FFmpeg  4.4.5
g729dec.c
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1 /*
2  * G.729, G729 Annex D decoders
3  * Copyright (c) 2008 Vladimir Voroshilov
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 #include <inttypes.h>
23 #include <string.h>
24 
25 #include "avcodec.h"
26 #include "libavutil/avutil.h"
27 #include "get_bits.h"
28 #include "audiodsp.h"
29 #include "internal.h"
30 
31 
32 #include "g729.h"
33 #include "lsp.h"
34 #include "celp_math.h"
35 #include "celp_filters.h"
36 #include "acelp_filters.h"
37 #include "acelp_pitch_delay.h"
38 #include "acelp_vectors.h"
39 #include "g729data.h"
40 #include "g729postfilter.h"
41 
42 /**
43  * minimum quantized LSF value (3.2.4)
44  * 0.005 in Q13
45  */
46 #define LSFQ_MIN 40
47 
48 /**
49  * maximum quantized LSF value (3.2.4)
50  * 3.135 in Q13
51  */
52 #define LSFQ_MAX 25681
53 
54 /**
55  * minimum LSF distance (3.2.4)
56  * 0.0391 in Q13
57  */
58 #define LSFQ_DIFF_MIN 321
59 
60 /// interpolation filter length
61 #define INTERPOL_LEN 11
62 
63 /**
64  * minimum gain pitch value (3.8, Equation 47)
65  * 0.2 in (1.14)
66  */
67 #define SHARP_MIN 3277
68 
69 /**
70  * maximum gain pitch value (3.8, Equation 47)
71  * (EE) This does not comply with the specification.
72  * Specification says about 0.8, which should be
73  * 13107 in (1.14), but reference C code uses
74  * 13017 (equals to 0.7945) instead of it.
75  */
76 #define SHARP_MAX 13017
77 
78 /**
79  * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13)
80  */
81 #define MR_ENERGY 1018156
82 
83 #define DECISION_NOISE 0
84 #define DECISION_INTERMEDIATE 1
85 #define DECISION_VOICE 2
86 
87 typedef enum {
91 } G729Formats;
92 
93 typedef struct {
94  uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
95  uint8_t parity_bit; ///< parity bit for pitch delay
96  uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits)
97  uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
98  uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
99  uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
102 
103 typedef struct {
104  /// past excitation signal buffer
106 
107  int16_t* exc; ///< start of past excitation data in buffer
108  int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
109 
110  /// (2.13) LSP quantizer outputs
111  int16_t past_quantizer_output_buf[MA_NP + 1][10];
112  int16_t* past_quantizer_outputs[MA_NP + 1];
113 
114  int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
115  int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
116  int16_t *lsp[2]; ///< pointers to lsp_buf
117 
118  int16_t quant_energy[4]; ///< (5.10) past quantized energy
119 
120  /// previous speech data for LP synthesis filter
121  int16_t syn_filter_data[10];
122 
123 
124  /// residual signal buffer (used in long-term postfilter)
125  int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
126 
127  /// previous speech data for residual calculation filter
128  int16_t res_filter_data[SUBFRAME_SIZE+10];
129 
130  /// previous speech data for short-term postfilter
131  int16_t pos_filter_data[SUBFRAME_SIZE+10];
132 
133  /// (1.14) pitch gain of current and five previous subframes
134  int16_t past_gain_pitch[6];
135 
136  /// (14.1) gain code from current and previous subframe
137  int16_t past_gain_code[2];
138 
139  /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
140  int16_t voice_decision;
141 
142  int16_t onset; ///< detected onset level (0-2)
143  int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
144  int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
145  int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
146  uint16_t rand_value; ///< random number generator value (4.4.4)
147  int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
148 
149  /// (14.14) high-pass filter data (past input)
150  int hpf_f[2];
151 
152  /// high-pass filter data (past output)
153  int16_t hpf_z[2];
155 
156 typedef struct {
158 
160 } G729Context;
161 
163  .ac_index_bits = {8,5},
164  .parity_bit = 1,
165  .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
166  .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
167  .fc_signs_bits = 4,
168  .fc_indexes_bits = 13,
169  .block_size = G729_8K_BLOCK_SIZE,
170 };
171 
173  .ac_index_bits = {8,4},
174  .parity_bit = 0,
175  .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
176  .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
177  .fc_signs_bits = 2,
178  .fc_indexes_bits = 9,
179  .block_size = G729D_6K4_BLOCK_SIZE,
180 };
181 
182 /**
183  * @brief pseudo random number generator
184  */
185 static inline uint16_t g729_prng(uint16_t value)
