Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / g729dec.c
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 {
88 FORMAT_G729_8K = 0,
89 FORMAT_G729D_6K4,
90 FORMAT_COUNT,
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
100 } G729FormatDescription;
101
102 typedef struct {
103 AudioDSPContext adsp;
104
105 /// past excitation signal buffer
106 int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
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)
126 int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
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
157 static const G729FormatDescription format_g729_8k = {
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
166 static const G729FormatDescription format_g729d_6k4 = {
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 * Get parity bit of bit 2..7
185 */
186 static inline int get_parity(uint8_t value)
187 {
188 return (0x6996966996696996ULL >> (value >> 2)) & 1;
189 }
190
191 /**
192 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
193 * @param[out] lsfq (2.13) quantized LSF coefficients
194 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
195 * @param ma_predictor switched MA predictor of LSP quantizer
196 * @param vq_1st first stage vector of quantizer
197 * @param vq_2nd_low second stage lower vector of LSP quantizer
198 * @param vq_2nd_high second stage higher vector of LSP quantizer
199 */
200 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
201 int16_t ma_predictor,
202 int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
203 {
204 int i,j;
205 static const uint8_t min_distance[2]={10, 5}; //(2.13)
206 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
207
208 for (i = 0; i < 5; i++) {
209 quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
210 quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
211 }
212
213 for (j = 0; j < 2; j++) {
214 for (i = 1; i < 10; i++) {
215 int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
216 if (diff > 0) {
217 quantizer_output[i - 1] -= diff;
218 quantizer_output[i ] += diff;
219 }
220 }
221 }
222
223 for (i = 0; i < 10; i++) {
224 int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
225 for (j = 0; j < MA_NP; j++)
226 sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
227
228 lsfq[i] = sum >> 15;
229 }
230
231 ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
232 }
233
234 /**
235 * Restores past LSP quantizer output using LSF from previous frame
236 * @param[in,out] lsfq (2.13) quantized LSF coefficients
237 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
238 * @param ma_predictor_prev MA predictor from previous frame
239 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
240 */
241 static void lsf_restore_from_previous(int16_t* lsfq,
242 int16_t* past_quantizer_outputs[MA_NP + 1],
243 int ma_predictor_prev)
244 {
245 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
246 int i,k;
247
248 for (i = 0; i < 10; i++) {
249 int tmp = lsfq[i] << 15;
250
251 for (k = 0; k < MA_NP; k++)
252 tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
253
254 quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
255 }
256 }
257
258 /**
259 * Constructs new excitation signal and applies phase filter to it
260 * @param[out] out constructed speech signal
261 * @param in original excitation signal
262 * @param fc_cur (2.13) original fixed-codebook vector
263 * @param gain_code (14.1) gain code
264 * @param subframe_size length of the subframe
265 */
266 static void g729d_get_new_exc(
267 int16_t* out,
268 const int16_t* in,
269 const int16_t* fc_cur,
270 int dstate,
271 int gain_code,
272 int subframe_size)
273 {
274 int i;
275 int16_t fc_new[SUBFRAME_SIZE];
276
277 ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
278
279 for(i=0; i<subframe_size; i++)
280 {
281 out[i] = in[i];
282 out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
283 out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
284 }
285 }
286
287 /**
288 * Makes decision about onset in current subframe
289 * @param past_onset decision result of previous subframe
290 * @param past_gain_code gain code of current and previous subframe
291 *
292 * @return onset decision result for current subframe
293 */
294 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
295 {
296 if((past_gain_code[0] >> 1) > past_gain_code[1])
297 return 2;
298 else
299 return FFMAX(past_onset-1, 0);
300 }
301
302 /**
303 * Makes decision about voice presence in current subframe
304 * @param onset onset level
305 * @param prev_voice_decision voice decision result from previous subframe
306 * @param past_gain_pitch pitch gain of current and previous subframes
