {
319  int16_t weight[LPC_ORDER];
320  int16_t min, max;
321  int shift, i;
322 
323  /* Calculate the VQ weighting vector */
324  weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
325  weight[LPC_ORDER - 1] = (1 << 20) /
326  (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
327 
328  for (i = 1; i < LPC_ORDER - 1; i++) {
329  min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
330  if (min > 0x20)
331  weight[i] = (1 << 20) / min;
332  else
333  weight[i] = INT16_MAX;
334  }
335 
336  /* Normalize */
337  max = 0;
338  for (i = 0; i < LPC_ORDER; i++)
339  max = FFMAX(weight[i], max);
340 
341  shift = ff_g723_1_normalize_bits(max, 15);
342  for (i = 0; i < LPC_ORDER; i++) {
343  weight[i] <<= shift;
344  }
345 
346  /* Compute the VQ target vector */
347  for (i = 0; i < LPC_ORDER; i++) {
348  lsp[i] -= dc_lsp[i] +
349  (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
350  }
351 
352  get_index(0, 0, 3);
353  get_index(1, 3, 3);
354  get_index(2, 6, 4);
355 }
356 
357 /**
358  * Perform IIR filtering.
359  *
360  * @param fir_coef FIR coefficients
361  * @param iir_coef IIR coefficients
362  * @param src source vector
363  * @param dest destination vector
364  */
365 static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
366  int16_t *src, int16_t *dest)
367 {
368  int m, n;
369 
370  for (m = 0; m < SUBFRAME_LEN; m++) {
371  int64_t filter = 0;
372  for (n = 1; n <= LPC_ORDER; n++) {
373  filter -= fir_coef[n - 1] * src[m - n] -
374  iir_coef[n - 1] * dest[m - n];
375  }
376 
377  dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) +
378  (1 << 15)) >> 16;
379  }
380 }
381 
382 /**
383  * Apply the formant perceptual weighting filter.
384  *
385  * @param flt_coef filter coefficients
386  * @param unq_lpc unquantized lpc vector
387  */
388 static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
389  int16_t *unq_lpc, int16_t *buf)
390 {
391  int16_t vector[FRAME_LEN + LPC_ORDER];
392  int i, j, k, l = 0;
393 
394  memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
395  memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
396  memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
397 
398  for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
399  for (k = 0; k < LPC_ORDER; k++) {
400  flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
401  (1 << 14)) >> 15;
402  flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
403  percept_flt_tbl[1][k] +
404  (1 << 14)) >> 15;
405  }
406  iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER,
407  vector + i, buf + i);
408  l += LPC_ORDER;
409  }
410  memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
411  memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
412 }
413 
414 /**
415  * Estimate the open loop pitch period.
416  *
417  * @param buf perceptually weighted speech
418  * @param start estimation is carried out from this position
419  */
420 static int estimate_pitch(int16_t *buf, int start)
421 {
422  int max_exp = 32;
423  int max_ccr = 0x4000;
424  int max_eng = 0x7fff;
425  int index = PITCH_MIN;
426  int offset = start - PITCH_MIN + 1;
427 
428  int ccr, eng, orig_eng, ccr_eng, exp;
429  int diff, temp;
430 
431  int i;
432 
433  orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
434 
435  for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
436  offset--;
437 
438  /* Update energy and compute correlation */
439  orig_eng += buf[offset] * buf[offset] -
440  buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
441  ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
442  if (ccr <= 0)
443  continue;
444 
445  /* Split into mantissa and exponent to maintain precision */
446  exp = ff_g723_1_normalize_bits(ccr, 31);
447  ccr = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16;
448  exp <<= 1;
449  ccr *= ccr;
450  temp = ff_g723_1_normalize_bits(ccr, 31);
451  ccr = ccr << temp >> 16;
452  exp += temp;
453 
454  temp = ff_g723_1_normalize_bits(orig_eng, 31);
455  eng = av_clipl_int32((int64_t) (orig_eng << temp) + (1 << 15)) >> 16;
456  exp -= temp;
457 
458  if (ccr >= eng) {
459  exp--;
460  ccr >>= 1;
461  }
462  if (exp > max_exp)
463  continue;
464 
465  if (exp + 1 < max_exp)
466  goto update;
467 
468  /* Equalize exponents before comparison */
469  if (exp + 1 == max_exp)
470  temp = max_ccr >> 1;
471  else
472  temp = max_ccr;
473  ccr_eng = ccr * max_eng;
474  diff = ccr_eng - eng * temp;
475  if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
476 update:
477  index = i;
478  max_exp = exp;
479  max_ccr = ccr;
480  max_eng = eng;
481  }
482  }
483  return index;
484 }
485 
486 /**
487  * Compute harmonic noise filter parameters.
