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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 "dsputil.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 {
105 
106  /// past excitation signal buffer
108 
109  int16_t* exc; ///< start of past excitation data in buffer
110  int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
111 
112  /// (2.13) LSP quantizer outputs
113  int16_t past_quantizer_output_buf[MA_NP + 1][10];
114  int16_t* past_quantizer_outputs[MA_NP + 1];
115 
116  int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
117  int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
118  int16_t *lsp[2]; ///< pointers to lsp_buf
119 
120  int16_t quant_energy[4]; ///< (5.10) past quantized energy
121 
122  /// previous speech data for LP synthesis filter
123  int16_t syn_filter_data[10];
124 
125 
126  /// residual signal buffer (used in long-term postfilter)
127  int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
128 
129  /// previous speech data for residual calculation filter
130  int16_t res_filter_data[SUBFRAME_SIZE+10];
131 
132  /// previous speech data for short-term postfilter
133  int16_t pos_filter_data[SUBFRAME_SIZE+10];
134 
135  /// (1.14) pitch gain of current and five previous subframes
136  int16_t past_gain_pitch[6];
137 
138  /// (14.1) gain code from current and previous subframe
139  int16_t past_gain_code[2];
140 
141  /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
142  int16_t voice_decision;
143 
144  int16_t onset; ///< detected onset level (0-2)
145  int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
146  int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
147  int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
148  uint16_t rand_value; ///< random number generator value (4.4.4)
149  int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
150 
151  /// (14.14) high-pass filter data (past input)
152  int hpf_f[2];
153 
154  /// high-pass filter data (past output)
155  int16_t hpf_z[2];
156 } G729Context;
157 
159  .ac_index_bits = {8,5},
160  .parity_bit = 1,
161  .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
162  .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
163  .fc_signs_bits = 4,
164  .fc_indexes_bits = 13,
165 };
166 
168  .ac_index_bits = {8,4},
169  .parity_bit = 0,
170  .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
171  .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
172  .fc_signs_bits = 2,
173  .fc_indexes_bits = 9,
174 };
175 
176 /**
177  * @brief pseudo random number generator
178  */
179 static inline uint16_t g729_prng(uint16_t value)
180 {
181  return 31821 * value + 13849;
182 }
183 
184 /**
185  * Get parity bit of bit 2..7
186  */
187 static inline int get_parity(uint8_t value)
188 {
189  return (0x6996966996696996ULL >> (value >> 2)) & 1;
190 }
191 
192 /**
193  * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
194  * @param[out] lsfq (2.13) quantized LSF coefficients
195  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
196  * @param ma_predictor switched MA predictor of LSP quantizer
197  * @param vq_1st first stage vector of quantizer
198  * @param vq_2nd_low second stage lower vector of LSP quantizer
199  * @param vq_2nd_high second stage higher vector of LSP quantizer
200  */
201 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
202  int16_t ma_predictor,
203  int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
204 {
205  int i,j;
206  static const uint8_t min_distance[2]={10, 5}; //(2.13)
207  int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
208 
209  for (i = 0; i < 5; i++) {
210  quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
211  quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
212  }
213 
214  for (j = 0; j < 2; j++) {
215  for (i = 1; i < 10; i++) {
216  int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
217  if (diff > 0) {
218  quantizer_output[i - 1] -= diff;
219  quantizer_output[i ] += diff;
220  }
221  }
222  }
223 
224  for (i = 0; i < 10; i++) {
225  int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
226  for (j = 0; j < MA_NP; j++)
227  sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
228 
229  lsfq[i] = sum >> 15;
230  }
231 
233 }
234 
235 /**
236  * Restores past LSP quantizer output using LSF from previous frame
237  * @param[in,out] lsfq (2.13) quantized LSF coefficients
238  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
239  * @param ma_predictor_prev MA predictor from previous frame
240  * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
241  */
242 static void lsf_restore_from_previous(int16_t* lsfq,
243  int16_t* past_quantizer_outputs[MA_NP + 1],
244  int ma_predictor_prev)
245 {
246  int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
247  int i,k;
248 
249  for (i = 0; i < 10; i++) {
250  int tmp = lsfq[i] << 15;
251 