186 {
187  return 31821 * value + 13849;
188 }
189 
190 /**
191  * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
192  * @param[out] lsfq (2.13) quantized LSF coefficients
193  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
194  * @param ma_predictor switched MA predictor of LSP quantizer
195  * @param vq_1st first stage vector of quantizer
196  * @param vq_2nd_low second stage lower vector of LSP quantizer
197  * @param vq_2nd_high second stage higher vector of LSP quantizer
198  */
199 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
200  int16_t ma_predictor,
201  int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
202 {
203  int i,j;
204  static const uint8_t min_distance[2]={10, 5}; //(2.13)
205  int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
206 
207  for (i = 0; i < 5; i++) {
208  quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
209  quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
210  }
211 
212  for (j = 0; j < 2; j++) {
213  for (i = 1; i < 10; i++) {
214  int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
215  if (diff > 0) {
216  quantizer_output[i - 1] -= diff;
217  quantizer_output[i ] += diff;
218  }
219  }
220  }
221 
222  for (i = 0; i < 10; i++) {
223  int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
224  for (j = 0; j < MA_NP; j++)
225  sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
226 
227  lsfq[i] = sum >> 15;
228  }
229 
231 }
232 
233 /**
234  * Restores past LSP quantizer output using LSF from previous frame
235  * @param[in,out] lsfq (2.13) quantized LSF coefficients
236  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
237  * @param ma_predictor_prev MA predictor from previous frame
238  * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
239  */
240 static void lsf_restore_from_previous(int16_t* lsfq,
241  int16_t* past_quantizer_outputs[MA_NP + 1],
242  int ma_predictor_prev)
243 {
244  int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
245  int i,k;
246 
247  for (i = 0; i < 10; i++) {
248  int tmp = lsfq[i] << 15;
249 
250  for (k = 0; k < MA_NP; k++)
251  tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
252 
253  quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
254  }
255 }
256 
257 /**
258  * Constructs new excitation signal and applies phase filter to it
259  * @param[out] out constructed speech signal
260  * @param in original excitation signal
261  * @param fc_cur (2.13) original fixed-codebook vector
262  * @param gain_code (14.1) gain code
263  * @param subframe_size length of the subframe
264  */
265 static void g729d_get_new_exc(
266  int16_t* out,
267  const int16_t* in,
268  const int16_t* fc_cur,
269  int dstate,
270  int gain_code,
271  int subframe_size)
272 {
273  int i;
274  int16_t fc_new[SUBFRAME_SIZE];
275 
276  ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
277 
278  for (i = 0; i < subframe_size; i++) {
279  out[i] = in[i];
280  out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
281  out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
282  }
283 }
284 
285 /**
286  * Makes decision about onset in current subframe
287  * @param past_onset decision result of previous subframe
288  * @param past_gain_code gain code of current and previous subframe
289  *
290  * @return onset decision result for current subframe
291  */
292 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
293 {
294  if ((past_gain_code[0] >> 1) > past_gain_code[1])
295  return 2;
296 
297  return FFMAX(past_onset-1, 0);
298 }
299 
300 /**
301  * Makes decision about voice presence in current subframe
302  * @param onset onset level
303  * @param prev_voice_decision voice decision result from previous subframe
304  * @param past_gain_pitch pitch gain of current and previous subframes
305  *
306  * @return voice decision result for current subframe
307  */
308 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
309 {
310  int i, low_gain_pitch_cnt, voice_decision;
311 
312  if (past_gain_pitch[0] >= 14745) { // 0.9
313  voice_decision = DECISION_VOICE;
314  } else if (past_gain_pitch[0] <= 9830) { // 0.6
315  voice_decision = DECISION_NOISE;
316  } else {
317  voice_decision = DECISION_INTERMEDIATE;
318  }
319 
320  for (i = 0, low_gain_pitch_cnt = 0; i < 6; i++)
321  if (past_gain_pitch[i] < 9830)
322  low_gain_pitch_cnt++;
323 
324  if (low_gain_pitch_cnt > 2 && !onset)
325  voice_decision = DECISION_NOISE;
326 
327  if (!onset && voice_decision > prev_voice_decision + 1)
328  voice_decision--;
329 
330  if (onset && voice_decision < DECISION_VOICE)
331  voice_decision++;
332 