488  *
489  * @param buf perceptually weighted speech
490  * @param pitch_lag open loop pitch period
491  * @param hf harmonic filter parameters
492  */
493 static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
494 {
495  int ccr, eng, max_ccr, max_eng;
496  int exp, max, diff;
497  int energy[15];
498  int i, j;
499 
500  for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
501  /* Compute residual energy */
502  energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
503  /* Compute correlation */
504  energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
505  }
506 
507  /* Compute target energy */
508  energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
509 
510  /* Normalize */
511  max = 0;
512  for (i = 0; i < 15; i++)
513  max = FFMAX(max, FFABS(energy[i]));
514 
515  exp = ff_g723_1_normalize_bits(max, 31);
516  for (i = 0; i < 15; i++) {
517  energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
518  (1 << 15)) >> 16;
519  }
520 
521  hf->index = -1;
522  hf->gain = 0;
523  max_ccr = 1;
524  max_eng = 0x7fff;
525 
526  for (i = 0; i <= 6; i++) {
527  eng = energy[i << 1];
528  ccr = energy[(i << 1) + 1];
529 
530  if (ccr <= 0)
531  continue;
532 
533  ccr = (ccr * ccr + (1 << 14)) >> 15;
534  diff = ccr * max_eng - eng * max_ccr;
535  if (diff > 0) {
536  max_ccr = ccr;
537  max_eng = eng;
538  hf->index = i;
539  }
540  }
541 
542  if (hf->index == -1) {
543  hf->index = pitch_lag;
544  return;
545  }
546 
547  eng = energy[14] * max_eng;
548  eng = (eng >> 2) + (eng >> 3);
549  ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
550  if (eng < ccr) {
551  eng = energy[(hf->index << 1) + 1];
552 
553  if (eng >= max_eng)
554  hf->gain = 0x2800;
555  else
556  hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
557  }
558  hf->index += pitch_lag - 3;
559 }
560 
561 /**
562  * Apply the harmonic noise shaping filter.
563  *
564  * @param hf filter parameters
565  */
566 static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
567 {
568  int i;
569 
570  for (i = 0; i < SUBFRAME_LEN; i++) {
571  int64_t temp = hf->gain * src[i - hf->index] << 1;
572  dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
573  }
574 }
575 
576 static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
577 {
578  int i;
579  for (i = 0; i < SUBFRAME_LEN; i++) {
580  int64_t temp = hf->gain * src[i - hf->index] << 1;
581  dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
582  (1 << 15)) >> 16;
583  }
584 }
585 
586 /**
587  * Combined synthesis and formant perceptual weighting filer.
588  *
589  * @param qnt_lpc quantized lpc coefficients
590  * @param perf_lpc perceptual filter coefficients
591  * @param perf_fir perceptual filter fir memory
592  * @param perf_iir perceptual filter iir memory
593  * @param scale the filter output will be scaled by 2^scale
594  */
595 static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
596  int16_t *perf_fir, int16_t *perf_iir,
597  const int16_t *src, int16_t *dest, int scale)
598 {
599  int i, j;
600  int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
601  int64_t buf[SUBFRAME_LEN];
602 
603  int16_t *bptr_16 = buf_16 + LPC_ORDER;
604 
605  memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
606  memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
607 
608  for (i = 0; i < SUBFRAME_LEN; i++) {
609  int64_t temp = 0;
610  for (j = 1; j <= LPC_ORDER; j++)
611  temp -= qnt_lpc[j - 1] * bptr_16[i - j];
612 
613  buf[i] = (src[i] << 15) + (temp << 3);
614  bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
615  }
616 
617  for (i = 0; i < SUBFRAME_LEN; i++) {
618  int64_t fir = 0, iir = 0;
619  for (j = 1; j <= LPC_ORDER; j++) {
620  fir -= perf_lpc[j - 1] * bptr_16[i - j];
621  iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
622  }
623  dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
624  (1 << 15)) >> 16;
625  }
626  memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
627  memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
628  sizeof(int16_t) * LPC_ORDER);
629 }
630 
631 /**
632  * Compute the adaptive codebook contribution.
