1/* 2 * G.723.1 compatible encoder 3 * Copyright (c) Mohamed Naufal <naufal22@gmail.com> 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/** 23 * @file 24 * G.723.1 compatible encoder 25 */ 26 27#include <stdint.h> 28#include <string.h> 29 30#include "libavutil/channel_layout.h" 31#include "libavutil/common.h" 32#include "libavutil/mem.h" 33#include "libavutil/opt.h" 34 35#include "avcodec.h" 36#include "celp_math.h" 37#include "codec_internal.h" 38#include "encode.h" 39#include "g723_1.h" 40 41#define BITSTREAM_WRITER_LE 42#include "put_bits.h" 43 44/** 45 * Hamming window coefficients scaled by 2^15 46 */ 47static const int16_t hamming_window[LPC_FRAME] = { 48 2621, 2631, 2659, 2705, 2770, 2853, 2955, 3074, 3212, 3367, 49 3541, 3731, 3939, 4164, 4405, 4663, 4937, 5226, 5531, 5851, 50 6186, 6534, 6897, 7273, 7661, 8062, 8475, 8899, 9334, 9780, 51 10235, 10699, 11172, 11653, 12141, 12636, 13138, 13645, 14157, 14673, 52 15193, 15716, 16242, 16769, 17298, 17827, 18356, 18884, 19411, 19935, 53 20457, 20975, 21489, 21999, 22503, 23002, 23494, 23978, 24455, 24924, 54 25384, 25834, 26274, 26704, 27122, 27529, 27924, 28306, 28675, 29031, 55 29373, 29700, 30012, 30310, 30592, 30857, 31107, 31340, 31557, 31756, 56 31938, 32102, 32249, 32377, 32488, 32580, 32654, 32710, 32747, 32766, 57 32766, 32747, 32710, 32654, 32580, 32488, 32377, 32249, 32102, 31938, 58 31756, 31557, 31340, 31107, 30857, 30592, 30310, 30012, 29700, 29373, 59 29031, 28675, 28306, 27924, 27529, 27122, 26704, 26274, 25834, 25384, 60 24924, 24455, 23978, 23494, 23002, 22503, 21999, 21489, 20975, 20457, 61 19935, 19411, 18884, 18356, 17827, 17298, 16769, 16242, 15716, 15193, 62 14673, 14157, 13645, 13138, 12636, 12141, 11653, 11172, 10699, 10235, 63 9780, 9334, 8899, 8475, 8062, 7661, 7273, 6897, 6534, 6186, 64 5851, 5531, 5226, 4937, 4663, 4405, 4164, 3939, 3731, 3541, 65 3367, 3212, 3074, 2955, 2853, 2770, 2705, 2659, 2631, 2621 66}; 67 68/** 69 * Binomial window coefficients scaled by 2^15 70 */ 71static const int16_t binomial_window[LPC_ORDER] = { 72 32749, 32695, 32604, 32477, 32315, 32118, 31887, 31622, 31324, 30995 73}; 74 75/** 76 * 0.994^i scaled by 2^15 77 */ 78static const int16_t bandwidth_expand[LPC_ORDER] = { 79 32571, 32376, 32182, 31989, 31797, 31606, 31416, 31228, 31040, 30854 80}; 81 82/** 83 * 0.5^i scaled by 2^15 84 */ 85static const int16_t percept_flt_tbl[2][LPC_ORDER] = { 86 /* Zero part */ 87 {29491, 26542, 23888, 21499, 19349, 17414, 15673, 14106, 12695, 11425}, 88 /* Pole part */ 89 {16384, 8192, 4096, 2048, 1024, 512, 256, 128, 64, 32} 90}; 91 92static av_cold int g723_1_encode_init(AVCodecContext *avctx) 93{ 94 G723_1_Context *s = avctx->priv_data; 95 G723_1_ChannelContext *p = &s->ch[0]; 96 97 if (avctx->sample_rate != 8000) { 98 av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n"); 99 return AVERROR(EINVAL); 100 } 101 102 if (avctx->bit_rate == 6300) { 103 p->cur_rate = RATE_6300; 104 } else if (avctx->bit_rate == 5300) { 105 av_log(avctx, AV_LOG_ERROR, "Use bitrate 6300 instead of 5300.\n"); 106 avpriv_report_missing_feature(avctx, "Bitrate 5300"); 107 return AVERROR_PATCHWELCOME; 108 } else { 109 av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\n"); 110 return AVERROR(EINVAL); 111 } 112 avctx->frame_size = 240; 113 memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t)); 114 115 return 0; 116} 117 118/** 119 * Remove DC component from the input signal. 120 * 121 * @param buf input signal 122 * @param fir zero memory 123 * @param iir pole memory 124 */ 125static void highpass_filter(int16_t *buf, int16_t *fir, int *iir) 126{ 127 int i; 128 for (i = 0; i < FRAME_LEN; i++) { 129 *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00); 130 *fir = buf[i]; 131 buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16; 132 } 133} 134 135/** 136 * Estimate autocorrelation of the input vector. 137 * 138 * @param buf input buffer 139 * @param autocorr autocorrelation coefficients vector 140 */ 141static void comp_autocorr(int16_t *buf, int16_t *autocorr) 142{ 143 int i, scale, temp; 144 int16_t vector[LPC_FRAME]; 145 146 ff_g723_1_scale_vector(vector, buf, LPC_FRAME); 147 148 /* Apply the Hamming window */ 149 for (i = 0; i < LPC_FRAME; i++) 150 vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15; 151 152 /* Compute the first autocorrelation coefficient */ 153 temp = ff_dot_product(vector, vector, LPC_FRAME); 154 155 /* Apply a white noise correlation factor of (1025/1024) */ 156 temp += temp >> 10; 157 158 /* Normalize */ 159 scale = ff_g723_1_normalize_bits(temp, 31); 160 autocorr[0] = av_clipl_int32((int64_t) (temp << scale) + 161 (1 << 15)) >> 16; 162 163 /* Compute the remaining coefficients */ 164 if (!autocorr[0]) { 165 memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t)); 166 } else { 167 for (i = 1; i <= LPC_ORDER; i++) { 168 temp = ff_dot_product(vector, vector + i, LPC_FRAME - i); 169 temp = MULL2((temp << scale), binomial_window[i - 1]); 170 autocorr[i] = av_clipl_int32((int64_t) temp + (1 << 15)) >> 16; 171 } 172 } 173} 174 175/** 176 * Use Levinson-Durbin recursion to compute LPC coefficients from 177 * autocorrelation values. 