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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
101 
102 typedef struct {
104 
105  /// past excitation signal buffer
107 
108  int16_t* exc; ///< start of past excitation data in buffer
109  int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
110 
111  /// (2.13) LSP quantizer outputs
112  int16_t past_quantizer_output_buf[MA_NP + 1][10];
113  int16_t* past_quantizer_outputs[MA_NP + 1];
114 
115  int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
116  int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
117  int16_t *lsp[2]; ///< pointers to lsp_buf
118 
119  int16_t quant_energy[4]; ///< (5.10) past quantized energy
120 
121  /// previous speech data for LP synthesis filter
122  int16_t syn_filter_data[10];
123 
124 
125  /// residual signal buffer (used in long-term postfilter)
127 
128  /// previous speech data for residual calculation filter
129  int16_t res_filter_data[SUBFRAME_SIZE+10];
130 
131  /// previous speech data for short-term postfilter
132  int16_t pos_filter_data[SUBFRAME_SIZE+10];
133 
134  /// (1.14) pitch gain of current and five previous subframes
135  int16_t past_gain_pitch[6];
136 
137  /// (14.1) gain code from current and previous subframe
138  int16_t past_gain_code[2];
139 
140  /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
141  int16_t voice_decision;
142 
143  int16_t onset; ///< detected onset level (0-2)
144  int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
145  int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
146  int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
147  uint16_t rand_value; ///< random number generator value (4.4.4)
148  int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
149 
150  /// (14.14) high-pass filter data (past input)
151  int hpf_f[2];
152 
153  /// high-pass filter data (past output)
154  int16_t hpf_z[2];
155 } G729Context;
156 
158  .ac_index_bits = {8,5},
159  .parity_bit = 1,
160  .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
161  .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
162  .fc_signs_bits = 4,
163  .fc_indexes_bits = 13,
164 };
165 
167  .ac_index_bits = {8,4},
168  .parity_bit = 0,
169  .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
170  .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
171  .fc_signs_bits = 2,
172  .fc_indexes_bits = 9,
173 };
174 
175 /**
176  * @brief pseudo random number generator
177  */
178 static inline uint16_t g729_prng(uint16_t value)
179 {
180  return 31821 * value + 13849;
181 }
182 
183 /**
184  * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
185  * @param[out] lsfq (2.13) quantized LSF coefficients
186  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
187  * @param ma_predictor switched MA predictor of LSP quantizer
188  * @param vq_1st first stage vector of quantizer
189  * @param vq_2nd_low second stage lower vector of LSP quantizer
190  * @param vq_2nd_high second stage higher vector of LSP quantizer
191  */
192 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
193  int16_t ma_predictor,
194  int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
195 {
196  int i,j;
197  static const uint8_t min_distance[2]={10, 5}; //(2.13)
198  int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
199 
200  for (i = 0; i < 5; i++) {
201  quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
202  quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
203  }
204 
205  for (j = 0; j < 2; j++) {
206  for (i = 1; i < 10; i++) {
207  int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
208  if (diff > 0) {
209  quantizer_output[i - 1] -= diff;
210  quantizer_output[i ] += diff;
211  }
212  }
213  }
214 
215  for (i = 0; i < 10; i++) {
216  int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
217  for (j = 0; j < MA_NP; j++)
218  sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
219 
220  lsfq[i] = sum >> 15;
221  }
222 
224 }
225 
226 /**
227  * Restores past LSP quantizer output using LSF from previous frame
228  * @param[in,out] lsfq (2.13) quantized LSF coefficients
229  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
230  * @param ma_predictor_prev MA predictor from previous frame
231  * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
232  */
233 static void lsf_restore_from_previous(int16_t* lsfq,
234  int16_t* past_quantizer_outputs[MA_NP + 1],