633  *
634  * @param buf input signal
635  * @param index the current subframe index
636  */
637 static void acb_search(G723_1_Context *p, int16_t *residual,
638  int16_t *impulse_resp, const int16_t *buf,
639  int index)
640 {
641  int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
642 
643  const int16_t *cb_tbl = adaptive_cb_gain85;
644 
645  int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
646 
647  int pitch_lag = p->pitch_lag[index >> 1];
648  int acb_lag = 1;
649  int acb_gain = 0;
650  int odd_frame = index & 1;
651  int iter = 3 + odd_frame;
652  int count = 0;
653  int tbl_size = 85;
654 
655  int i, j, k, l, max;
656  int64_t temp;
657 
658  if (!odd_frame) {
659  if (pitch_lag == PITCH_MIN)
660  pitch_lag++;
661  else
662  pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
663  }
664 
665  for (i = 0; i < iter; i++) {
666  ff_g723_1_get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
667 
668  for (j = 0; j < SUBFRAME_LEN; j++) {
669  temp = 0;
670  for (k = 0; k <= j; k++)
671  temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
672  flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
673  (1 << 15)) >> 16;
674  }
675 
676  for (j = PITCH_ORDER - 2; j >= 0; j--) {
677  flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
678  for (k = 1; k < SUBFRAME_LEN; k++) {
679  temp = (flt_buf[j + 1][k - 1] << 15) +
680  residual[j] * impulse_resp[k];
681  flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
682  }
683  }
684 
685  /* Compute crosscorrelation with the signal */
686  for (j = 0; j < PITCH_ORDER; j++) {
687  temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
688  ccr_buf[count++] = av_clipl_int32(temp << 1);
689  }
690 
691  /* Compute energies */
692  for (j = 0; j < PITCH_ORDER; j++) {
693  ccr_buf[count++] = ff_g723_1_dot_product(flt_buf[j], flt_buf[j],
694  SUBFRAME_LEN);
695  }
696 
697  for (j = 1; j < PITCH_ORDER; j++) {
698  for (k = 0; k < j; k++) {
699  temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
700  ccr_buf[count++] = av_clipl_int32(temp << 2);
701  }
702  }
703  }
704 
705  /* Normalize and shorten */
706  max = 0;
707  for (i = 0; i < 20 * iter; i++)
708  max = FFMAX(max, FFABS(ccr_buf[i]));
709 
710  temp = ff_g723_1_normalize_bits(max, 31);
711 
712  for (i = 0; i < 20 * iter; i++)
713  ccr_buf[i] = av_clipl_int32((int64_t) (ccr_buf[i] << temp) +
714  (1 << 15)) >> 16;
715 
716  max = 0;
717  for (i = 0; i < iter; i++) {
718  /* Select quantization table */
719  if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
720  odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
721  cb_tbl = adaptive_cb_gain170;
722  tbl_size = 170;
723  }
724 
725  for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
726  temp = 0;
727  for (l = 0; l < 20; l++)
728  temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
729  temp = av_clipl_int32(temp);
730 
731  if (temp > max) {
732  max = temp;
733  acb_gain = j;
734  acb_lag = i;
735  }
736  }
737  }
738 
739  if (!odd_frame) {
740  pitch_lag += acb_lag - 1;
741  acb_lag = 1;
742  }
743 
744  p->pitch_lag[index >> 1] = pitch_lag;
745  p->subframe[index].ad_cb_lag = acb_lag;
746  p->subframe[index].ad_cb_gain = acb_gain;
747 }
748 
749 /**
750  * Subtract the adaptive codebook contribution from the input
751  * to obtain the residual.