178 * 179 * @param lpc LPC coefficients vector 180 * @param autocorr autocorrelation coefficients vector 181 * @param error prediction error 182 */ 183static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error) 184{ 185 int16_t vector[LPC_ORDER]; 186 int16_t partial_corr; 187 int i, j, temp; 188 189 memset(lpc, 0, LPC_ORDER * sizeof(int16_t)); 190 191 for (i = 0; i < LPC_ORDER; i++) { 192 /* Compute the partial correlation coefficient */ 193 temp = 0; 194 for (j = 0; j < i; j++) 195 temp -= lpc[j] * autocorr[i - j - 1]; 196 temp = ((autocorr[i] << 13) + temp) << 3; 197 198 if (FFABS(temp) >= (error << 16)) 199 break; 200 201 partial_corr = temp / (error << 1); 202 203 lpc[i] = av_clipl_int32((int64_t) (partial_corr << 14) + 204 (1 << 15)) >> 16; 205 206 /* Update the prediction error */ 207 temp = MULL2(temp, partial_corr); 208 error = av_clipl_int32((int64_t) (error << 16) - temp + 209 (1 << 15)) >> 16; 210 211 memcpy(vector, lpc, i * sizeof(int16_t)); 212 for (j = 0; j < i; j++) { 213 temp = partial_corr * vector[i - j - 1] << 1; 214 lpc[j] = av_clipl_int32((int64_t) (lpc[j] << 16) - temp + 215 (1 << 15)) >> 16; 216 } 217 } 218} 219 220/** 221 * Calculate LPC coefficients for the current frame. 222 * 223 * @param buf current frame 224 * @param prev_data 2 trailing subframes of the previous frame 225 * @param lpc LPC coefficients vector 226 */ 227static void comp_lpc_coeff(int16_t *buf, int16_t *lpc) 228{ 229 int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES]; 230 int16_t *autocorr_ptr = autocorr; 231 int16_t *lpc_ptr = lpc; 232 int i, j; 233 234 for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { 235 comp_autocorr(buf + i, autocorr_ptr); 236 levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]); 237 238 lpc_ptr += LPC_ORDER; 239 autocorr_ptr += LPC_ORDER + 1; 240 } 241} 242 243static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp) 244{ 245 int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference 246 ///< polynomials (F1, F2) ordered as 247 ///< f1[0], f2[0], ...., f1[5], f2[5] 248 249 int max, shift, cur_val, prev_val, count, p; 250 int i, j; 251 int64_t temp; 252 253 /* Initialize f1[0] and f2[0] to 1 in Q25 */ 254 for (i = 0; i < LPC_ORDER; i++) 255 lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15; 256 257 /* Apply bandwidth expansion on the LPC coefficients */ 258 f[0] = f[1] = 1 << 25; 259 260 /* Compute the remaining coefficients */ 261 for (i = 0; i < LPC_ORDER / 2; i++) { 262 /* f1 */ 263 f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12); 264 /* f2 */ 265 f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12); 266 } 267 268 /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */ 269 f[LPC_ORDER] >>= 1; 270 f[LPC_ORDER + 1] >>= 1; 271 272 /* Normalize and shorten */ 273 max = FFABS(f[0]); 274 for (i = 1; i < LPC_ORDER + 2; i++) 275 max = FFMAX(max, FFABS(f[i])); 276 277 shift = ff_g723_1_normalize_bits(max, 31); 278 279 for (i = 0; i < LPC_ORDER + 2; i++) 280 f[i] = av_clipl_int32((int64_t) (f[i] << shift) + (1 << 15)) >> 16; 281 282 /** 283 * Evaluate F1 and F2 at uniform intervals of pi/256 along the 284 * unit circle and check for zero crossings. 285 */ 286 p = 0; 287 temp = 0; 288 for (i = 0; i <= LPC_ORDER / 2; i++) 289 temp += f[2 * i] * G723_1_COS_TAB_FIRST_ELEMENT; 290 prev_val = av_clipl_int32(temp << 1); 291 count = 0; 292 for (i = 1; i < COS_TBL_SIZE / 2; i++) { 293 /* Evaluate */ 294 temp = 0; 295 for (j = 0; j <= LPC_ORDER / 2; j++) 296 temp += f[LPC_ORDER - 2 * j + p] * ff_g723_1_cos_tab[i * j % COS_TBL_SIZE]; 297 cur_val = av_clipl_int32(temp << 1); 298 299 /* Check for sign change, indicating a zero crossing */ 300 if ((cur_val ^ prev_val) < 0) { 301 int abs_cur = FFABS(cur_val); 302 int abs_prev = FFABS(prev_val); 303 int sum = abs_cur + abs_prev; 304 305 shift = ff_g723_1_normalize_bits(sum, 31); 306 sum <<= shift; 307 abs_prev = abs_prev << shift >> 8; 308 lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16); 309 310 if (count == LPC_ORDER) 311 break; 312 313 /* Switch between sum and difference polynomials */ 314 p ^= 1; 315 316 /* Evaluate */ 317 temp = 0; 318 for (j = 0; j <= LPC_ORDER / 2; j++) 319 temp += f[LPC_ORDER - 2 * j + p] * 320 ff_g723_1_cos_tab[i * j % COS_TBL_SIZE]; 321 cur_val = av_clipl_int32(temp << 1); 322 } 323 prev_val = cur_val; 324 } 325 326 if (count != LPC_ORDER) 327 memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t)); 328} 329 330/** 331 * Quantize the current LSP subvector. 