333  return voice_decision;
334 }
335 
336 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
337 {
338  int64_t res = 0;
339 
340  while (order--)
341  res += *v1++ * *v2++;
342 
343  if (res > INT32_MAX) return INT32_MAX;
344  else if (res < INT32_MIN) return INT32_MIN;
345 
346  return res;
347 }
348 
350 {
351  G729Context *s = avctx->priv_data;
353  int c,i,k;
354 
355  if (avctx->channels < 1 || avctx->channels > 2) {
356  av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
357  return AVERROR(EINVAL);
358  }
360 
361  /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
362  avctx->frame_size = SUBFRAME_SIZE << 1;
363 
364  ctx =
365  s->channel_context = av_mallocz(sizeof(G729ChannelContext) * avctx->channels);
366  if (!ctx)
367  return AVERROR(ENOMEM);
368 
369  for (c = 0; c < avctx->channels; c++) {
370  ctx->gain_coeff = 16384; // 1.0 in (1.14)
371 
372  for (k = 0; k < MA_NP + 1; k++) {
373  ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
374  for (i = 1; i < 11; i++)
375  ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
376  }
377 
378  ctx->lsp[0] = ctx->lsp_buf[0];
379  ctx->lsp[1] = ctx->lsp_buf[1];
380  memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
381 
382  ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
383 
384  ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
385 
386  /* random seed initialization */
387  ctx->rand_value = 21845;
388 
389  /* quantized prediction error */
390  for (i = 0; i < 4; i++)
391  ctx->quant_energy[i] = -14336; // -14 in (5.10)
392 
393  ctx++;
394  }
395 
396  ff_audiodsp_init(&s->adsp);
397  s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
398 
399  return 0;
400 }
401 
402 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
403  AVPacket *avpkt)
404 {
405  const uint8_t *buf = avpkt->data;
406  int buf_size = avpkt->size;
407  int16_t *out_frame;
408  GetBitContext gb;
410  int c, i;
411  int16_t *tmp;
412  G729Formats packet_type;
413  G729Context *s = avctx->priv_data;
414  G729ChannelContext *ctx = s->channel_context;
415  int16_t lp[2][11]; // (3.12)
416  uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
417  uint8_t quantizer_1st; ///< first stage vector of quantizer
418  uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
419  uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
420 
421  int pitch_delay_int[2]; // pitch delay, integer part
422  int pitch_delay_3x; // pitch delay, multiplied by 3
423  int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
424  int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
425  int j, ret;
426  int gain_before, gain_after;
427  AVFrame *frame = data;
428 
430  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
431  return ret;
432 
433  if (buf_size && buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels) == 0) {
434  packet_type = FORMAT_G729_8K;
436  //Reset voice decision
437  ctx->onset = 0;
438  ctx->voice_decision = DECISION_VOICE;
439  av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
440  } else if (buf_size == G729D_6K4_BLOCK_SIZE * avctx->channels && avctx->codec_id != AV_CODEC_ID_ACELP_KELVIN) {
441  packet_type = FORMAT_G729D_6K4;
443  av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
444  } else {
445  av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
446  return AVERROR_INVALIDDATA;
447  }
448 
449  for (c = 0; c < avctx->channels; c++) {
450  int frame_erasure = 0; ///< frame erasure detected during decoding
451  int bad_pitch = 0; ///< parity check failed
452  int is_periodic = 0; ///< whether one of the subframes is declared as periodic or not
453  out_frame = (int16_t*)frame->data[c];
454  if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
455  if (*buf != ((avctx->channels - 1 - c) * 0x80 | 2))
456  avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
457  buf++;
458  }
459 
460  for (i = 0; i < format->block_size; i++)
461  frame_erasure |= buf[i];
463 
464  init_get_bits8(&gb, buf, format->block_size);
465 
466  ma_predictor = get_bits(&gb, 1);
467  quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
468  quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
469  quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
470 
471  if (frame_erasure) {
472  lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
473  ctx->ma_predictor_prev);
474  } else {
475  lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
476  ma_predictor,
477  quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
478  ctx->ma_predictor_prev = ma_predictor;
479  }
480 
481  tmp = ctx->past_quantizer_outputs[MA_NP];
