FFmpeg: libavcodec/g723_1.c Source File

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g723_1.c

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1 /*

2 * G.723.1 compatible decoder

5 *

6 * This file is part of FFmpeg.

7 *

8 * FFmpeg is free software; you can redistribute it and/or

9 * modify it under the terms of the GNU Lesser General Public

10 * License as published by the Free Software Foundation; either

11 * version 2.1 of the License, or (at your option) any later version.

12 *

13 * FFmpeg is distributed in the hope that it will be useful,

14 * but WITHOUT ANY WARRANTY; without even the implied warranty of

15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

16 * Lesser General Public License for more details.

17 *

18 * You should have received a copy of the GNU Lesser General Public

19 * License along with FFmpeg; if not, write to the Free Software

20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

21 */

23 /**

24 * @file

25 * G.723.1 compatible decoder

26 */

28 #define BITSTREAM_READER_LE

29 #include "libavutil/channel_layout.h"

30 #include "libavutil/mem.h"

31 #include "libavutil/opt.h"

32 #include "avcodec.h"

33 #include "get_bits.h"

34 #include "acelp_vectors.h"

35 #include "celp_filters.h"

36 #include "celp_math.h"

37 #include "g723_1_data.h"

38 #include "internal.h"

40 #define CNG_RANDOM_SEED 12345

42 typedef struct g723_1_context {

43 AVClass *class;

45 G723_1_Subframe subframe[4];

46 enum FrameType cur_frame_type;

47 enum FrameType past_frame_type;

48 enum Rate cur_rate;

49 uint8_t lsp_index[LSP_BANDS];

50 int pitch_lag[2];

51 int erased_frames;

53 int16_t prev_lsp[LPC_ORDER];

54 int16_t sid_lsp[LPC_ORDER];

55 int16_t prev_excitation[PITCH_MAX];

56 int16_t excitation[PITCH_MAX + FRAME_LEN + 4];

57 int16_t synth_mem[LPC_ORDER];

58 int16_t fir_mem[LPC_ORDER];

59 int iir_mem[LPC_ORDER];

61 int random_seed;

62 int cng_random_seed;

63 int interp_index;

64 int interp_gain;

65 int sid_gain;

66 int cur_gain;

67 int reflection_coef;

68 int pf_gain; ///< formant postfilter

69 ///< gain scaling unit memory

70 int postfilter;

72 int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];

73 int16_t prev_data[HALF_FRAME_LEN];

74 int16_t prev_weight_sig[PITCH_MAX];

77 int16_t hpf_fir_mem; ///< highpass filter fir

78 int hpf_iir_mem; ///< and iir memories

79 int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir

80 int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories

82 int16_t harmonic_mem[PITCH_MAX];

83 } G723_1_Context;

85 static av_cold int g723_1_decode_init(AVCodecContext *avctx)

86 {

87 G723_1_Context *p = avctx->priv_data;

89 avctx->channel_layout = AV_CH_LAYOUT_MONO;

90 avctx->sample_fmt = AV_SAMPLE_FMT_S16;

91 avctx->channels = 1;

92 p->pf_gain = 1 << 12;

94 memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));

95 memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));

97 p->cng_random_seed = CNG_RANDOM_SEED;

98 p->past_frame_type = SID_FRAME;

100 return 0;

101 }

102

103 /**

104 * Unpack the frame into parameters.

105 *

106 * @param p the context

107 * @param buf pointer to the input buffer

108 * @param buf_size size of the input buffer

109 */

110 static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,

111 int buf_size)

112 {

113 GetBitContext gb;

114 int ad_cb_len;

115 int temp, info_bits, i;

116

117 init_get_bits(&gb, buf, buf_size * 8);

118

119 /* Extract frame type and rate info */

120 info_bits = get_bits(&gb, 2);

121

122 if (info_bits == 3) {

123 p->cur_frame_type = UNTRANSMITTED_FRAME;

124 return 0;

125 }

126

127 /* Extract 24 bit lsp indices, 8 bit for each band */

128 p->lsp_index[2] = get_bits(&gb, 8);

129 p->lsp_index[1] = get_bits(&gb, 8);

130 p->lsp_index[0] = get_bits(&gb, 8);