332 * 333 * @param num band number 334 * @param offset offset of the current subvector in an LPC_ORDER vector 335 * @param size size of the current subvector 336 */ 337#define get_index(num, offset, size) \ 338{ \ 339 int error, max = -1; \ 340 int16_t temp[4]; \ 341 int i, j; \ 342 \ 343 for (i = 0; i < LSP_CB_SIZE; i++) { \ 344 for (j = 0; j < size; j++){ \ 345 temp[j] = (weight[j + (offset)] * ff_g723_1_lsp_band##num[i][j] + \ 346 (1 << 14)) >> 15; \ 347 } \ 348 error = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1; \ 349 error -= ff_g723_1_dot_product(ff_g723_1_lsp_band##num[i], temp, size); \ 350 if (error > max) { \ 351 max = error; \ 352 lsp_index[num] = i; \ 353 } \ 354 } \ 355} 356 357/** 358 * Vector quantize the LSP frequencies. 359 * 360 * @param lsp the current lsp vector 361 * @param prev_lsp the previous lsp vector 362 */ 363static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp) 364{ 365 int16_t weight[LPC_ORDER]; 366 int16_t min, max; 367 int shift, i; 368 369 /* Calculate the VQ weighting vector */ 370 weight[0] = (1 << 20) / (lsp[1] - lsp[0]); 371 weight[LPC_ORDER - 1] = (1 << 20) / 372 (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]); 373 374 for (i = 1; i < LPC_ORDER - 1; i++) { 375 min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]); 376 if (min > 0x20) 377 weight[i] = (1 << 20) / min; 378 else 379 weight[i] = INT16_MAX; 380 } 381 382 /* Normalize */ 383 max = 0; 384 for (i = 0; i < LPC_ORDER; i++) 385 max = FFMAX(weight[i], max); 386 387 shift = ff_g723_1_normalize_bits(max, 15); 388 for (i = 0; i < LPC_ORDER; i++) { 389 weight[i] <<= shift; 390 } 391 392 /* Compute the VQ target vector */ 393 for (i = 0; i < LPC_ORDER; i++) { 394 lsp[i] -= dc_lsp[i] + 395 (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15); 396 } 397 398 get_index(0, 0, 3); 399 get_index(1, 3, 3); 400 get_index(2, 6, 4); 401} 402 403/** 404 * Perform IIR filtering. 405 * 406 * @param fir_coef FIR coefficients 407 * @param iir_coef IIR coefficients 408 * @param src source vector 409 * @param dest destination vector 410 */ 411static void iir_filter(int16_t *fir_coef, int16_t *iir_coef, 412 int16_t *src, int16_t *dest) 413{ 414 int m, n; 415 416 for (m = 0; m < SUBFRAME_LEN; m++) { 417 int64_t filter = 0; 418 for (n = 1; n <= LPC_ORDER; n++) { 419 filter -= fir_coef[n - 1] * src[m - n] - 420 iir_coef[n - 1] * dest[m - n]; 421 } 422 423 dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + 424 (1 << 15)) >> 16; 425 } 426} 427 428/** 429 * Apply the formant perceptual weighting filter. 430 * 431 * @param flt_coef filter coefficients 432 * @param unq_lpc unquantized lpc vector 433 */ 434static void perceptual_filter(G723_1_ChannelContext *p, int16_t *flt_coef, 435 int16_t *unq_lpc, int16_t *buf) 436{ 437 int16_t vector[FRAME_LEN + LPC_ORDER]; 438 int i, j, k, l = 0; 439 440 memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER); 441 memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER); 442 memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN); 443 444 for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { 445 for (k = 0; k < LPC_ORDER; k++) { 446 flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] + 447 (1 << 14)) >> 15; 448 flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] * 449 percept_flt_tbl[1][k] + 450 (1 << 14)) >> 15; 451 } 452 iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, 453 vector + i, buf + i); 454 l += LPC_ORDER; 455 } 456 memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER); 457 memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER); 458} 459 460/** 461 * Estimate the open loop pitch period. 462 * 463 * @param buf perceptually weighted speech 464 * @param start estimation is carried out from this position 465 */ 466static int estimate_pitch(int16_t *buf, int start) 467{ 468 int max_exp = 32; 469 int max_ccr = 0x4000; 470 int max_eng = 0x7fff; 471 int index = PITCH_MIN; 472 int offset = start - PITCH_MIN + 1; 473 474 int ccr, eng, orig_eng, ccr_eng, exp; 475 int diff, temp; 476 477 int i; 478 479 orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN); 480 481 for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) { 482 offset--; 483 484 /* Update energy and compute correlation */ 485 orig_eng += buf[offset] * buf[offset] - 486 buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN]; 487 ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN); 488 if (ccr <= 0) 489 continue; 490 491 /* Split into mantissa and exponent to maintain precision */ 492 exp = ff_g723_1_normalize_bits(ccr, 31); 493 ccr = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16; 494 exp <<= 1; 495 ccr *= ccr; 496 temp = ff_g723_1_normalize_bits(ccr, 31); 497 ccr = ccr << temp >> 16; 498 exp += temp; 499 500 temp = ff_g723_1_normalize_bits(orig_eng, 31); 501 eng = av_clipl_int32((int64_t) (orig_eng << temp) + (1 << 15)) >> 16; 502 exp -= temp; 503 504 if (ccr >= eng) { 505 exp--; 506 ccr >>= 1; 507 } 508 if (exp > max_exp) 509 continue; 510 511 if (exp + 1 < max_exp) 512 goto update; 513 514 /* Equalize exponents before comparison */ 515 if (exp + 1 == max_exp) 516 temp = max_ccr >> 1; 517 else 518 temp = max_ccr; 519 ccr_eng = ccr * max_eng; 520 diff = ccr_eng - eng * temp; 521 if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) { 522update: 523 index = i; 524 max_exp = exp; 525 max_ccr = ccr; 526 max_eng = eng; 527 } 528 } 529 return index; 530} 531 532/** 533 * Compute harmonic noise filter parameters. 534 * 535 * @param buf perceptually weighted speech 536 * @param pitch_lag open loop pitch period 537 * @param hf harmonic filter parameters 538 */ 539static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf) 540{ 541 int ccr, eng, max_ccr, max_eng; 542 int exp, max, diff; 543 int energy[15]; 544 int i, j; 545 546 for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) { 547 /* Compute residual energy */ 548 energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN); 549 /* Compute correlation */ 550 energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN); 551 } 552 553 /* Compute target energy */ 554 energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN); 555 556 /* Normalize */ 557 max = 0; 558 for (i = 0; i < 15; i++) 559 max = FFMAX(max, FFABS(energy[i])); 560 561 exp = ff_g723_1_normalize_bits(max, 31); 562 for (i = 0; i < 15; i++) { 563 energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) + 564 (1 << 15)) >> 16; 565 } 566 567 hf->index = -1; 568 hf->gain = 0; 569 max_ccr = 1; 570 max_eng = 0x7fff; 571 572 for (i = 0; i <= 6; i++) { 573 eng = energy[i << 1]; 574 ccr = energy[(i << 1) + 1]; 575 576 if (ccr <= 0) 577 continue; 578 579 ccr = (ccr * ccr + (1 << 14)) >> 15; 580 diff = ccr * max_eng - eng * max_ccr; 581 if (diff > 0) { 582 max_ccr = ccr; 583 max_eng = eng; 584 hf->index = i; 585 } 586 } 587 588 if (hf->index == -1) { 589 hf->index = pitch_lag; 590 return; 591 } 592 593 eng = energy[14] * max_eng; 594 eng = (eng >> 2) + (eng >> 3); 595 ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1]; 596 if (eng < ccr) { 597 eng = energy[(hf->index << 1) + 1]; 598 599 if (eng >= max_eng) 600 hf->gain = 0x2800; 601 else 602 hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15; 603 } 604 hf->index += pitch_lag - 3; 605} 606 607/** 608 * Apply the harmonic noise shaping filter. 609 * 610 * @param hf filter parameters 611 */ 612static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest) 613{ 614 int i; 615 616 for (i = 0; i < SUBFRAME_LEN; i++) { 617 int64_t temp = hf->gain * src[i - hf->index] << 1; 618 dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16; 619 } 620} 621 622static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest) 623{ 624 int i; 625 for (i = 0; i < SUBFRAME_LEN; i++) { 626 int64_t temp = hf->gain * src[i - hf->index] << 1; 627 dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp + 628 (1 << 15)) >> 16; 629 } 630} 631 632/** 633 * Combined synthesis and formant perceptual weighting filer. 634 * 635 * @param qnt_lpc quantized lpc coefficients 636 * @param perf_lpc perceptual filter coefficients 637 * @param perf_fir perceptual filter fir memory 638 * @param perf_iir perceptual filter iir memory 639 * @param scale the filter output will be scaled by 2^scale 640 */ 641static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc, 642 int16_t *perf_fir, int16_t *perf_iir, 643 const int16_t *src, int16_t *dest, int scale) 644{ 645 int i, j; 646 int16_t buf_16[SUBFRAME_LEN + LPC_ORDER]; 647 int64_t buf[SUBFRAME_LEN]; 648 649 int16_t *bptr_16 = buf_16 + LPC_ORDER; 650 651 memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER); 652 memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER); 653 654 for (i = 0; i < SUBFRAME_LEN; i++) { 655 int64_t temp = 0; 656 for (j = 1; j <= LPC_ORDER; j++) 657 temp -= qnt_lpc[j - 1] * bptr_16[i - j]; 658 659 buf[i] = (src[i] << 15) + (temp << 3); 660 bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16; 661 } 662 663 for (i = 0; i < SUBFRAME_LEN; i++) { 664 int64_t fir = 0, iir = 0; 665 for (j = 1; j <= LPC_ORDER; j++) { 666 fir -= perf_lpc[j - 1] * bptr_16[i - j]; 667 iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j]; 668 } 669 dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) + 670 (1 << 15)) >> 16; 671 } 672 memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER); 673 memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER, 674 sizeof(int16_t) * LPC_ORDER); 675} 676 677/** 678 * Compute the adaptive codebook contribution. 