252  for (k = 0; k < MA_NP; k++)
253  tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
254 
255  quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
256  }
257 }
258 
259 /**
260  * Constructs new excitation signal and applies phase filter to it
261  * @param[out] out constructed speech signal
262  * @param in original excitation signal
263  * @param fc_cur (2.13) original fixed-codebook vector
264  * @param gain_code (14.1) gain code
265  * @param subframe_size length of the subframe
266  */
267 static void g729d_get_new_exc(
268  int16_t* out,
269  const int16_t* in,
270  const int16_t* fc_cur,
271  int dstate,
272  int gain_code,
273  int subframe_size)
274 {
275  int i;
276  int16_t fc_new[SUBFRAME_SIZE];
277 
278  ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
279 
280  for(i=0; i<subframe_size; i++)
281  {
282  out[i] = in[i];
283  out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
284  out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
285  }
286 }
287 
288 /**
289  * Makes decision about onset in current subframe
290  * @param past_onset decision result of previous subframe
291  * @param past_gain_code gain code of current and previous subframe
292  *
293  * @return onset decision result for current subframe
294  */
295 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
296 {
297  if((past_gain_code[0] >> 1) > past_gain_code[1])
298  return 2;
299  else
300  return FFMAX(past_onset-1, 0);
301 }
302 
303 /**
304  * Makes decision about voice presence in current subframe
305  * @param onset onset level
306  * @param prev_voice_decision voice decision result from previous subframe
307  * @param past_gain_pitch pitch gain of current and previous subframes
308  *
309  * @return voice decision result for current subframe
310  */
311 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
312 {
313  int i, low_gain_pitch_cnt, voice_decision;
314 
315  if(past_gain_pitch[0] >= 14745) // 0.9
316  voice_decision = DECISION_VOICE;
317  else if (past_gain_pitch[0] <= 9830) // 0.6
318  voice_decision = DECISION_NOISE;
319  else
320  voice_decision = DECISION_INTERMEDIATE;
321 
322  for(i=0, low_gain_pitch_cnt=0; i<6; i++)
323  if(past_gain_pitch[i] < 9830)
324  low_gain_pitch_cnt++;
325 
326  if(low_gain_pitch_cnt > 2 && !onset)
327  voice_decision = DECISION_NOISE;
328 
329  if(!onset && voice_decision > prev_voice_decision + 1)
330  voice_decision--;
331 
332  if(onset && voice_decision < DECISION_VOICE)
333  voice_decision++;
334 
335  return voice_decision;
336 }
337 
338 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
339 {
340  int res = 0;
341 
342  while (order--)
343  res += *v1++ * *v2++;
344 
345  return res;
346 }
347 
349 {
350  G729Context* ctx = avctx->priv_data;
351  int i,k;
352 
353  if (avctx->channels != 1) {
354  av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
355  return AVERROR(EINVAL);
356  }
357  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
358 
359  /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
360  avctx->frame_size = SUBFRAME_SIZE << 1;
361 
362  ctx->gain_coeff = 16384; // 1.0 in (1.14)
363 
364  for (k = 0; k < MA_NP + 1; k++) {
366  for (i = 1; i < 11; i++)
367  ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
368  }
369 
370  ctx->lsp[0] = ctx->lsp_buf[0];
371  ctx->lsp[1] = ctx->lsp_buf[1];
372  memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
373 
375 
377 
378  /* random seed initialization */
379  ctx->rand_value = 21845;
380 
381  /* quantized prediction error */
382  for(i=0; i<4; i++)
383  ctx->quant_energy[i] = -14336; // -14 in (5.10)
384 
385  ff_dsputil_init(&ctx->dsp, avctx);
387 
389  avctx->coded_frame = &ctx->frame;
390 
391  return 0;
392 }
393 
394 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
395  AVPacket *avpkt)
396 {
397  const uint8_t *buf = avpkt->data;
398  int buf_size = avpkt->size;
399  int16_t *out_frame;
400  GetBitContext gb;
401  const G729FormatDescription *format;
402  int frame_erasure = 0; ///< frame erasure detected during decoding
403  int bad_pitch = 0; ///< parity check failed
404  int i;
405  int16_t *tmp;
406  G729Formats packet_type;
407  G729Context *ctx = avctx->priv_data;
408  int16_t lp[2][11]; // (3.12)
409  uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
410  uint8_t quantizer_1st; ///< first stage vector of quantizer
411  uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
412  uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
413 
414  int pitch_delay_int[2]; // pitch delay, integer part
415  int pitch_delay_3x; // pitch delay, multiplied by 3
416  int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
417  int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
418  int j, ret;
419  int gain_before, gain_after;
420  int is_periodic = 0; // whether one of the subframes is declared as periodic or not
421 