235  int ma_predictor_prev)
236 {
237  int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
238  int i,k;
239 
240  for (i = 0; i < 10; i++) {
241  int tmp = lsfq[i] << 15;
242 
243  for (k = 0; k < MA_NP; k++)
244  tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
245 
246  quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
247  }
248 }
249 
250 /**
251  * Constructs new excitation signal and applies phase filter to it
252  * @param[out] out constructed speech signal
253  * @param in original excitation signal
254  * @param fc_cur (2.13) original fixed-codebook vector
255  * @param gain_code (14.1) gain code
256  * @param subframe_size length of the subframe
257  */
258 static void g729d_get_new_exc(
259  int16_t* out,
260  const int16_t* in,
261  const int16_t* fc_cur,
262  int dstate,
263  int gain_code,
264  int subframe_size)
265 {
266  int i;
267  int16_t fc_new[SUBFRAME_SIZE];
268 
269  ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
270 
271  for(i=0; i<subframe_size; i++)
272  {
273  out[i] = in[i];
274  out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
275  out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
276  }
277 }
278 
279 /**
280  * Makes decision about onset in current subframe
281  * @param past_onset decision result of previous subframe
282  * @param past_gain_code gain code of current and previous subframe
283  *
284  * @return onset decision result for current subframe
285  */
286 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
287 {
288  if((past_gain_code[0] >> 1) > past_gain_code[1])
289  return 2;
290  else
291  return FFMAX(past_onset-1, 0);
292 }
293 
294 /**
295  * Makes decision about voice presence in current subframe
296  * @param onset onset level
297  * @param prev_voice_decision voice decision result from previous subframe
298  * @param past_gain_pitch pitch gain of current and previous subframes
299  *
300  * @return voice decision result for current subframe
301  */
302 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
303 {
304  int i, low_gain_pitch_cnt, voice_decision;
305 
306  if(past_gain_pitch[0] >= 14745) // 0.9
307  voice_decision = DECISION_VOICE;
308  else if (past_gain_pitch[0] <= 9830) // 0.6
309  voice_decision = DECISION_NOISE;
310  else
311  voice_decision = DECISION_INTERMEDIATE;
312 
313  for(i=0, low_gain_pitch_cnt=0; i<6; i++)
314  if(past_gain_pitch[i] < 9830)
315  low_gain_pitch_cnt++;
316 
317  if(low_gain_pitch_cnt > 2 && !onset)
318  voice_decision = DECISION_NOISE;
319 
320  if(!onset && voice_decision > prev_voice_decision + 1)
321  voice_decision--;
322 
323  if(onset && voice_decision < DECISION_VOICE)
324  voice_decision++;
325 
326  return voice_decision;
327 }
328 
329 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
330 {
331  int res = 0;
332 
333  while (order--)
334  res += *v1++ * *v2++;
335 
336  return res;
337 }
338 
340 {
341  G729Context* ctx = avctx->priv_data;
342  int i,k;
343 
344  if (avctx->channels != 1) {
345  av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
346  return AVERROR(EINVAL);
347  }
348  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
349 
350  /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
351  avctx->frame_size = SUBFRAME_SIZE << 1;
352 
353  ctx->gain_coeff = 16384; // 1.0 in (1.14)
354 
355  for (k = 0; k < MA_NP + 1; k++) {
357  for (i = 1; i < 11; i++)
358  ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
359  }
360 
361  ctx->lsp[0] = ctx->lsp_buf[0];
362  ctx->lsp[1] = ctx->lsp_buf[1];
363  memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
364 
366 
368 
369  /* random seed initialization */
370  ctx->rand_value = 21845;
371 
372  /* quantized prediction error */
373  for(i=0; i<4; i++)
374  ctx->quant_energy[i] = -14336; // -14 in (5.10)
375 
376  ff_audiodsp_init(&ctx->adsp);
378 
379  return 0;
380 }
381 
382 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
383  AVPacket *avpkt)
384 {
385  const uint8_t *buf = avpkt->data;
386  int buf_size = avpkt->size;
387  int16_t *out_frame;
388  GetBitContext gb;
390  int frame_erasure = 0; ///< frame erasure detected during decoding
391  int bad_pitch = 0; ///< parity check failed
392  int i;
393  int16_t *tmp;
394  G729Formats packet_type;
395  G729Context *ctx = avctx->priv_data;
396  int16_t lp[2][11]; // (3.12)
397  uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
398  uint8_t quantizer_1st; ///< first stage vector of quantizer