482  memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
483  MA_NP * sizeof(int16_t*));
484  ctx->past_quantizer_outputs[0] = tmp;
485 
486  ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
487 
488  ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
489 
490  FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
491 
492  for (i = 0; i < 2; i++) {
493  int gain_corr_factor;
494 
495  uint8_t ac_index; ///< adaptive codebook index
496  uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
497  int fc_indexes; ///< fixed-codebook indexes
498  uint8_t gc_1st_index; ///< gain codebook (first stage) index
499  uint8_t gc_2nd_index; ///< gain codebook (second stage) index
500 
501  ac_index = get_bits(&gb, format->ac_index_bits[i]);
502  if (!i && format->parity_bit)
503  bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
504  fc_indexes = get_bits(&gb, format->fc_indexes_bits);
505  pulses_signs = get_bits(&gb, format->fc_signs_bits);
506  gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
507  gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
508 
509  if (frame_erasure) {
510  pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
511  } else if (!i) {
512  if (bad_pitch) {
513  pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
514  } else {
515  pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
516  }
517  } else {
518  int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
520 
521  if (packet_type == FORMAT_G729D_6K4) {
522  pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
523  } else {
524  pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
525  }
526  }
527 
528  /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
529  pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
530  if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
531  av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
532  pitch_delay_int[i] = PITCH_DELAY_MAX;
533  }
534 
535  if (frame_erasure) {
536  ctx->rand_value = g729_prng(ctx->rand_value);
537  fc_indexes = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
538 
539  ctx->rand_value = g729_prng(ctx->rand_value);
540  pulses_signs = ctx->rand_value;
541  }
542 
543 
544  memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
545  switch (packet_type) {
546  case FORMAT_G729_8K:
549  fc_indexes, pulses_signs, 3, 3);
550  break;
551  case FORMAT_G729D_6K4:
554  fc_indexes, pulses_signs, 1, 4);
555  break;
556  }
557 
558  /*
559  This filter enhances harmonic components of the fixed-codebook vector to
560  improve the quality of the reconstructed speech.
561 
562  / fc_v[i], i < pitch_delay
563  fc_v[i] = <
564  \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
565  */
566  if (SUBFRAME_SIZE > pitch_delay_int[i])
567  ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
568  fc + pitch_delay_int[i],
569  fc, 1 << 14,
570  av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
571  0, 14,
572  SUBFRAME_SIZE - pitch_delay_int[i]);
573 
574  memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
575  ctx->past_gain_code[1] = ctx->past_gain_code[0];
576 
577  if (frame_erasure) {
578  ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
579  ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
580 
581  gain_corr_factor = 0;
582  } else {
583  if (packet_type == FORMAT_G729D_6K4) {
584  ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
585  cb_gain_2nd_6k4[gc_2nd_index][0];
586  gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
587  cb_gain_2nd_6k4[gc_2nd_index][1];
588 
589  /* Without check below overflow can occur in ff_acelp_update_past_gain.
590  It is not issue for G.729, because gain_corr_factor in it's case is always
591  greater than 1024, while in G.729D it can be even zero. */
592  gain_corr_factor = FFMAX(gain_corr_factor, 1024);
593  #ifndef G729_BITEXACT
594  gain_corr_factor >>= 1;
595  #endif
596  } else {
597  ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
598  cb_gain_2nd_8k[gc_2nd_index][0];
599  gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
600  cb_gain_2nd_8k[gc_2nd_index][1];
601  }
602 
603  /* Decode the fixed-codebook gain. */
604  ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
605  fc, MR_ENERGY,
606  ctx->quant_energy,
608  SUBFRAME_SIZE, 4);
609  #ifdef G729_BITEXACT
610  /*
611  This correction required to get bit-exact result with
612  reference code, because gain_corr_factor in G.729D is
613  two times larger than in original G.729.
614 
615  If bit-exact result is not issue then gain_corr_factor
616  can be simpler divided by 2 before call to g729_get_gain_code
617  instead of using correction below.