422  ctx->frame.nb_samples = SUBFRAME_SIZE<<1;
423  if ((ret = ff_get_buffer(avctx, &ctx->frame)) < 0) {
424  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
425  return ret;
426  }
427  out_frame = (int16_t*) ctx->frame.data[0];
428 
429  if (buf_size == 10) {
430  packet_type = FORMAT_G729_8K;
431  format = &format_g729_8k;
432  //Reset voice decision
433  ctx->onset = 0;
435  av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
436  } else if (buf_size == 8) {
437  packet_type = FORMAT_G729D_6K4;
438  format = &format_g729d_6k4;
439  av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
440  } else {
441  av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
442  return AVERROR_INVALIDDATA;
443  }
444 
445  for (i=0; i < buf_size; i++)
446  frame_erasure |= buf[i];
447  frame_erasure = !frame_erasure;
448 
449  init_get_bits(&gb, buf, 8*buf_size);
450 
451  ma_predictor = get_bits(&gb, 1);
452  quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
453  quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
454  quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
455 
456  if(frame_erasure)
458  ctx->ma_predictor_prev);
459  else {
461  ma_predictor,
462  quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
463  ctx->ma_predictor_prev = ma_predictor;
464  }
465 
466  tmp = ctx->past_quantizer_outputs[MA_NP];
467  memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
468  MA_NP * sizeof(int16_t*));
469  ctx->past_quantizer_outputs[0] = tmp;
470 
471  ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
472 
473  ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
474 
475  FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
476 
477  for (i = 0; i < 2; i++) {
478  int gain_corr_factor;
479 
480  uint8_t ac_index; ///< adaptive codebook index
481  uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
482  int fc_indexes; ///< fixed-codebook indexes
483  uint8_t gc_1st_index; ///< gain codebook (first stage) index
484  uint8_t gc_2nd_index; ///< gain codebook (second stage) index
485 
486  ac_index = get_bits(&gb, format->ac_index_bits[i]);
487  if(!i && format->parity_bit)
488  bad_pitch = get_parity(ac_index) == get_bits1(&gb);
489  fc_indexes = get_bits(&gb, format->fc_indexes_bits);
490  pulses_signs = get_bits(&gb, format->fc_signs_bits);
491  gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
492  gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
493 
494  if (frame_erasure)
495  pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
496  else if(!i) {
497  if (bad_pitch)
498  pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
499  else
500  pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
501  } else {
502  int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
504 
505  if(packet_type == FORMAT_G729D_6K4)
506  pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
507  else
508  pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
509  }
510 
511  /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
512  pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
513  if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
514  av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
515  pitch_delay_int[i] = PITCH_DELAY_MAX;
516  }
517 
518  if (frame_erasure) {
519  ctx->rand_value = g729_prng(ctx->rand_value);
520  fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
521 
522  ctx->rand_value = g729_prng(ctx->rand_value);
523  pulses_signs = ctx->rand_value;
524  }
525 
526 
527  memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
528  switch (packet_type) {
529  case FORMAT_G729_8K:
532  fc_indexes, pulses_signs, 3, 3);
533  break;
534  case FORMAT_G729D_6K4:
537  fc_indexes, pulses_signs, 1, 4);
538  break;
539  }
540 
541  /*
542  This filter enhances harmonic components of the fixed-codebook vector to
543  improve the quality of the reconstructed speech.
544 
545  / fc_v[i], i < pitch_delay
546  fc_v[i] = <
547  \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
548  */
549  ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
550  fc + pitch_delay_int[i],
551  fc, 1 << 14,
552  av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
553  0, 14,
554  SUBFRAME_SIZE - pitch_delay_int[i]);
555 
556  memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
557  ctx->past_gain_code[1] = ctx->past_gain_code[0];
558 
559  if (frame_erasure) {
560  ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
561  ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
562 
563  gain_corr_factor = 0;
564  } else {
565  if (packet_type == FORMAT_G729D_6K4) {
566  ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
567  cb_gain_2nd_6k4[gc_2nd_index][0];
568  gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
569  cb_gain_2nd_6k4[gc_2nd_index][1];
570 
571  /* Without check below overflow can occur in ff_acelp_update_past_gain.