399  uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
400  uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
401 
402  int pitch_delay_int[2]; // pitch delay, integer part
403  int pitch_delay_3x; // pitch delay, multiplied by 3
404  int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
405  int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
406  int j, ret;
407  int gain_before, gain_after;
408  int is_periodic = 0; // whether one of the subframes is declared as periodic or not
409  AVFrame *frame = data;
410 
411  frame->nb_samples = SUBFRAME_SIZE<<1;
412  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
413  return ret;
414  out_frame = (int16_t*) frame->data[0];
415 
416  if (buf_size % 10 == 0) {
417  packet_type = FORMAT_G729_8K;
418  format = &format_g729_8k;
419  //Reset voice decision
420  ctx->onset = 0;
422  av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
423  } else if (buf_size == 8) {
424  packet_type = FORMAT_G729D_6K4;
425  format = &format_g729d_6k4;
426  av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
427  } else {
428  av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
429  return AVERROR_INVALIDDATA;
430  }
431 
432  for (i=0; i < buf_size; i++)
433  frame_erasure |= buf[i];
434  frame_erasure = !frame_erasure;
435 
436  init_get_bits(&gb, buf, 8*buf_size);
437 
438  ma_predictor = get_bits(&gb, 1);
439  quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
440  quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
441  quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
442 
443  if(frame_erasure)
445  ctx->ma_predictor_prev);
446  else {
448  ma_predictor,
449  quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
450  ctx->ma_predictor_prev = ma_predictor;
451  }
452 
453  tmp = ctx->past_quantizer_outputs[MA_NP];
454  memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
455  MA_NP * sizeof(int16_t*));
456  ctx->past_quantizer_outputs[0] = tmp;
457 
458  ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
459 
460  ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
461 
462  FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
463 
464  for (i = 0; i < 2; i++) {
465  int gain_corr_factor;
466 
467  uint8_t ac_index; ///< adaptive codebook index
468  uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
469  int fc_indexes; ///< fixed-codebook indexes
470  uint8_t gc_1st_index; ///< gain codebook (first stage) index
471  uint8_t gc_2nd_index; ///< gain codebook (second stage) index
472 
473  ac_index = get_bits(&gb, format->ac_index_bits[i]);
474  if(!i && format->parity_bit)
475  bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
476  fc_indexes = get_bits(&gb, format->fc_indexes_bits);
477  pulses_signs = get_bits(&gb, format->fc_signs_bits);
478  gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
479  gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
480 
481  if (frame_erasure)
482  pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
483  else if(!i) {
484  if (bad_pitch)
485  pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
486  else
487  pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
488  } else {
489  int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
491 
492  if(packet_type == FORMAT_G729D_6K4)
493  pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
494  else
495  pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
496  }
497 
498  /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
499  pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
500  if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
501  av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
502  pitch_delay_int[i] = PITCH_DELAY_MAX;
503  }
504 
505  if (frame_erasure) {
506  ctx->rand_value = g729_prng(ctx->rand_value);
507  fc_indexes = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
508 
509  ctx->rand_value = g729_prng(ctx->rand_value);
510  pulses_signs = ctx->rand_value;
511  }
512 
513 
514  memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
515  switch (packet_type) {
516  case FORMAT_G729_8K:
519  fc_indexes, pulses_signs, 3, 3);
520  break;
521  case FORMAT_G729D_6K4:
524  fc_indexes, pulses_signs, 1, 4);
525  break;
526  }
527 
528  /*
529  This filter enhances harmonic components of the fixed-codebook vector to
530  improve the quality of the reconstructed speech.