618  */
619  if (packet_type == FORMAT_G729D_6K4) {
620  gain_corr_factor >>= 1;
621  ctx->past_gain_code[0] >>= 1;
622  }
623  #endif
624  }
625  ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
626 
627  /* Routine requires rounding to lowest. */
629  ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
631  (pitch_delay_3x % 3) << 1,
632  10, SUBFRAME_SIZE);
633 
635  ctx->exc + i * SUBFRAME_SIZE, fc,
636  (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
637  ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
638  1 << 13, 14, SUBFRAME_SIZE);
639 
640  memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
641 
643  synth+10,
644  &lp[i][1],
645  ctx->exc + i * SUBFRAME_SIZE,
647  10,
648  1,
649  0,
650  0x800))
651  /* Overflow occurred, downscale excitation signal... */
652  for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
653  ctx->exc_base[j] >>= 2;
654 
655  /* ... and make synthesis again. */
656  if (packet_type == FORMAT_G729D_6K4) {
657  int16_t exc_new[SUBFRAME_SIZE];
658 
659  ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
660  ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
661 
662  g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
663 
665  synth+10,
666  &lp[i][1],
667  exc_new,
669  10,
670  0,
671  0,
672  0x800);
673  } else {
675  synth+10,
676  &lp[i][1],
677  ctx->exc + i * SUBFRAME_SIZE,
679  10,
680  0,
681  0,
682  0x800);
683  }
684  /* Save data (without postfilter) for use in next subframe. */
685  memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
686 
687  /* Calculate gain of unfiltered signal for use in AGC. */
688  gain_before = 0;
689  for (j = 0; j < SUBFRAME_SIZE; j++)
690  gain_before += FFABS(synth[j+10]);
691 
692  /* Call postfilter and also update voicing decision for use in next frame. */
694  &s->adsp,
695  &ctx->ht_prev_data,
696  &is_periodic,
697  &lp[i][0],
698  pitch_delay_int[0],
699  ctx->residual,
700  ctx->res_filter_data,
701  ctx->pos_filter_data,
702  synth+10,
703  SUBFRAME_SIZE);
704 
705  /* Calculate gain of filtered signal for use in AGC. */
706  gain_after = 0;
707  for (j = 0; j < SUBFRAME_SIZE; j++)
708  gain_after += FFABS(synth[j+10]);
709 
710  ctx->gain_coeff = ff_g729_adaptive_gain_control(
711  gain_before,
712  gain_after,
713  synth+10,
715  ctx->gain_coeff);
716 
717  if (frame_erasure) {
718  ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
719  } else {
720  ctx->pitch_delay_int_prev = pitch_delay_int[i];
721  }
722 
723  memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
725  out_frame + i*SUBFRAME_SIZE,
726  ctx->hpf_f,
727  synth+10,
728  SUBFRAME_SIZE);
729  memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
730  }
731 
732  ctx->was_periodic = is_periodic;
733 
734  /* Save signal for use in next frame. */
735  memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
736 
737  buf += format->block_size;
738  ctx++;
739  }
740 
741  *got_frame_ptr = 1;
742  return (format->block_size + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels;
743 }
744 
746 {
747  G729Context *s = avctx->priv_data;
748  av_freep(&s->channel_context);
749 
750  return 0;
751 }
752 
754  .name = "g729",
755  .long_name = NULL_IF_CONFIG_SMALL("G.729"),
756  .type = AVMEDIA_TYPE_AUDIO,
757  .id = AV_CODEC_ID_G729,
758  .priv_data_size = sizeof(G729Context),
759  .init = decoder_init,
760  .decode = decode_frame,
761  .close = decode_close,
762  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
763 };
764 
766  .name = "acelp.kelvin",
767  .long_name = NULL_IF_CONFIG_SMALL("Sipro ACELP.KELVIN"),
768  .type = AVMEDIA_TYPE_AUDIO,
770  .priv_data_size = sizeof(G729Context),
771  .init = decoder_init,
772  .decode = decode_frame,
773  .close = decode_close,
774  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
775 };
const int16_t ff_acelp_interp_filter[61]
low-pass Finite Impulse Response filter coefficients.
Definition: acelp_filters.c:30
void ff_acelp_high_pass_filter(int16_t *out, int hpf_f[2], const int16_t *in, int length)
high-pass filtering and upscaling (4.2.5 of G.729).
Definition: acelp_filters.c:99
void ff_acelp_interpolate(int16_t *out, const int16_t *in, const int16_t *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Generic FIR interpolation routine.