752  *
753  * @param buf target vector
754  */
755 static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
756  int16_t *buf)
757 {
758  int i, j;
759  /* Subtract adaptive CB contribution to obtain the residual */
760  for (i = 0; i < SUBFRAME_LEN; i++) {
761  int64_t temp = buf[i] << 14;
762  for (j = 0; j <= i; j++)
763  temp -= residual[j] * impulse_resp[i - j];
764 
765  buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
766  }
767 }
768 
769 /**
770  * Quantize the residual signal using the fixed codebook (MP-MLQ).
771  *
772  * @param optim optimized fixed codebook parameters
773  * @param buf excitation vector
774  */
775 static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
776  int16_t *buf, int pulse_cnt, int pitch_lag)
777 {
778  FCBParam param;
779  int16_t impulse_r[SUBFRAME_LEN];
780  int16_t temp_corr[SUBFRAME_LEN];
781  int16_t impulse_corr[SUBFRAME_LEN];
782 
783  int ccr1[SUBFRAME_LEN];
784  int ccr2[SUBFRAME_LEN];
785  int amp, err, max, max_amp_index, min, scale, i, j, k, l;
786 
787  int64_t temp;
788 
789  /* Update impulse response */
790  memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
791  param.dirac_train = 0;
792  if (pitch_lag < SUBFRAME_LEN - 2) {
793  param.dirac_train = 1;
794  ff_g723_1_gen_dirac_train(impulse_r, pitch_lag);
795  }
796 
797  for (i = 0; i < SUBFRAME_LEN; i++)
798  temp_corr[i] = impulse_r[i] >> 1;
799 
800  /* Compute impulse response autocorrelation */
801  temp = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
802 
803  scale = ff_g723_1_normalize_bits(temp, 31);
804  impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
805 
806  for (i = 1; i < SUBFRAME_LEN; i++) {
807  temp = ff_g723_1_dot_product(temp_corr + i, temp_corr,
808  SUBFRAME_LEN - i);
809  impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
810  }
811 
812  /* Compute crosscorrelation of impulse response with residual signal */
813  scale -= 4;
814  for (i = 0; i < SUBFRAME_LEN; i++) {
815  temp = ff_g723_1_dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
816  if (scale < 0)
817  ccr1[i] = temp >> -scale;
818  else
819  ccr1[i] = av_clipl_int32(temp << scale);
820  }
821 
822  /* Search loop */
823  for (i = 0; i < GRID_SIZE; i++) {
824  /* Maximize the crosscorrelation */
825  max = 0;
826  for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
827  temp = FFABS(ccr1[j]);
828  if (temp >= max) {
829  max = temp;
830  param.pulse_pos[0] = j;
831  }
832  }
833 
834  /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
835  amp = max;
836  min = 1 << 30;
837  max_amp_index = GAIN_LEVELS - 2;
838  for (j = max_amp_index; j >= 2; j--) {
839  temp = av_clipl_int32((int64_t) fixed_cb_gain[j] *
840  impulse_corr[0] << 1);
841  temp = FFABS(temp - amp);
842  if (temp < min) {
843  min = temp;
844  max_amp_index = j;
845  }
846  }
847 
848  max_amp_index--;
849  /* Select additional gain values */
850  for (j = 1; j < 5; j++) {
851  for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
852  temp_corr[k] = 0;
853  ccr2[k] = ccr1[k];
854  }
855  param.amp_index = max_amp_index + j - 2;
856  amp = fixed_cb_gain[param.amp_index];
857 
858  param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
859  temp_corr[param.pulse_pos[0]] = 1;
860 
861  for (k = 1; k < pulse_cnt; k++) {
862  max = INT_MIN;
863  for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
864  if (temp_corr[l])
865  continue;
866  temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