531 
532  / fc_v[i], i < pitch_delay
533  fc_v[i] = <
534  \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
535  */
536  ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
537  fc + pitch_delay_int[i],
538  fc, 1 << 14,
539  av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
540  0, 14,
541  SUBFRAME_SIZE - pitch_delay_int[i]);
542 
543  memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
544  ctx->past_gain_code[1] = ctx->past_gain_code[0];
545 
546  if (frame_erasure) {
547  ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
548  ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
549 
550  gain_corr_factor = 0;
551  } else {
552  if (packet_type == FORMAT_G729D_6K4) {
553  ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
554  cb_gain_2nd_6k4[gc_2nd_index][0];
555  gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
556  cb_gain_2nd_6k4[gc_2nd_index][1];
557 
558  /* Without check below overflow can occur in ff_acelp_update_past_gain.
559  It is not issue for G.729, because gain_corr_factor in it's case is always
560  greater than 1024, while in G.729D it can be even zero. */
561  gain_corr_factor = FFMAX(gain_corr_factor, 1024);
562 #ifndef G729_BITEXACT
563  gain_corr_factor >>= 1;
564 #endif
565  } else {
566  ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
567  cb_gain_2nd_8k[gc_2nd_index][0];
568  gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
569  cb_gain_2nd_8k[gc_2nd_index][1];
570  }
571 
572  /* Decode the fixed-codebook gain. */
573  ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->adsp, gain_corr_factor,
574  fc, MR_ENERGY,
575  ctx->quant_energy,
577  SUBFRAME_SIZE, 4);
578 #ifdef G729_BITEXACT
579  /*
580  This correction required to get bit-exact result with
581  reference code, because gain_corr_factor in G.729D is
582  two times larger than in original G.729.
583 
584  If bit-exact result is not issue then gain_corr_factor
585  can be simpler divided by 2 before call to g729_get_gain_code
586  instead of using correction below.
587  */
588  if (packet_type == FORMAT_G729D_6K4) {
589  gain_corr_factor >>= 1;
590  ctx->past_gain_code[0] >>= 1;
591  }
592 #endif
593  }
594  ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
595 
596  /* Routine requires rounding to lowest. */
597  ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
598  ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
600  (pitch_delay_3x % 3) << 1,
601  10, SUBFRAME_SIZE);
602 
603  ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
604  ctx->exc + i * SUBFRAME_SIZE, fc,
605  (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
606  ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
607  1 << 13, 14, SUBFRAME_SIZE);
608 
609  memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
610 
612  synth+10,
613  &lp[i][1],
614  ctx->exc + i * SUBFRAME_SIZE,
615  SUBFRAME_SIZE,
616  10,
617  1,
618  0,
619  0x800))
620  /* Overflow occurred, downscale excitation signal... */
621  for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
622  ctx->exc_base[j] >>= 2;
623 
624  /* ... and make synthesis again. */
625  if (packet_type == FORMAT_G729D_6K4) {
626  int16_t exc_new[SUBFRAME_SIZE];
627 
628  ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
630 
631  g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
632 
634  synth+10,
635  &lp[i][1],
636  exc_new,
637  SUBFRAME_SIZE,
638  10,
639  0,
640  0,
641  0x800);
642  } else {
644  synth+10,
645  &lp[i][1],
646  ctx->exc + i * SUBFRAME_SIZE,
647  SUBFRAME_SIZE,
648  10,
649  0,
650  0,
651  0x800);
652  }
653  /* Save data (without postfilter) for use in next subframe. */
654  memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
655 
656  /* Calculate gain of unfiltered signal for use in AGC. */
657  gain_before = 0;
658  for (j = 0; j < SUBFRAME_SIZE; j++)
659  gain_before += FFABS(synth[j+10]);
660 
661  /* Call postfilter and also update voicing decision for use in next frame. */
663  &ctx->adsp,
664  &ctx->ht_prev_data,
665  &is_periodic,
666  &lp[i][0],
667  pitch_delay_int[0],
668  ctx->residual,
669  ctx->res_filter_data,
670  ctx->pos_filter_data,
671  synth+10,
672  SUBFRAME_SIZE);
673 
674  /* Calculate gain of filtered signal for use in AGC. */
675  gain_after = 0;
676  for(j=0; j<SUBFRAME_SIZE; j++)
677  gain_after += FFABS(synth[j+10]);