Definition: acelp_filters.c:44
void ff_acelp_update_past_gain(int16_t *quant_energy, int gain_corr_factor, int log2_ma_pred_order, int erasure)
Update past quantized energies.
int16_t ff_acelp_decode_gain_code(AudioDSPContext *adsp, int gain_corr_factor, const int16_t *fc_v, int mr_energy, const int16_t *quant_energy, const int16_t *ma_prediction_coeff, int subframe_size, int ma_pred_order)
Decode the adaptive codebook gain and add correction (4.1.5 and 3.9.1 of G.729).
static int ff_acelp_decode_4bit_to_2nd_delay3(int ac_index, int pitch_delay_min)
Decode pitch delay with 1/3 precision.
#define PITCH_DELAY_MIN
#define PITCH_DELAY_MAX
static int ff_acelp_decode_8bit_to_1st_delay3(int ac_index)
Decode pitch delay of the first subframe encoded by 8 bits with 1/3 resolution.
static int ff_acelp_decode_5_6_bit_to_2nd_delay3(int ac_index, int pitch_delay_min)
Decode pitch delay of the second subframe encoded by 5 or 6 bits with 1/3 precision.
const uint8_t ff_fc_2pulses_9bits_track1_gray[16]
Definition: acelp_vectors.c:31
const uint8_t ff_fc_4pulses_8bits_tracks_13[16]
Definition: acelp_vectors.c:63
const uint8_t ff_fc_2pulses_9bits_track2_gray[32]
Definition: acelp_vectors.c:43
void ff_acelp_fc_pulse_per_track(int16_t *fc_v, const uint8_t *tab1, const uint8_t *tab2, int pulse_indexes, int pulse_signs, int pulse_count, int bits)
Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
const uint8_t ff_fc_4pulses_8bits_track_4[32]
Definition: acelp_vectors.c:68
void ff_acelp_weighted_vector_sum(int16_t *out, const int16_t *in_a, const int16_t *in_b, int16_t weight_coeff_a, int16_t weight_coeff_b, int16_t rounder, int shift, int length)
weighted sum of two vectors with rounding.
static const char *const format[]
Definition: af_aiir.c:456
#define av_cold
Definition: attributes.h:88
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)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
int32_t
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
Convenience header that includes libavutil's core.
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:562
#define s(width, name)
Definition: cbs_vp9.c:257
void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in, const int16_t *filter, int len)
Circularly convolve fixed vector with a phase dispersion impulse response filter (D....
Definition: celp_filters.c:30
int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs, const int16_t *in, int buffer_length, int filter_length, int stop_on_overflow, int shift, int rounder)
LP synthesis filter.
Definition: celp_filters.c:60
#define FFSWAP(type, a, b)
Definition: common.h:108
#define FFMIN(a, b)
Definition: common.h:105
#define av_mod_uintp2
Definition: common.h:149
#define av_parity
Definition: common.h:182
#define av_clip
Definition: common.h:122
#define FFMAX(a, b)
Definition: common.h:103
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
long long int64_t
Definition: coverity.c:34
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static AVFrame * frame
double value
Definition: eval.c:98
static void frame_erasure(EVRCContext *e, float *samples)
Definition: evrcdec.c:652
#define SUBFRAME_SIZE
Definition: evrcdec.c:41
#define G729_8K_BLOCK_SIZE
Definition: g729.h:30
#define G729D_6K4_BLOCK_SIZE
Definition: g729.h:31
static const int16_t cb_gain_2nd_8k[1<< GC_2ND_IDX_BITS_8K][2]
gain codebook (second stage), 8k mode (3.9.2 of G.729)
Definition: g729data.h:229
static const int16_t lsp_init[10]
initial LSP coefficients belongs to virtual frame preceding the first frame of the stream
Definition: g729data.h:351
#define GC_1ST_IDX_BITS_6K4
gain codebook (first stage) index, 6.4k mode (size in bits)
Definition: g729data.h:35
static const int16_t cb_lsp_1st[1<< VQ_1ST_BITS][10]
first stage LSP codebook (10-dimensional, with 128 entries (3.24 of G.729)
Definition: g729data.h:42
static const int16_t cb_gain_1st_6k4[1<< GC_1ST_IDX_BITS_6K4][2]
gain codebook (first stage), 6.4k mode (D.3.9.2 of G.729)
Definition: g729data.h:251
#define GC_1ST_IDX_BITS_8K
gain codebook (first stage) index, 8k mode (size in bits)
Definition: g729data.h:32
static const int16_t cb_ma_predictor[2][MA_NP][10]
4th order Moving Average (MA) Predictor codebook (3.2.4 of G.729)
Definition: g729data.h:300
#define GC_2ND_IDX_BITS_6K4
gain codebook (second stage) index, 6.4k mode (size in bits)
Definition: g729data.h:36
#define VQ_1ST_BITS
first stage vector of quantizer (size in bits)
Definition: g729data.h:29
static const int16_t cb_lsp_2nd[1<< VQ_2ND_BITS][10]
second stage LSP codebook, high and low parts (both 5-dimensional, with 32 entries (3....