679 * 680 * @param buf input signal 681 * @param index the current subframe index 682 */ 683static void acb_search(G723_1_ChannelContext *p, int16_t *residual, 684 int16_t *impulse_resp, const int16_t *buf, 685 int index) 686{ 687 int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN]; 688 689 const int16_t *cb_tbl = ff_g723_1_adaptive_cb_gain85; 690 691 int ccr_buf[PITCH_ORDER * SUBFRAMES << 2]; 692 693 int pitch_lag = p->pitch_lag[index >> 1]; 694 int acb_lag = 1; 695 int acb_gain = 0; 696 int odd_frame = index & 1; 697 int iter = 3 + odd_frame; 698 int count = 0; 699 int tbl_size = 85; 700 701 int i, j, k, l, max; 702 int64_t temp; 703 704 if (!odd_frame) { 705 if (pitch_lag == PITCH_MIN) 706 pitch_lag++; 707 else 708 pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5); 709 } 710 711 for (i = 0; i < iter; i++) { 712 ff_g723_1_get_residual(residual, p->prev_excitation, pitch_lag + i - 1); 713 714 for (j = 0; j < SUBFRAME_LEN; j++) { 715 temp = 0; 716 for (k = 0; k <= j; k++) 717 temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k]; 718 flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) + 719 (1 << 15)) >> 16; 720 } 721 722 for (j = PITCH_ORDER - 2; j >= 0; j--) { 723 flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15; 724 for (k = 1; k < SUBFRAME_LEN; k++) { 725 temp = (flt_buf[j + 1][k - 1] << 15) + 726 residual[j] * impulse_resp[k]; 727 flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16; 728 } 729 } 730 731 /* Compute crosscorrelation with the signal */ 732 for (j = 0; j < PITCH_ORDER; j++) { 733 temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN); 734 ccr_buf[count++] = av_clipl_int32(temp << 1); 735 } 736 737 /* Compute energies */ 738 for (j = 0; j < PITCH_ORDER; j++) { 739 ccr_buf[count++] = ff_g723_1_dot_product(flt_buf[j], flt_buf[j], 740 SUBFRAME_LEN); 741 } 742 743 for (j = 1; j < PITCH_ORDER; j++) { 744 for (k = 0; k < j; k++) { 745 temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN); 746 ccr_buf[count++] = av_clipl_int32(temp << 2); 747 } 748 } 749 } 750 751 /* Normalize and shorten */ 752 max = 0; 753 for (i = 0; i < 20 * iter; i++) 754 max = FFMAX(max, FFABS(ccr_buf[i])); 755 756 temp = ff_g723_1_normalize_bits(max, 31); 757 758 for (i = 0; i < 20 * iter; i++) 759 ccr_buf[i] = av_clipl_int32((int64_t) (ccr_buf[i] << temp) + 760 (1 << 15)) >> 16; 761 762 max = 0; 763 for (i = 0; i < iter; i++) { 764 /* Select quantization table */ 765 if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 || 766 odd_frame && pitch_lag >= SUBFRAME_LEN - 2) { 767 cb_tbl = ff_g723_1_adaptive_cb_gain170; 768 tbl_size = 170; 769 } 770 771 for (j = 0, k = 0; j < tbl_size; j++, k += 20) { 772 temp = 0; 773 for (l = 0; l < 20; l++) 774 temp += ccr_buf[20 * i + l] * cb_tbl[k + l]; 775 temp = av_clipl_int32(temp); 776 777 if (temp > max) { 778 max = temp; 779 acb_gain = j; 780 acb_lag = i; 781 } 782 } 783 } 784 785 if (!odd_frame) { 786 pitch_lag += acb_lag - 1; 787 acb_lag = 1; 788 } 789 790 p->pitch_lag[index >> 1] = pitch_lag; 791 p->subframe[index].ad_cb_lag = acb_lag; 792 p->subframe[index].ad_cb_gain = acb_gain; 793} 794 795/** 796 * Subtract the adaptive codebook contribution from the input 797 * to obtain the residual. 798 * 799 * @param buf target vector 800 */ 801static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp, 802 int16_t *buf) 803{ 804 int i, j; 805 /* Subtract adaptive CB contribution to obtain the residual */ 806 for (i = 0; i < SUBFRAME_LEN; i++) { 807 int64_t temp = buf[i] << 14; 808 for (j = 0; j <= i; j++) 809 temp -= residual[j] * impulse_resp[i - j]; 810 811 buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16; 812 } 813} 814 815/** 816 * Quantize the residual signal using the fixed codebook (MP-MLQ). 