678 
680  gain_before,
681  gain_after,
682  synth+10,
683  SUBFRAME_SIZE,
684  ctx->gain_coeff);
685 
686  if (frame_erasure)
688  else
689  ctx->pitch_delay_int_prev = pitch_delay_int[i];
690 
691  memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
693  out_frame + i*SUBFRAME_SIZE,
694  ctx->hpf_f,
695  synth+10,
696  SUBFRAME_SIZE);
697  memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
698  }
699 
700  ctx->was_periodic = is_periodic;
701 
702  /* Save signal for use in next frame. */
703  memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
704 
705  *got_frame_ptr = 1;
706  return packet_type == FORMAT_G729_8K ? 10 : 8;
707 }
708 
710  .name = "g729",
711  .long_name = NULL_IF_CONFIG_SMALL("G.729"),
712  .type = AVMEDIA_TYPE_AUDIO,
713  .id = AV_CODEC_ID_G729,
714  .priv_data_size = sizeof(G729Context),
715  .init = decoder_init,
716  .decode = decode_frame,
717  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
718 };
uint16_t rand_value
random number generator value (4.4.4)
Definition: g729dec.c:147
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_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).
AudioDSPContext adsp
Definition: g729dec.c:103
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
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
int32_t(* scalarproduct_int16)(const int16_t *v1, const int16_t *v2, int len)
Calculate scalar product of two vectors.
Definition: audiodsp.h:29
#define LSFQ_MIN
minimum quantized LSF value (3.2.4) 0.005 in Q13
Definition: g729dec.c:46
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
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
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
void ff_acelp_lsf2lsp(int16_t *lsp, const int16_t *lsf, int lp_order)
Convert LSF to LSP.
Definition: lsp.c:83
int16_t res_filter_data[SUBFRAME_SIZE+10]
previous speech data for residual calculation filter
Definition: g729dec.c:129
#define GC_2ND_IDX_BITS_8K
gain codebook (second stage) index, 8k mode (size in bits)
Definition: g729data.h:33
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 unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:247
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
const int16_t ff_acelp_interp_filter[61]
low-pass Finite Impulse Response filter coefficients.
Definition: acelp_filters.c:30
#define MR_ENERGY
MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7...
Definition: g729dec.c:81
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
int hpf_f[2]
(14.14) high-pass filter data (past input)
Definition: g729dec.c:151
int size
Definition: avcodec.h:1589
const uint8_t ff_fc_2pulses_9bits_track1_gray[16]
Definition: acelp_vectors.c:42
int ff_acelp_decode_4bit_to_2nd_delay3(int ac_index, int pitch_delay_min)
Decode pitch delay with 1/3 precision.
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
external API header
av_cold void ff_audiodsp_init(AudioDSPContext *c)
Definition: audiodsp.c:106
static const G729FormatDescription format_g729_8k
Definition: g729dec.c:157
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.
int16_t lsp_buf[2][10]
(0.15) LSP coefficients (previous and current frames) (3.2.5)
Definition: g729dec.c:116
#define LSFQ_DIFF_MIN
minimum LSF distance (3.2.4) 0.0391 in Q13
Definition: g729dec.c:58
AVCodec.
Definition: avcodec.h:3559
int16_t past_quantizer_output_buf[MA_NP+1][10]
(2.13) LSP quantizer outputs
Definition: g729dec.c:112
static int32_t scalarproduct_int16_c(const int16_t *v1, const int16_t *v2, int order)
Definition: g729dec.c:329
int16_t * exc
start of past excitation data in buffer
Definition: g729dec.c:108
uint8_t fc_indexes_bits
size (in bits) of fixed-codebook index entry
Definition: g729dec.c:99
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2426
const uint8_t ff_fc_4pulses_8bits_track_4[32]
Track|Pulse| Positions 4 | 3 | 3, 8, 13, 18, 23, 28, 33, 38, 43, 48, 53, 58, 63, 68, 73, 78 | | 4, 9, 14, 19, 24, 29, 34, 39, 44, 49, 54, 59, 64, 69, 74, 79
Definition: acelp_vectors.c:79
uint8_t
#define av_cold
Definition: attributes.h:82
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:258
#define PITCH_DELAY_MAX
AVCodec ff_g729_decoder
Definition: g729dec.c:709
static AVFrame * frame
int16_t onset
detected onset level (0-2)
Definition: g729dec.c:143
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.