Definition: g729data.h:177
static const int16_t cb_ma_predictor_sum_inv[2][10]
Definition: g729data.h:335
static const int16_t cb_ma_predictor_sum[2][10]
Definition: g729data.h:321
static const uint16_t ma_prediction_coeff[4]
MA prediction coefficients (3.9.1 of G.729, near Equation 69)
Definition: g729data.h:343
static const int16_t cb_gain_2nd_6k4[1<< GC_2ND_IDX_BITS_6K4][2]
gain codebook (second stage), 6.4k mode (D.3.9.2 of G.729)
Definition: g729data.h:266
#define MA_NP
Moving Average (MA) prediction order.
Definition: g729data.h:27
static const int16_t cb_gain_1st_8k[1<< GC_1ST_IDX_BITS_8K][2]
gain codebook (first stage), 8k mode (3.9.2 of G.729)
Definition: g729data.h:215
#define VQ_2ND_BITS
second stage vector of quantizer (size in bits)
Definition: g729data.h:30
#define GC_2ND_IDX_BITS_8K
gain codebook (second stage) index, 8k mode (size in bits)
Definition: g729data.h:33
static const int16_t phase_filter[3][40]
additional "phase" post-processing filter impulse response (D.6.2 of G.729)
Definition: g729data.h:361
int16_t ff_g729_adaptive_gain_control(int gain_before, int gain_after, int16_t *speech, int subframe_size, int16_t gain_prev)
Adaptive gain control (4.2.4)
void ff_g729_postfilter(AudioDSPContext *adsp, int16_t *ht_prev_data, int *voicing, const int16_t *lp_filter_coeffs, int pitch_delay_int, int16_t *residual, int16_t *res_filter_data, int16_t *pos_filter_data, int16_t *speech, int subframe_size)
Signal postfiltering (4.2)
#define RES_PREV_DATA_SIZE
Amount of past residual signal data stored in buffer.
bitstream reader API header.
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time,...
Definition: codec.h:95
@ AV_CODEC_ID_G729
Definition: codec_id.h:477
@ AV_CODEC_ID_ACELP_KELVIN
Definition: codec_id.h:515
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
int i
Definition: input.c:407
av_cold void ff_audiodsp_init(AudioDSPContext *c)
Definition: audiodsp.c:106
static void g729d_get_new_exc(int16_t *out, const int16_t *in, const int16_t *fc_cur, int dstate, int gain_code, int subframe_size)
Constructs new excitation signal and applies phase filter to it.
Definition: g729dec.c:265
static uint16_t g729_prng(uint16_t value)
pseudo random number generator
Definition: g729dec.c:185
static int32_t scalarproduct_int16_c(const int16_t *v1, const int16_t *v2, int order)
Definition: g729dec.c:336
G729Formats
Definition: g729dec.c:87
@ FORMAT_G729D_6K4
Definition: g729dec.c:89
@ FORMAT_COUNT
Definition: g729dec.c:90
@ FORMAT_G729_8K
Definition: g729dec.c:88
static const G729FormatDescription format_g729_8k
Definition: g729dec.c:162
static av_cold int decode_close(AVCodecContext *avctx)
Definition: g729dec.c:745
static int g729d_onset_decision(int past_onset, const int16_t *past_gain_code)
Makes decision about onset in current subframe.
Definition: g729dec.c:292
#define SHARP_MIN
minimum gain pitch value (3.8, Equation 47) 0.2 in (1.14)
Definition: g729dec.c:67
#define DECISION_INTERMEDIATE
Definition: g729dec.c:84
static void lsf_restore_from_previous(int16_t *lsfq, int16_t *past_quantizer_outputs[MA_NP+1], int ma_predictor_prev)
Restores past LSP quantizer output using LSF from previous frame.
Definition: g729dec.c:240
#define DECISION_VOICE
Definition: g729dec.c:85
#define MR_ENERGY
MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7....