817 * 818 * @param optim optimized fixed codebook parameters 819 * @param buf excitation vector 820 */ 821static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp, 822 int16_t *buf, int pulse_cnt, int pitch_lag) 823{ 824 FCBParam param; 825 int16_t impulse_r[SUBFRAME_LEN]; 826 int16_t temp_corr[SUBFRAME_LEN]; 827 int16_t impulse_corr[SUBFRAME_LEN]; 828 829 int ccr1[SUBFRAME_LEN]; 830 int ccr2[SUBFRAME_LEN]; 831 int amp, err, max, max_amp_index, min, scale, i, j, k, l; 832 833 int64_t temp; 834 835 /* Update impulse response */ 836 memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN); 837 param.dirac_train = 0; 838 if (pitch_lag < SUBFRAME_LEN - 2) { 839 param.dirac_train = 1; 840 ff_g723_1_gen_dirac_train(impulse_r, pitch_lag); 841 } 842 843 for (i = 0; i < SUBFRAME_LEN; i++) 844 temp_corr[i] = impulse_r[i] >> 1; 845 846 /* Compute impulse response autocorrelation */ 847 temp = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN); 848 849 scale = ff_g723_1_normalize_bits(temp, 31); 850 impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16; 851 852 for (i = 1; i < SUBFRAME_LEN; i++) { 853 temp = ff_g723_1_dot_product(temp_corr + i, temp_corr, 854 SUBFRAME_LEN - i); 855 impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16; 856 } 857 858 /* Compute crosscorrelation of impulse response with residual signal */ 859 scale -= 4; 860 for (i = 0; i < SUBFRAME_LEN; i++) { 861 temp = ff_g723_1_dot_product(buf + i, impulse_r, SUBFRAME_LEN - i); 862 if (scale < 0) 863 ccr1[i] = temp >> -scale; 864 else 865 ccr1[i] = av_clipl_int32(temp << scale); 866 } 867 868 /* Search loop */ 869 for (i = 0; i < GRID_SIZE; i++) { 870 /* Maximize the crosscorrelation */ 871 max = 0; 872 for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) { 873 temp = FFABS(ccr1[j]); 874 if (temp >= max) { 875 max = temp; 876 param.pulse_pos[0] = j; 877 } 878 } 879 880 /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */ 881 amp = max; 882 min = 1 << 30; 883 max_amp_index = GAIN_LEVELS - 2; 884 for (j = max_amp_index; j >= 2; j--) { 885 temp = av_clipl_int32((int64_t) ff_g723_1_fixed_cb_gain[j] * 886 impulse_corr[0] << 1); 887 temp = FFABS(temp - amp); 888 if (temp < min) { 889 min = temp; 890 max_amp_index = j; 891 } 892 } 893 894 max_amp_index--; 895 /* Select additional gain values */ 896 for (j = 1; j < 5; j++) { 897 for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) { 898 temp_corr[k] = 0; 899 ccr2[k] = ccr1[k]; 900 } 901 param.amp_index = max_amp_index + j - 2; 902 amp = ff_g723_1_fixed_cb_gain[param.amp_index]; 903 904 param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp; 905 temp_corr[param.pulse_pos[0]] = 1; 906 907 for (k = 1; k < pulse_cnt; k++) { 908 max = INT_MIN; 909 for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) { 910 if (temp_corr[l]) 911 continue; 912 temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])]; 913 temp = av_clipl_int32((int64_t) temp * 914 param.pulse_sign[k - 1] << 1); 915 ccr2[l] -= temp; 916 temp = FFABS(ccr2[l]); 917 if (temp > max) { 918 max = temp; 919 param.pulse_pos[k] = l; 920 } 921 } 922 923 param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ? 924 -amp : amp; 925 temp_corr[param.pulse_pos[k]] = 1; 926 } 927 928 /* Create the error vector */ 929 memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN); 930 931 for (k = 0; k < pulse_cnt; k++) 932 temp_corr[param.pulse_pos[k]] = param.pulse_sign[k]; 933 934 for (k = SUBFRAME_LEN - 1; k >= 0; k--) { 935 temp = 0; 936 for (l = 0; l <= k; l++) { 937 int prod = av_clipl_int32((int64_t) temp_corr[l] * 938 impulse_r[k - l] << 1); 939 temp = av_clipl_int32(temp + prod); 940 } 941 temp_corr[k] = temp << 2 >> 16; 942 } 943 944 /* Compute square of error */ 945 err = 0; 946 for (k = 0; k < SUBFRAME_LEN; k++) { 947 int64_t prod; 948 prod = av_clipl_int32((int64_t) buf[k] * temp_corr[k] << 1); 949 err = av_clipl_int32(err - prod); 950 prod = av_clipl_int32((int64_t) temp_corr[k] * temp_corr[k]); 951 err = av_clipl_int32(err + prod); 952 } 953 954 /* Minimize */ 955 if (err < optim->min_err) { 956 optim->min_err = err; 957 optim->grid_index = i; 958 optim->amp_index = param.amp_index; 959 optim->dirac_train = param.dirac_train; 960 961 for (k = 0; k < pulse_cnt; k++) { 962 optim->pulse_sign[k] = param.pulse_sign[k]; 963 optim->pulse_pos[k] = param.pulse_pos[k]; 964 } 965 } 966 } 967 } 968} 969 970/** 971 * Encode the pulse position and gain of the current subframe. 972 * 973 * @param optim optimized fixed CB parameters 974 * @param buf excitation vector 975 */ 976static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim, 977 int16_t *buf, int pulse_cnt) 978{ 979 int i, j; 980 981 j = PULSE_MAX - pulse_cnt; 982 983 subfrm->pulse_sign = 0; 984 subfrm->pulse_pos = 0; 985 986 for (i = 0; i < SUBFRAME_LEN >> 1; i++) { 987 int val = buf[optim->grid_index + (i << 1)]; 988 if (!val) { 989 subfrm->pulse_pos += ff_g723_1_combinatorial_table[j][i]; 990 } else { 991 subfrm->pulse_sign <<= 1; 992 if (val < 0) 993 subfrm->pulse_sign++; 994 j++; 995 996 if (j == PULSE_MAX) 997 break; 998 } 999 } 1000 subfrm->amp_index = optim->amp_index; 1001 subfrm->grid_index = optim->grid_index; 1002 subfrm->dirac_train = optim->dirac_train; 1003} 1004 1005/** 1006 * Compute the fixed codebook excitation. 