867  temp = av_clipl_int32((int64_t) temp *
868  param.pulse_sign[k - 1] << 1);
869  ccr2[l] -= temp;
870  temp = FFABS(ccr2[l]);
871  if (temp > max) {
872  max = temp;
873  param.pulse_pos[k] = l;
874  }
875  }
876 
877  param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
878  -amp : amp;
879  temp_corr[param.pulse_pos[k]] = 1;
880  }
881 
882  /* Create the error vector */
883  memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
884 
885  for (k = 0; k < pulse_cnt; k++)
886  temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
887 
888  for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
889  temp = 0;
890  for (l = 0; l <= k; l++) {
891  int prod = av_clipl_int32((int64_t) temp_corr[l] *
892  impulse_r[k - l] << 1);
893  temp = av_clipl_int32(temp + prod);
894  }
895  temp_corr[k] = temp << 2 >> 16;
896  }
897 
898  /* Compute square of error */
899  err = 0;
900  for (k = 0; k < SUBFRAME_LEN; k++) {
901  int64_t prod;
902  prod = av_clipl_int32((int64_t) buf[k] * temp_corr[k] << 1);
903  err = av_clipl_int32(err - prod);
904  prod = av_clipl_int32((int64_t) temp_corr[k] * temp_corr[k]);
905  err = av_clipl_int32(err + prod);
906  }
907 
908  /* Minimize */
909  if (err < optim->min_err) {
910  optim->min_err = err;
911  optim->grid_index = i;
912  optim->amp_index = param.amp_index;
913  optim->dirac_train = param.dirac_train;
914 
915  for (k = 0; k < pulse_cnt; k++) {
916  optim->pulse_sign[k] = param.pulse_sign[k];
917  optim->pulse_pos[k] = param.pulse_pos[k];
918  }
919  }
920  }
921  }
922 }
923 
924 /**
925  * Encode the pulse position and gain of the current subframe.
926  *
927  * @param optim optimized fixed CB parameters
928  * @param buf excitation vector
929  */
930 static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
931  int16_t *buf, int pulse_cnt)
932 {
933  int i, j;
934 
935  j = PULSE_MAX - pulse_cnt;
936 
937  subfrm->pulse_sign = 0;
938  subfrm->pulse_pos = 0;
939 
940  for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
941  int val = buf[optim->grid_index + (i << 1)];
942  if (!val) {
943  subfrm->pulse_pos += combinatorial_table[j][i];
944  } else {
945  subfrm->pulse_sign <<= 1;
946  if (val < 0)
947  subfrm->pulse_sign++;
948  j++;
949 
950  if (j == PULSE_MAX)
951  break;
952  }
953  }
954  subfrm->amp_index = optim->amp_index;
955  subfrm->grid_index = optim->grid_index;
956  subfrm->dirac_train = optim->dirac_train;
957 }
958 
959 /**
960  * Compute the fixed codebook excitation.
961  *
962  * @param buf target vector
963  * @param impulse_resp impulse response of the combined filter
964  */
965 static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
966  int16_t *buf, int index)
967 {
968  FCBParam optim;
969  int pulse_cnt = pulses[index];
970  int i;
971 
972  optim.min_err = 1 << 30;
973  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
974 
975  if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
976  get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
977  p->pitch_lag[index >> 1]);
978  }
979 
980  /* Reconstruct the excitation */
981  memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
982  for (i = 0; i < pulse_cnt; i++)
983  buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
984 
985  pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
986 
987  if (optim.dirac_train)
988  ff_g723_1_gen_dirac_train(buf, p->pitch_lag[index >> 1]);
989 }
990 
991 /**
992  * Pack the frame parameters into output bitstream.