Definition: g729dec.c:81
#define DECISION_NOISE
Definition: g729dec.c:83
AVCodec ff_acelp_kelvin_decoder
Definition: g729dec.c:765
static const G729FormatDescription format_g729d_6k4
Definition: g729dec.c:172
#define SHARP_MAX
maximum gain pitch value (3.8, Equation 47) (EE) This does not comply with the specification.
Definition: g729dec.c:76
static void lsf_decode(int16_t *lsfq, int16_t *past_quantizer_outputs[MA_NP+1], int16_t ma_predictor, int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
Definition: g729dec.c:199
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: g729dec.c:402
static av_cold int decoder_init(AVCodecContext *avctx)
Definition: g729dec.c:349
#define LSFQ_MIN
minimum quantized LSF value (3.2.4) 0.005 in Q13
Definition: g729dec.c:46
#define INTERPOL_LEN
interpolation filter length
Definition: g729dec.c:61
#define LSFQ_DIFF_MIN
minimum LSF distance (3.2.4) 0.0391 in Q13
Definition: g729dec.c:58
static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t *past_gain_pitch)
Makes decision about voice presence in current subframe.
Definition: g729dec.c:308
#define LSFQ_MAX
maximum quantized LSF value (3.2.4) 3.135 in Q13
Definition: g729dec.c:52
AVCodec ff_g729_decoder
Definition: g729dec.c:753
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
void ff_acelp_lp_decode(int16_t *lp_1st, int16_t *lp_2nd, const int16_t *lsp_2nd, const int16_t *lsp_prev, int lp_order)
Interpolate LSP for the first subframe and convert LSP -> LP for both subframes (3....
Definition: lsp.c:171
void ff_acelp_lsf2lsp(int16_t *lsp, const int16_t *lsf, int lp_order)
Convert LSF to LSP.
Definition: lsp.c:83
void ff_acelp_reorder_lsf(int16_t *lsfq, int lsfq_min_distance, int lsfq_min, int lsfq_max, int lp_order)
(I.F) means fixed-point value with F fractional and I integer bits
Definition: lsp.c:33
const char data[16]
Definition: mxf.c:142
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int channels
number of audio channels
Definition: avcodec.h:1197
enum AVCodecID codec_id
Definition: avcodec.h:546
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
int16_t was_periodic
whether previous frame was declared as periodic or not (4.4)
Definition: g729dec.c:143
uint16_t rand_value
random number generator value (4.4.4)
Definition: g729dec.c:146
int16_t ht_prev_data
previous data for 4.2.3, equation 86
Definition: g729dec.c:144
int gain_coeff
(1.14) gain coefficient (4.2.4)
Definition: g729dec.c:145
int16_t voice_decision
voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
Definition: g729dec.c:140
int16_t * exc
start of past excitation data in buffer
Definition: g729dec.c:107
int pitch_delay_int_prev
integer part of previous subframe's pitch delay (4.1.3)
Definition: g729dec.c:108
int16_t onset
detected onset level (0-2)
Definition: g729dec.c:142
int ma_predictor_prev
switched MA predictor of LSP quantizer from last good frame
Definition: g729dec.c:147
G729ChannelContext * channel_context
Definition: g729dec.c:159
AudioDSPContext adsp
Definition: g729dec.c:157
uint8_t parity_bit
parity bit for pitch delay
Definition: g729dec.c:95
uint8_t gc_2nd_index_bits
gain codebook (second stage) index (size in bits)
Definition: g729dec.c:97
uint8_t ac_index_bits[2]
adaptive codebook index for second subframe (size in bits)
Definition: g729dec.c:94
uint8_t fc_indexes_bits
size (in bits) of fixed-codebook index entry
Definition: g729dec.c:99
uint8_t fc_signs_bits
number of pulses in fixed-codebook vector
Definition: g729dec.c:98
uint8_t gc_1st_index_bits
gain codebook (first stage) index (size in bits)
Definition: g729dec.c:96
#define avpriv_request_sample(...)
#define av_freep(p)
#define av_log(a,...)
static uint8_t tmp[11]
Definition: aes_ctr.c:27
FILE * out
Definition: movenc.c:54
AVFormatContext * ctx
Definition: movenc.c:48
if(ret< 0)
Definition: vf_mcdeint.c:282
static av_always_inline int diff(const uint32_t a, const uint32_t b)
static double c[64]