#define DECISION_VOICE
Definition: g729dec.c:85
uint8_t * data
Definition: avcodec.h:1588
int16_t past_gain_code[2]
(14.1) gain code from current and previous subframe
Definition: g729dec.c:138
bitstream reader API header.
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:302
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)
#define DECISION_NOISE
Definition: g729dec.c:83
int16_t * past_quantizer_outputs[MA_NP+1]
Definition: g729dec.c:113
#define av_log(a,...)
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:192
int16_t hpf_z[2]
high-pass filter data (past output)
Definition: g729dec.c:154
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
uint8_t gc_2nd_index_bits
gain codebook (second stage) index (size in bits)
Definition: g729dec.c:97
int16_t ht_prev_data
previous data for 4.2.3, equation 86
Definition: g729dec.c:145
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
int16_t lsfq[10]
(2.13) quantized LSF coefficients from previous frame
Definition: g729dec.c:115
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
#define DECISION_INTERMEDIATE
Definition: g729dec.c:84
const char * name
Name of the codec implementation.
Definition: avcodec.h:3566
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.6.2 of G.729 and 6.1.5 of AMR).
Definition: celp_filters.c:30
static av_cold int decoder_init(AVCodecContext *avctx)
Definition: g729dec.c:339
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
uint8_t ac_index_bits[2]
adaptive codebook index for second subframe (size in bits)
Definition: g729dec.c:94
#define FFMAX(a, b)
Definition: common.h:94
uint64_t residual
Definition: dirac_vlc.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
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 uint16_t fc[]
Definition: dcaenc.h:41
uint8_t parity_bit
parity bit for pitch delay
Definition: g729dec.c:95
#define FFMIN(a, b)
Definition: common.h:96
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
uint8_t fc_signs_bits
number of pulses in fixed-codebook vector
Definition: g729dec.c:98
#define GC_2ND_IDX_BITS_6K4
gain codebook (second stage) index, 6.4k mode (size in bits)
Definition: g729data.h:36
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
int16_t residual[SUBFRAME_SIZE+RES_PREV_DATA_SIZE]
residual signal buffer (used in long-term postfilter)
Definition: g729dec.c:126
int pitch_delay_int_prev
integer part of previous subframe's pitch delay (4.1.3)
Definition: g729dec.c:109
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
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.
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 (3.2.5 and 3.2.6 of G.729)
Definition: lsp.c:171
static const int16_t cb_ma_predictor_sum_inv[2][10]
12 ...
Definition: g729data.h:335
int ma_predictor_prev
switched MA predictor of LSP quantizer from last good frame
Definition: g729dec.c:148
int16_t * lsp[2]
pointers to lsp_buf
Definition: g729dec.c:117
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2438
#define SHARP_MIN
minimum gain pitch value (3.8, Equation 47) 0.2 in (1.14)
Definition: g729dec.c:67
Libavcodec external API header.
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:233
uint8_t gc_1st_index_bits
gain codebook (first stage) index (size in bits)
Definition: g729dec.c:96
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: g729dec.c:382
int16_t past_gain_pitch[6]
(1.14) pitch gain of current and five previous subframes
Definition: g729dec.c:135
main external API structure.