572  It is not issue for G.729, because gain_corr_factor in it's case is always
573  greater than 1024, while in G.729D it can be even zero. */
574  gain_corr_factor = FFMAX(gain_corr_factor, 1024);
575 #ifndef G729_BITEXACT
576  gain_corr_factor >>= 1;
577 #endif
578  } else {
579  ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
580  cb_gain_2nd_8k[gc_2nd_index][0];
581  gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
582  cb_gain_2nd_8k[gc_2nd_index][1];
583  }
584 
585  /* Decode the fixed-codebook gain. */
586  ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
587  fc, MR_ENERGY,
588  ctx->quant_energy,
590  SUBFRAME_SIZE, 4);
591 #ifdef G729_BITEXACT
592  /*
593  This correction required to get bit-exact result with
594  reference code, because gain_corr_factor in G.729D is
595  two times larger than in original G.729.
596 
597  If bit-exact result is not issue then gain_corr_factor
598  can be simpler divided by 2 before call to g729_get_gain_code
599  instead of using correction below.
600  */
601  if (packet_type == FORMAT_G729D_6K4) {
602  gain_corr_factor >>= 1;
603  ctx->past_gain_code[0] >>= 1;
604  }
605 #endif
606  }
607  ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
608 
609  /* Routine requires rounding to lowest. */
610  ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
611  ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
613  (pitch_delay_3x % 3) << 1,
614  10, SUBFRAME_SIZE);
615 
616  ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
617  ctx->exc + i * SUBFRAME_SIZE, fc,
618  (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
619  ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
620  1 << 13, 14, SUBFRAME_SIZE);
621 
622  memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
623 
625  synth+10,
626  &lp[i][1],
627  ctx->exc + i * SUBFRAME_SIZE,
628  SUBFRAME_SIZE,
629  10,
630  1,
631  0,
632  0x800))
633  /* Overflow occurred, downscale excitation signal... */
634  for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
635  ctx->exc_base[j] >>= 2;
636 
637  /* ... and make synthesis again. */
638  if (packet_type == FORMAT_G729D_6K4) {
639  int16_t exc_new[SUBFRAME_SIZE];
640 
641  ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
643 
644  g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
645 
647  synth+10,
648  &lp[i][1],
649  exc_new,
650  SUBFRAME_SIZE,
651  10,
652  0,
653  0,
654  0x800);
655  } else {
657  synth+10,
658  &lp[i][1],
659  ctx->exc + i * SUBFRAME_SIZE,
660  SUBFRAME_SIZE,
661  10,
662  0,
663  0,
664  0x800);
665  }
666  /* Save data (without postfilter) for use in next subframe. */
667  memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
668 
669  /* Calculate gain of unfiltered signal for use in AGC. */
670  gain_before = 0;
671  for (j = 0; j < SUBFRAME_SIZE; j++)
672  gain_before += FFABS(synth[j+10]);
673 
674  /* Call postfilter and also update voicing decision for use in next frame. */
676  &ctx->dsp,
677  &ctx->ht_prev_data,
678  &is_periodic,
679  &lp[i][0],
680  pitch_delay_int[0],
681  ctx->residual,
682  ctx->res_filter_data,
683  ctx->pos_filter_data,
684  synth+10,
685  SUBFRAME_SIZE);
686 
687  /* Calculate gain of filtered signal for use in AGC. */
688  gain_after = 0;
689  for(j=0; j<SUBFRAME_SIZE; j++)
690  gain_after += FFABS(synth[j+10]);
691 
693  gain_before,
694  gain_after,
695  synth+10,
696  SUBFRAME_SIZE,
697  ctx->gain_coeff);
698 
699  if (frame_erasure)
701  else
702  ctx->pitch_delay_int_prev = pitch_delay_int[i];
703 
704  memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
706  out_frame + i*SUBFRAME_SIZE,
707  ctx->hpf_f,
708  synth+10,
709  SUBFRAME_SIZE);
710  memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
711  }
712 
713  ctx->was_periodic = is_periodic;
714 
715  /* Save signal for use in next frame. */
716  memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
717 
718  *got_frame_ptr = 1;
719  *(AVFrame*)data = ctx->frame;
720  return buf_size;
721 }
722 
724  .name = "g729",
725  .type = AVMEDIA_TYPE_AUDIO,
726  .id = AV_CODEC_ID_G729,
727  .priv_data_size = sizeof(G729Context),
728  .init = decoder_init,
729  .decode = decode_frame,
730  .capabilities = CODEC_CAP_DR1,
731  .long_name = NULL_IF_CONFIG_SMALL("G.729"),
732 };