summaryrefslogtreecommitdiff
path: root/libavcodec/g729dec.c (plain)
blob: 2e1bf18e4e49f40dc501938ee05172beac53f94d
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
87typedef enum {
88 FORMAT_G729_8K = 0,
89 FORMAT_G729D_6K4,
90 FORMAT_COUNT,
91} G729Formats;
92
93typedef 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
102typedef 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
157static 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
166static 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 */
178static 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 */
192static 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
223 ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
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 */
233static 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 */
258static 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 */
286static 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 */
302static 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
329static 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
339static av_cold int decoder_init(AVCodecContext * avctx)
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++) {
356 ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[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
365 ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
366
367 ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
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);
377 ctx->adsp.scalarproduct_int16 = scalarproduct_int16_c;
378
379 return 0;
380}
381
382static 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;
389 const G729FormatDescription *format;
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;
421 ctx->voice_decision = DECISION_VOICE;
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)
444 lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
445 ctx->ma_predictor_prev);
446 else {
447 lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
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,
490 PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
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:
517 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
518 ff_fc_4pulses_8bits_track_4,
519 fc_indexes, pulses_signs, 3, 3);
520 break;
521 case FORMAT_G729D_6K4:
522 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
523 ff_fc_2pulses_9bits_track2_gray,
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,
576 ma_prediction_coeff,
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,
599 ff_acelp_interp_filter, 6,
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
611 if (ff_celp_lp_synthesis_filter(
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);
629 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
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
633 ff_celp_lp_synthesis_filter(
634 synth+10,
635 &lp[i][1],
636 exc_new,
637 SUBFRAME_SIZE,
638 10,
639 0,
640 0,
641 0x800);
642 } else {
643 ff_celp_lp_synthesis_filter(
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. */
662 ff_g729_postfilter(
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
679 ctx->gain_coeff = ff_g729_adaptive_gain_control(
680 gain_before,
681 gain_after,
682 synth+10,
683 SUBFRAME_SIZE,
684 ctx->gain_coeff);
685
686 if (frame_erasure)
687 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
688 else
689 ctx->pitch_delay_int_prev = pitch_delay_int[i];
690
691 memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
692 ff_acelp_high_pass_filter(
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
709AVCodec ff_g729_decoder = {
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};
719
generated by cgit at 2024-05-17 05:00:09 (GMT)