Definition: avcodec.h:1657
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:928
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
int16_t quant_energy[4]
(5.10) past quantized energy
Definition: g729dec.c:119
void * buf
Definition: avisynth_c.h:553
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:299
const uint8_t ff_fc_4pulses_8bits_tracks_13[16]
Track|Pulse| Positions 1 | 0 | 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75 2 | 1 | 1, 6, 11, 16, 21, 26, 31, 36, 41, 46, 51, 56, 61, 66, 71, 76 3 | 2 | 2, 7, 12, 17, 22, 27, 32, 37, 42, 47, 52, 57, 62, 67, 72, 77
Definition: acelp_vectors.c:74
static const char * format
Definition: movenc.c:47
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time...
Definition: avcodec.h:1006
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:406
#define INTERPOL_LEN
interpolation filter length
Definition: g729dec.c:61
G729Formats
Definition: g729dec.c:87
int16_t was_periodic
whether previous frame was declared as periodic or not (4.4)
Definition: g729dec.c:144
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 MA_NP
Moving Average (MA) prediction order.
Definition: g729data.h:27
const uint8_t ff_fc_2pulses_9bits_track2_gray[32]
Track|Pulse| Positions 2 | 1 | 0, 7, 14, 20, 27, 34, 1, 21 | | 2, 9, 15, 22, 29, 35, 6, 26 | | 4,10, 17, 24, 30, 37, 11, 31 | | 5,12, 19, 25, 32, 39, 16, 36
Definition: acelp_vectors.c:54
#define LSFQ_MAX
maximum quantized LSF value (3.2.4) 3.135 in Q13
Definition: g729dec.c:52
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:198
#define SHARP_MAX
maximum gain pitch value (3.8, Equation 47) (EE) This does not comply with the specification.
Definition: g729dec.c:76
#define av_parity
Definition: intmath.h:158
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:752
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
#define VQ_2ND_BITS
second stage vector of quantizer (size in bits)
Definition: g729data.h:30
#define GC_1ST_IDX_BITS_8K
gain codebook (first stage) index, 8k mode (size in bits)
Definition: g729data.h:32
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:282
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
signed 16 bits
Definition: samplefmt.h:61
int16_t syn_filter_data[10]
previous speech data for LP synthesis filter
Definition: g729dec.c:122
static int g729d_onset_decision(int past_onset, const int16_t *past_gain_code)
Makes decision about onset in current subframe.
Definition: g729dec.c:286
int16_t exc_base[2 *SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN]
past excitation signal buffer
Definition: g729dec.c:106
#define VQ_1ST_BITS
first stage vector of quantizer (size in bits)
Definition: g729data.h:29
#define GC_1ST_IDX_BITS_6K4
gain codebook (first stage) index, 6.4k mode (size in bits)
Definition: g729data.h:35
void * priv_data
Definition: avcodec.h:1699
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
Definition: avcodec.h:2419
#define SUBFRAME_SIZE
Definition: evrcdec.c:41
static uint8_t tmp[8]
Definition: des.c:38
static const int16_t cb_ma_predictor_sum[2][10]
15 3 cb_ma_predictor_sum[j][i] = floor( 2 * (1...
Definition: g729data.h:321
static void frame_erasure(EVRCContext *e, float *samples)
Definition: evrcdec.c:652
static uint16_t g729_prng(uint16_t value)
pseudo random number generator
Definition: g729dec.c:178
#define RES_PREV_DATA_SIZE
Amount of past residual signal data stored in buffer.
int16_t voice_decision
voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
Definition: g729dec.c:141
static const G729FormatDescription format_g729d_6k4
Definition: g729dec.c:166
FILE * out
Definition: movenc.c:54
int16_t pos_filter_data[SUBFRAME_SIZE+10]
previous speech data for short-term postfilter
Definition: g729dec.c:132
#define PITCH_DELAY_MIN
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
int gain_coeff
(1.14) gain coefficient (4.2.4)
Definition: g729dec.c:146
#define FFSWAP(type, a, b)
Definition: common.h:99
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 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
This structure stores compressed data.
Definition: avcodec.h:1565
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:956
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 const int16_t phase_filter[3][40]
additional "phase" post-processing filter impulse response (D.6.2 of G.729)
Definition: g729data.h:361