307 *
308 * @return voice decision result for current subframe
309 */
310 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
311 {
312 int i, low_gain_pitch_cnt, voice_decision;
313
314 if(past_gain_pitch[0] >= 14745) // 0.9
315 voice_decision = DECISION_VOICE;
316 else if (past_gain_pitch[0] <= 9830) // 0.6
317 voice_decision = DECISION_NOISE;
318 else
319 voice_decision = DECISION_INTERMEDIATE;
320
321 for(i=0, low_gain_pitch_cnt=0; i<6; i++)
322 if(past_gain_pitch[i] < 9830)
323 low_gain_pitch_cnt++;
324
325 if(low_gain_pitch_cnt > 2 && !onset)
326 voice_decision = DECISION_NOISE;
327
328 if(!onset && voice_decision > prev_voice_decision + 1)
329 voice_decision--;
330
331 if(onset && voice_decision < DECISION_VOICE)
332 voice_decision++;
333
334 return voice_decision;
335 }
336
337 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
338 {
339 int res = 0;
340
341 while (order--)
342 res += *v1++ * *v2++;
343
344 return res;
345 }
346
347 static av_cold int decoder_init(AVCodecContext * avctx)
348 {
349 G729Context* ctx = avctx->priv_data;
350 int i,k;
351
352 if (avctx->channels != 1) {
353 av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
354 return AVERROR(EINVAL);
355 }
356 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
357
358 /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
359 avctx->frame_size = SUBFRAME_SIZE << 1;
360
361 ctx->gain_coeff = 16384; // 1.0 in (1.14)
362
363 for (k = 0; k < MA_NP + 1; k++) {
364 ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
365 for (i = 1; i < 11; i++)
366 ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
367 }
368
369 ctx->lsp[0] = ctx->lsp_buf[0];
370 ctx->lsp[1] = ctx->lsp_buf[1];
371 memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
372
373 ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
374
375 ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
376
377 /* random seed initialization */
378 ctx->rand_value = 21845;
379
380 /* quantized prediction error */
381 for(i=0; i<4; i++)
382 ctx->quant_energy[i] = -14336; // -14 in (5.10)
383
384 ff_audiodsp_init(&ctx->adsp);
385 ctx->adsp.scalarproduct_int16 = scalarproduct_int16_c;
386
387 return 0;
388 }
389
390 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
391 AVPacket *avpkt)
392 {
393 const uint8_t *buf = avpkt->data;
394 int buf_size = avpkt->size;
395 int16_t *out_frame;
396 GetBitContext gb;
397 const G729FormatDescription *format;
398 int frame_erasure = 0; ///< frame erasure detected during decoding
399 int bad_pitch = 0; ///< parity check failed
400 int i;
401 int16_t *tmp;
402 G729Formats packet_type;
403 G729Context *ctx = avctx->priv_data;
404 int16_t lp[2][11]; // (3.12)
405 uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
406 uint8_t quantizer_1st; ///< first stage vector of quantizer
407 uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
408 uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
409
410 int pitch_delay_int[2]; // pitch delay, integer part
411 int pitch_delay_3x; // pitch delay, multiplied by 3
412 int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
413 int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
414 int j, ret;
415 int gain_before, gain_after;
416 int is_periodic = 0; // whether one of the subframes is declared as periodic or not
417 AVFrame *frame = data;
418
419 frame->nb_samples = SUBFRAME_SIZE<<1;
420 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
421 return ret;
422 out_frame = (int16_t*) frame->data[0];
423
424 if (buf_size == 10) {
425 packet_type = FORMAT_G729_8K;
426 format = &format_g729_8k;
427 //Reset voice decision
428 ctx->onset = 0;
429 ctx->voice_decision = DECISION_VOICE;
430 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
431 } else if (buf_size == 8) {
432 packet_type = FORMAT_G729D_6K4;
433 format = &format_g729d_6k4;
434 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
435 } else {
436 av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
437 return AVERROR_INVALIDDATA;
438 }
439
440 for (i=0; i < buf_size; i++)
441 frame_erasure |= buf[i];
442 frame_erasure = !frame_erasure;
443
444 init_get_bits(&gb, buf, 8*buf_size);
445
446 ma_predictor = get_bits(&gb, 1);
447 quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
448 quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
449 quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
450
451 if(frame_erasure)
452 lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
453 ctx->ma_predictor_prev);
454 else {
455 lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
456 ma_predictor,
457 quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