1007 * 1008 * @param buf target vector 1009 * @param impulse_resp impulse response of the combined filter 1010 */ 1011static void fcb_search(G723_1_ChannelContext *p, int16_t *impulse_resp, 1012 int16_t *buf, int index) 1013{ 1014 FCBParam optim; 1015 int pulse_cnt = pulses[index]; 1016 int i; 1017 1018 optim.min_err = 1 << 30; 1019 get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN); 1020 1021 if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) { 1022 get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, 1023 p->pitch_lag[index >> 1]); 1024 } 1025 1026 /* Reconstruct the excitation */ 1027 memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN); 1028 for (i = 0; i < pulse_cnt; i++) 1029 buf[optim.pulse_pos[i]] = optim.pulse_sign[i]; 1030 1031 pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt); 1032 1033 if (optim.dirac_train) 1034 ff_g723_1_gen_dirac_train(buf, p->pitch_lag[index >> 1]); 1035} 1036 1037/** 1038 * Pack the frame parameters into output bitstream. 1039 * 1040 * @param frame output buffer 1041 * @param size size of the buffer 1042 */ 1043static void pack_bitstream(G723_1_ChannelContext *p, AVPacket *avpkt, int info_bits) 1044{ 1045 PutBitContext pb; 1046 int i, temp; 1047 1048 init_put_bits(&pb, avpkt->data, avpkt->size); 1049 1050 put_bits(&pb, 2, info_bits); 1051 1052 put_bits(&pb, 8, p->lsp_index[2]); 1053 put_bits(&pb, 8, p->lsp_index[1]); 1054 put_bits(&pb, 8, p->lsp_index[0]); 1055 1056 put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN); 1057 put_bits(&pb, 2, p->subframe[1].ad_cb_lag); 1058 put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN); 1059 put_bits(&pb, 2, p->subframe[3].ad_cb_lag); 1060 1061 /* Write 12 bit combined gain */ 1062 for (i = 0; i < SUBFRAMES; i++) { 1063 temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS + 1064 p->subframe[i].amp_index; 1065 if (p->cur_rate == RATE_6300) 1066 temp += p->subframe[i].dirac_train << 11; 1067 put_bits(&pb, 12, temp); 1068 } 1069 1070 put_bits(&pb, 1, p->subframe[0].grid_index); 1071 put_bits(&pb, 1, p->subframe[1].grid_index); 1072 put_bits(&pb, 1, p->subframe[2].grid_index); 1073 put_bits(&pb, 1, p->subframe[3].grid_index); 1074 1075 if (p->cur_rate == RATE_6300) { 1076 put_bits(&pb, 1, 0); /* reserved bit */ 1077 1078 /* Write 13 bit combined position index */ 1079 temp = (p->subframe[0].pulse_pos >> 16) * 810 + 1080 (p->subframe[1].pulse_pos >> 14) * 90 + 1081 (p->subframe[2].pulse_pos >> 16) * 9 + 1082 (p->subframe[3].pulse_pos >> 14); 1083 put_bits(&pb, 13, temp); 1084 1085 put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff); 1086 put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff); 1087 put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff); 1088 put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff); 1089 1090 put_bits(&pb, 6, p->subframe[0].pulse_sign); 1091 put_bits(&pb, 5, p->subframe[1].pulse_sign); 1092 put_bits(&pb, 6, p->subframe[2].pulse_sign); 1093 put_bits(&pb, 5, p->subframe[3].pulse_sign); 1094 } 1095 1096 flush_put_bits(&pb); 1097} 1098 1099static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, 1100 const AVFrame *frame, int *got_packet_ptr) 1101{ 1102 G723_1_Context *s = avctx->priv_data; 1103 G723_1_ChannelContext *p = &s->ch[0]; 1104 int16_t unq_lpc[LPC_ORDER * SUBFRAMES]; 1105 int16_t qnt_lpc[LPC_ORDER * SUBFRAMES]; 1106 int16_t cur_lsp[LPC_ORDER]; 1107 int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1]; 1108 int16_t vector[FRAME_LEN + PITCH_MAX]; 1109 int offset, ret, i, j, info_bits = 0; 1110 int16_t *in, *start; 1111 HFParam hf[4]; 1112 1113 /* duplicate input */ 1114 start = in = av_memdup(frame->data[0], frame->nb_samples * sizeof(int16_t)); 1115 if (!in) 1116 return AVERROR(ENOMEM); 1117 1118 highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem); 1119 1120 memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t)); 1121 memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t)); 1122 1123 comp_lpc_coeff(vector, unq_lpc); 1124 lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp); 1125 lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp); 1126 1127 /* Update memory */ 1128 memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN, 1129 sizeof(int16_t) * SUBFRAME_LEN); 1130 memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in, 1131 sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN)); 1132 memcpy(p->prev_data, in + HALF_FRAME_LEN, 1133 sizeof(int16_t) * HALF_FRAME_LEN); 1134 memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN); 1135 1136 