993  *
994  * @param frame output buffer
995  * @param size size of the buffer
996  */
997 static int pack_bitstream(G723_1_Context *p, AVPacket *avpkt)
998 {
999  PutBitContext pb;
1000  int info_bits = 0;
1001  int i, temp;
1002 
1003  init_put_bits(&pb, avpkt->data, avpkt->size);
1004 
1005  put_bits(&pb, 2, info_bits);
1006 
1007  put_bits(&pb, 8, p->lsp_index[2]);
1008  put_bits(&pb, 8, p->lsp_index[1]);
1009  put_bits(&pb, 8, p->lsp_index[0]);
1010 
1011  put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
1012  put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
1013  put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
1014  put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
1015 
1016  /* Write 12 bit combined gain */
1017  for (i = 0; i < SUBFRAMES; i++) {
1018  temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
1019  p->subframe[i].amp_index;
1020  if (p->cur_rate == RATE_6300)
1021  temp += p->subframe[i].dirac_train << 11;
1022  put_bits(&pb, 12, temp);
1023  }
1024 
1025  put_bits(&pb, 1, p->subframe[0].grid_index);
1026  put_bits(&pb, 1, p->subframe[1].grid_index);
1027  put_bits(&pb, 1, p->subframe[2].grid_index);
1028  put_bits(&pb, 1, p->subframe[3].grid_index);
1029 
1030  if (p->cur_rate == RATE_6300) {
1031  skip_put_bits(&pb, 1); /* reserved bit */
1032 
1033  /* Write 13 bit combined position index */
1034  temp = (p->subframe[0].pulse_pos >> 16) * 810 +
1035  (p->subframe[1].pulse_pos >> 14) * 90 +
1036  (p->subframe[2].pulse_pos >> 16) * 9 +
1037  (p->subframe[3].pulse_pos >> 14);
1038  put_bits(&pb, 13, temp);
1039 
1040  put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
1041  put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
1042  put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
1043  put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
1044 
1045  put_bits(&pb, 6, p->subframe[0].pulse_sign);
1046  put_bits(&pb, 5, p->subframe[1].pulse_sign);
1047  put_bits(&pb, 6, p->subframe[2].pulse_sign);
1048  put_bits(&pb, 5, p->subframe[3].pulse_sign);
1049  }
1050 
1051  flush_put_bits(&pb);
1052  return frame_size[info_bits];
1053 }
1054 
1055 static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
1056  const AVFrame *frame, int *got_packet_ptr)
1057 {
1058  G723_1_Context *p = avctx->priv_data;
1059  int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
1060  int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
1061  int16_t cur_lsp[LPC_ORDER];
1062  int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
1063  int16_t vector[FRAME_LEN + PITCH_MAX];
1064  int offset, ret, i, j;
1065  int16_t *in, *start;
1066  HFParam hf[4];
1067 
1068  /* duplicate input */
1069  start = in = av_malloc(frame->nb_samples * sizeof(int16_t));
1070  if (!in)
1071  return AVERROR(ENOMEM);
1072  memcpy(in, frame->data[0], frame->nb_samples * sizeof(int16_t));
1073 
1074  highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
1075 
1076  memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
1077  memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
1078 
1079  comp_lpc_coeff(vector, unq_lpc);
1080  lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
1081  lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
1082 
1083  /* Update memory */
1084  memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
1085  sizeof(int16_t) * SUBFRAME_LEN);
1086  memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
1087  sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
1088  memcpy(p->prev_data, in + HALF_FRAME_LEN,
1089  sizeof(int16_t) * HALF_FRAME_LEN);
1090  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
1091 
1092  perceptual_filter(p, weighted_lpc, unq_lpc, vector);
1093 
1094  memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
1095  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
1096  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
1097 
1098  ff_g723_1_scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
1099 
1100  p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
1101  p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
1102 
1103  for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1104  comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
1105 
1106  memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
1107  memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
1108  memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
1109 
1110  for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
1111  harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
1112 
1113  ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
1114  ff_g723_1_lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
1115 
1116  memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
1117 
1118  offset = 0;
1119  for (i = 0; i < SUBFRAMES; i++) {
1120  int16_t impulse_resp[SUBFRAME_LEN];
1121  int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
1122  int16_t flt_in[SUBFRAME_LEN];
1123  int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
1124 
1125  /**
1126  * Compute the combined impulse response of the synthesis filter,
1127  * formant perceptual weighting filter and harmonic noise shaping filter
1128  */
1129  memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