458 ctx->ma_predictor_prev = ma_predictor;
459 }
460
461 tmp = ctx->past_quantizer_outputs[MA_NP];
462 memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
463 MA_NP * sizeof(int16_t*));
464 ctx->past_quantizer_outputs[0] = tmp;
465
466 ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
467
468 ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
469
470 FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
471
472 for (i = 0; i < 2; i++) {
473 int gain_corr_factor;
474
475 uint8_t ac_index; ///< adaptive codebook index
476 uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
477 int fc_indexes; ///< fixed-codebook indexes
478 uint8_t gc_1st_index; ///< gain codebook (first stage) index
479 uint8_t gc_2nd_index; ///< gain codebook (second stage) index
480
481 ac_index = get_bits(&gb, format->ac_index_bits[i]);
482 if(!i && format->parity_bit)
483 bad_pitch = get_parity(ac_index) == get_bits1(&gb);
484 fc_indexes = get_bits(&gb, format->fc_indexes_bits);
485 pulses_signs = get_bits(&gb, format->fc_signs_bits);
486 gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
487 gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
488
489 if (frame_erasure)
490 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
491 else if(!i) {
492 if (bad_pitch)
493 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
494 else
495 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
496 } else {
497 int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
498 PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
499
500 if(packet_type == FORMAT_G729D_6K4)
501 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
502 else
503 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
504 }
505
506 /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
507 pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
508 if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
509 av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
510 pitch_delay_int[i] = PITCH_DELAY_MAX;
511 }
512
513 if (frame_erasure) {
514 ctx->rand_value = g729_prng(ctx->rand_value);
515 fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
516
517 ctx->rand_value = g729_prng(ctx->rand_value);
518 pulses_signs = ctx->rand_value;
519 }
520
521
522 memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
523 switch (packet_type) {
524 case FORMAT_G729_8K:
525 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
526 ff_fc_4pulses_8bits_track_4,
527 fc_indexes, pulses_signs, 3, 3);
528 break;
529 case FORMAT_G729D_6K4:
530 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
531 ff_fc_2pulses_9bits_track2_gray,
532 fc_indexes, pulses_signs, 1, 4);
533 break;
534 }
535
536 /*
537 This filter enhances harmonic components of the fixed-codebook vector to
538 improve the quality of the reconstructed speech.
539
540 / fc_v[i], i < pitch_delay
541 fc_v[i] = <
542 \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
543 */
544 ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
545 fc + pitch_delay_int[i],
546 fc, 1 << 14,
547 av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
548 0, 14,
549 SUBFRAME_SIZE - pitch_delay_int[i]);
550
551 memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
552 ctx->past_gain_code[1] = ctx->past_gain_code[0];
553
554 if (frame_erasure) {
555 ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
556 ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
557
558 gain_corr_factor = 0;
559 } else {
560 if (packet_type == FORMAT_G729D_6K4) {
561 ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
562 cb_gain_2nd_6k4[gc_2nd_index][0];
563 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
564 cb_gain_2nd_6k4[gc_2nd_index][1];
565
566 /* Without check below overflow can occur in ff_acelp_update_past_gain.
567 It is not issue for G.729, because gain_corr_factor in it's case is always
568 greater than 1024, while in G.729D it can be even zero. */
569 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
570 #ifndef G729_BITEXACT
571 gain_corr_factor >>= 1;
572 #endif
573 } else {
574 ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
575 cb_gain_2nd_8k[gc_2nd_index][0];
576 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
577 cb_gain_2nd_8k[gc_2nd_index][1];
578 }
579
580 /* Decode the fixed-codebook gain. */
581 ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->adsp, gain_corr_factor,
582 fc, MR_ENERGY,
583 ctx->quant_energy,
584 ma_prediction_coeff,
585 SUBFRAME_SIZE, 4);
586 #ifdef G729_BITEXACT
587 /*
588 This correction required to get bit-exact result with
589 reference code, because gain_corr_factor in G.729D is
590 two times larger than in original G.729.