perceptual_filter(p, weighted_lpc, unq_lpc, vector); 1137 1138 memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN); 1139 memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX); 1140 memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN); 1141 1142 ff_g723_1_scale_vector(vector, vector, FRAME_LEN + PITCH_MAX); 1143 1144 p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX); 1145 p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN); 1146 1147 for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) 1148 comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j); 1149 1150 memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX); 1151 memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN); 1152 memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX); 1153 1154 for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) 1155 harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i); 1156 1157 ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0); 1158 ff_g723_1_lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp); 1159 1160 memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER); 1161 1162 offset = 0; 1163 for (i = 0; i < SUBFRAMES; i++) { 1164 int16_t impulse_resp[SUBFRAME_LEN]; 1165 int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1]; 1166 int16_t flt_in[SUBFRAME_LEN]; 1167 int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER]; 1168 1169 /** 1170 * Compute the combined impulse response of the synthesis filter, 1171 * formant perceptual weighting filter and harmonic noise shaping filter 1172 */ 1173 memset(zero, 0, sizeof(int16_t) * LPC_ORDER); 1174 memset(vector, 0, sizeof(int16_t) * PITCH_MAX); 1175 memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN); 1176 1177 flt_in[0] = 1 << 13; /* Unit impulse */ 1178 synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1), 1179 zero, zero, flt_in, vector + PITCH_MAX, 1); 1180 harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp); 1181 1182 /* Compute the combined zero input response */ 1183 flt_in[0] = 0; 1184 memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER); 1185 memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER); 1186 1187 synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1), 1188 fir, iir, flt_in, vector + PITCH_MAX, 0); 1189 memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX); 1190 harmonic_noise_sub(hf + i, vector + PITCH_MAX, in); 1191 1192 acb_search(p, residual, impulse_resp, in, i); 1193 ff_g723_1_gen_acb_excitation(residual, p->prev_excitation, 1194 p->pitch_lag[i >> 1], &p->subframe[i], 1195 p->cur_rate); 1196 sub_acb_contrib(residual, impulse_resp, in); 1197 1198 fcb_search(p, impulse_resp, in, i); 1199 1200 /* Reconstruct the excitation */ 1201 ff_g723_1_gen_acb_excitation(impulse_resp, p->prev_excitation, 1202 p->pitch_lag[i >> 1], &p->subframe[i], 1203 RATE_6300); 1204 1205 memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN, 1206 sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN)); 1207 for (j = 0; j < SUBFRAME_LEN; j++) 1208 in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]); 1209 memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in, 1210 sizeof(int16_t) * SUBFRAME_LEN); 1211 1212 /* Update filter memories */ 1213 synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1), 1214 p->perf_fir_mem, p->perf_iir_mem, 1215 in, vector + PITCH_MAX, 0); 1216 memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN, 1217 sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN)); 1218 memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX, 1219 sizeof(int16_t) * SUBFRAME_LEN); 1220 1221 in += SUBFRAME_LEN; 1222 offset += LPC_ORDER; 1223 } 1224 1225 av_free(start); 1226 1227 ret = ff_get_encode_buffer(avctx, avpkt, frame_size[info_bits], 0); 1228 if (ret < 0) 1229 return ret; 1230 1231 *got_packet_ptr = 1; 1232 pack_bitstream(p, avpkt, info_bits); 1233 return 0; 1234} 1235 1236static const FFCodecDefault defaults[] = { 1237 { "b", "6300" }, 1238 { NULL }, 1239}; 1240 1241const FFCodec ff_g723_1_encoder = { 1242 .p.name = "g723_1", 1243 .p.long_name = NULL_IF_CONFIG_SMALL("G.723.1"), 1244 .p.type = AVMEDIA_TYPE_AUDIO, 1245 .p.id = AV_CODEC_ID_G723_1, 1246 .p.capabilities = AV_CODEC_CAP_DR1, 1247 .priv_data_size = sizeof(G723_1_Context), 1248 .init = g723_1_encode_init, 1249 FF_CODEC_ENCODE_CB(g723_1_encode_frame), 1250 .defaults = defaults, 1251 .p.sample_fmts = (const enum AVSampleFormat[]) { 1252 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE 1253 }, 1254 .p.ch_layouts = (const AVChannelLayout[]){ 1255 AV_CHANNEL_LAYOUT_MONO, { 0 } 1256 }, 1257 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, 1258}; 1259