1130  memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
1131  memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
1132 
1133  flt_in[0] = 1 << 13; /* Unit impulse */
1134  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
1135  zero, zero, flt_in, vector + PITCH_MAX, 1);
1136  harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
1137 
1138  /* Compute the combined zero input response */
1139  flt_in[0] = 0;
1140  memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
1141  memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
1142 
1143  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
1144  fir, iir, flt_in, vector + PITCH_MAX, 0);
1145  memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
1146  harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
1147 
1148  acb_search(p, residual, impulse_resp, in, i);
1150  p->pitch_lag[i >> 1], &p->subframe[i],
1151  p->cur_rate);
1152  sub_acb_contrib(residual, impulse_resp, in);
1153 
1154  fcb_search(p, impulse_resp, in, i);
1155 
1156  /* Reconstruct the excitation */
1158  p->pitch_lag[i >> 1], &p->subframe[i],
1159  RATE_6300);
1160 
1161  memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
1162  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
1163  for (j = 0; j < SUBFRAME_LEN; j++)
1164  in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
1165  memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
1166  sizeof(int16_t) * SUBFRAME_LEN);
1167 
1168  /* Update filter memories */
1169  synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
1170  p->perf_fir_mem, p->perf_iir_mem,
1171  in, vector + PITCH_MAX, 0);
1172  memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
1173  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
1174  memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
1175  sizeof(int16_t) * SUBFRAME_LEN);
1176 
1177  in += SUBFRAME_LEN;
1178  offset += LPC_ORDER;
1179  }
1180 
1181  av_free(start);
1182 
1183  if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0)
1184  return ret;
1185 
1186  *got_packet_ptr = 1;
1187  avpkt->size = pack_bitstream(p, avpkt);
1188  return 0;
1189 }
1190 
1192  .name = "g723_1",
1193  .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
1194  .type = AVMEDIA_TYPE_AUDIO,
1195  .id = AV_CODEC_ID_G723_1,
1196  .priv_data_size = sizeof(G723_1_Context),
1198  .encode2 = g723_1_encode_frame,
1199  .sample_fmts = (const enum AVSampleFormat[]) {
1201  },
1202 };
const char const char void * val
Definition: avisynth_c.h:634
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:930
#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:95
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:184
int ad_cb_gain
Definition: g723_1.h:82
int pitch_lag[2]
Definition: g723_1.h:127
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:206
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1714
int16_t prev_weight_sig[PITCH_MAX]
Definition: g723_1.h:153
memory handling functions
else temp
Definition: vf_mcdeint.c:259
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
G723.1 unpacked data subframe.
Definition: g723_1.h:80
int16_t fir_mem[LPC_ORDER]
Definition: g723_1.h:135
static float cos_tab[256]
Definition: dca_lbr.c:128
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition: g723_1.h:720
int size
Definition: avcodec.h:1581
static void skip_put_bits(PutBitContext *s, int n)
Skip the given number of bits.
Definition: put_bits.h:249
#define PITCH_ORDER
Definition: g723_1.h:45
int min_err
Definition: g723_1.h:111
int index
Definition: g723_1.h:103
AVCodec.
Definition: avcodec.h:3542
#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
int dirac_train
Definition: g723_1.h:114
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 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:755
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
Optimized fixed codebook excitation parameters.
Definition: g723_1.h:110
AVOptions.
#define LPC_ORDER
Definition: g723_1.h:40
static const int16_t adaptive_cb_gain85[85 *20]
Definition: g723_1.h:733
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, uint8_t clip)
Definition: cfhd.c:80
static AVFrame * frame
int pulse_sign
Definition: g723_1.h:84
uint8_t * data
Definition: avcodec.h:1580
static const int16_t percept_flt_tbl[2][LPC_ORDER]
0.5^i scaled by 2^15
Definition: g723_1.h:1428
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:137
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 GRID_SIZE
Definition: g723_1.h:46
#define av_log(a,...)
unsigned m
Definition: audioconvert.c:187
int16_t prev_data[HALF_FRAME_LEN]
Definition: g723_1.h:152
static const int16_t adaptive_cb_gain170[170 *20]
Definition: g723_1.h:949
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:627
#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:1191
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:1055
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:176
#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: avcodec.h:3549
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
int16_t prev_excitation[PITCH_MAX]
Definition: g723_1.h:132
static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
Apply the harmonic noise shaping filter.