591
592 If bit-exact result is not issue then gain_corr_factor
593 can be simpler divided by 2 before call to g729_get_gain_code
594 instead of using correction below.
595 */
596 if (packet_type == FORMAT_G729D_6K4) {
597 gain_corr_factor >>= 1;
598 ctx->past_gain_code[0] >>= 1;
599 }
600 #endif
601 }
602 ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
603
604 /* Routine requires rounding to lowest. */
605 ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
606 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
607 ff_acelp_interp_filter, 6,
608 (pitch_delay_3x % 3) << 1,
609 10, SUBFRAME_SIZE);
610
611 ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
612 ctx->exc + i * SUBFRAME_SIZE, fc,
613 (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
614 ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
615 1 << 13, 14, SUBFRAME_SIZE);
616
617 memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
618
619 if (ff_celp_lp_synthesis_filter(
620 synth+10,
621 &lp[i][1],
622 ctx->exc + i * SUBFRAME_SIZE,
623 SUBFRAME_SIZE,
624 10,
625 1,
626 0,
627 0x800))
628 /* Overflow occurred, downscale excitation signal... */
629 for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
630 ctx->exc_base[j] >>= 2;
631
632 /* ... and make synthesis again. */
633 if (packet_type == FORMAT_G729D_6K4) {
634 int16_t exc_new[SUBFRAME_SIZE];
635
636 ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
637 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
638
639 g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
640
641 ff_celp_lp_synthesis_filter(
642 synth+10,
643 &lp[i][1],
644 exc_new,
645 SUBFRAME_SIZE,
646 10,
647 0,
648 0,
649 0x800);
650 } else {
651 ff_celp_lp_synthesis_filter(
652 synth+10,
653 &lp[i][1],
654 ctx->exc + i * SUBFRAME_SIZE,
655 SUBFRAME_SIZE,
656 10,
657 0,
658 0,
659 0x800);
660 }
661 /* Save data (without postfilter) for use in next subframe. */
662 memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
663
664 /* Calculate gain of unfiltered signal for use in AGC. */
665 gain_before = 0;
666 for (j = 0; j < SUBFRAME_SIZE; j++)
667 gain_before += FFABS(synth[j+10]);
668
669 /* Call postfilter and also update voicing decision for use in next frame. */
670 ff_g729_postfilter(
671 &ctx->adsp,
672 &ctx->ht_prev_data,
673 &is_periodic,
674 &lp[i][0],
675 pitch_delay_int[0],
676 ctx->residual,
677 ctx->res_filter_data,
678 ctx->pos_filter_data,
679 synth+10,
680 SUBFRAME_SIZE);
681
682 /* Calculate gain of filtered signal for use in AGC. */
683 gain_after = 0;
684 for(j=0; j<SUBFRAME_SIZE; j++)
685 gain_after += FFABS(synth[j+10]);
686
687 ctx->gain_coeff = ff_g729_adaptive_gain_control(
688 gain_before,
689 gain_after,
690 synth+10,
691 SUBFRAME_SIZE,
692 ctx->gain_coeff);
693
694 if (frame_erasure)
695 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
696 else
697 ctx->pitch_delay_int_prev = pitch_delay_int[i];
698
699 memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
700 ff_acelp_high_pass_filter(
701 out_frame + i*SUBFRAME_SIZE,
702 ctx->hpf_f,
703 synth+10,
704 SUBFRAME_SIZE);
705 memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
706 }
707
708 ctx->was_periodic = is_periodic;
709
710 /* Save signal for use in next frame. */
711 memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
712
713 *got_frame_ptr = 1;
714 return buf_size;
715 }
716
717 AVCodec ff_g729_decoder = {
718 .name = "g729",
719 .long_name = NULL_IF_CONFIG_SMALL("G.729"),
720 .type = AVMEDIA_TYPE_AUDIO,
721 .id = AV_CODEC_ID_G729,
722 .priv_data_size = sizeof(G729Context),
723 .init = decoder_init,
724 .decode = decode_frame,
725 .capabilities = CODEC_CAP_DR1,
726 };