Definition: g723_1enc.c:566
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
GLsizei count
Definition: opengl_enc.c:109
static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
Vector quantize the LSP frequencies.
Definition: g723_1enc.c:317
#define FFMAX(a, b)
Definition: common.h:94
int8_t exp
Definition: eval.c:64
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
G723_1_Subframe subframe[4]
Definition: g723_1.h:122
#define PITCH_MAX
Definition: g723_1.h:44
static const int16_t fixed_cb_gain[GAIN_LEVELS]
Definition: g723_1.h:727
static void fcb_search(G723_1_Context *p, int16_t *impulse_resp, int16_t *buf, int index)
Compute the fixed codebook excitation.
Definition: g723_1enc.c:965
static av_cold int g723_1_encode_init(AVCodecContext *avctx)
Definition: g723_1enc.c:43
enum Rate cur_rate
Definition: g723_1.h:125
static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
Definition: g723_1enc.c:576
static av_always_inline void update(SilenceDetectContext *s, AVFrame *insamples, int is_silence, int64_t nb_samples_notify, AVRational time_base)
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
static int pack_bitstream(G723_1_Context *p, AVPacket *avpkt)
Pack the frame parameters into output bitstream.
Definition: g723_1enc.c:997
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:365
static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
Calculate LPC coefficients for the current frame.
Definition: g723_1enc.c:181
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
int n
Definition: avisynth_c.h:547
#define src
Definition: vp9dsp.c:530
#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:2430
int frame_size
Definition: mxfenc.c:1821
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:229
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int sample_rate
samples per second
Definition: avcodec.h:2410
main external API structure.
Definition: avcodec.h:1649
static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
Remove DC component from the input signal.
Definition: g723_1enc.c:79
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
G.723.1 types, functions and data tables.
void * buf
Definition: avisynth_c.h:553
#define PULSE_MAX
Definition: dss_sp.c:32
static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef, int16_t *unq_lpc, int16_t *buf)
Apply the formant perceptual weighting filter.
Definition: g723_1enc.c:388
static const int16_t hamming_window[LPC_FRAME]
Hamming window coefficients scaled by 2^15.
Definition: g723_1.h:1390
int index
Definition: gxfenc.c:89
int16_t harmonic_mem[PITCH_MAX]
Definition: g723_1.h:160
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1690
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:595
int16_t hpf_fir_mem
highpass filter fir
Definition: g723_1.h:155
static int weight(int i, int blen, int offset)
Definition: diracdec.c:1429
#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:775
#define get_index(num, offset, size)
Quantize the current LSP subvector.
Definition: g723_1enc.c:291
int16_t prev_lsp[LPC_ORDER]
Definition: g723_1.h:130
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:198
int hpf_iir_mem
and iir memories
Definition: g723_1.h:156
static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
Definition: g723_1enc.c:197
#define SUBFRAMES
Definition: dcaenc.c:42
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
int16_t perf_iir_mem[LPC_ORDER]
and iir memories
Definition: g723_1.h:158
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:420
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
Harmonic filter parameters.
Definition: g723_1.h:102
void * priv_data
Definition: avcodec.h:1691
static av_always_inline int diff(const uint32_t a, const uint32_t b)
#define av_free(p)
#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:2411
static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
Compute harmonic noise filter parameters.
Definition: g723_1enc.c:493
uint8_t lsp_index[LSP_BANDS]
Definition: g723_1.h:126
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:1421
int iir_mem[LPC_ORDER]
Definition: g723_1.h:136
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
void INT64 start
Definition: avisynth_c.h:553
int16_t perf_fir_mem[LPC_ORDER]
perceptual filter fir
Definition: g723_1.h:157
static void acb_search(G723_1_Context *p, int16_t *residual, int16_t *impulse_resp, const int16_t *buf, int index)
Compute the adaptive codebook contribution.
Definition: g723_1enc.c:637
float min
This structure stores compressed data.
Definition: avcodec.h:1557
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
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:1414
int ad_cb_lag
adaptive codebook lag
Definition: g723_1.h:81
int gain
Definition: g723_1.h:104
bitstream writer API