131

132 if (info_bits == 2) {

133 p->cur_frame_type = SID_FRAME;

134 p->subframe[0].amp_index = get_bits(&gb, 6);

135 return 0;

136 }

137

138 /* Extract the info common to both rates */

139 p->cur_rate = info_bits ? RATE_5300 : RATE_6300;

140 p->cur_frame_type = ACTIVE_FRAME;

141

142 p->pitch_lag[0] = get_bits(&gb, 7);

143 if (p->pitch_lag[0] > 123) /* test if forbidden code */

144 return -1;

145 p->pitch_lag[0] += PITCH_MIN;

146 p->subframe[1].ad_cb_lag = get_bits(&gb, 2);

147

148 p->pitch_lag[1] = get_bits(&gb, 7);

149 if (p->pitch_lag[1] > 123)

150 return -1;

151 p->pitch_lag[1] += PITCH_MIN;

152 p->subframe[3].ad_cb_lag = get_bits(&gb, 2);

153 p->subframe[0].ad_cb_lag = 1;

154 p->subframe[2].ad_cb_lag = 1;

155

156 for (i = 0; i < SUBFRAMES; i++) {

157 /* Extract combined gain */

158 temp = get_bits(&gb, 12);

159 ad_cb_len = 170;

160 p->subframe[i].dirac_train = 0;

161 if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {

162 p->subframe[i].dirac_train = temp >> 11;

163 temp &= 0x7FF;

164 ad_cb_len = 85;

165 }

166 p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);

167 if (p->subframe[i].ad_cb_gain < ad_cb_len) {

168 p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *

169 GAIN_LEVELS;

170 } else {

171 return -1;

172 }

173 }

174

175 p->subframe[0].grid_index = get_bits1(&gb);

176 p->subframe[1].grid_index = get_bits1(&gb);

177 p->subframe[2].grid_index = get_bits1(&gb);

178 p->subframe[3].grid_index = get_bits1(&gb);

179

180 if (p->cur_rate == RATE_6300) {

181 skip_bits1(&gb); /* skip reserved bit */

182

183 /* Compute pulse_pos index using the 13-bit combined position index */

184 temp = get_bits(&gb, 13);

185 p->subframe[0].pulse_pos = temp / 810;

186

187 temp -= p->subframe[0].pulse_pos * 810;

188 p->subframe[1].pulse_pos = FASTDIV(temp, 90);

189

190 temp -= p->subframe[1].pulse_pos * 90;

191 p->subframe[2].pulse_pos = FASTDIV(temp, 9);

192 p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;

193

194 p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +

195 get_bits(&gb, 16);

196 p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +

197 get_bits(&gb, 14);

198 p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +

199 get_bits(&gb, 16);

200 p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +

201 get_bits(&gb, 14);

202

203 p->subframe[0].pulse_sign = get_bits(&gb, 6);

204 p->subframe[1].pulse_sign = get_bits(&gb, 5);

205 p->subframe[2].pulse_sign = get_bits(&gb, 6);

206 p->subframe[3].pulse_sign = get_bits(&gb, 5);

207 } else { /* 5300 bps */

208 p->subframe[0].pulse_pos = get_bits(&gb, 12);

209 p->subframe[1].pulse_pos = get_bits(&gb, 12);

210 p->subframe[2].pulse_pos = get_bits(&gb, 12);

211 p->subframe[3].pulse_pos = get_bits(&gb, 12);

212

213 p->subframe[0].pulse_sign = get_bits(&gb, 4);

214 p->subframe[1].pulse_sign = get_bits(&gb, 4);

215 p->subframe[2].pulse_sign = get_bits(&gb, 4);

216 p->subframe[3].pulse_sign = get_bits(&gb, 4);

217 }

218

219 return 0;

220 }

221

222 /**

223 * Bitexact implementation of sqrt(val/2).

224 */

225 static int16_t square_root(unsigned val)

226 {

227 av_assert2(!(val & 0x80000000));

228

229 return (ff_sqrt(val << 1) >> 1) & (~1);

230 }

231

232 /**

233 * Calculate the number of left-shifts required for normalizing the input.

234 *

235 * @param num input number

236 * @param width width of the input, 15 or 31 bits

237 */

238 static int normalize_bits(int num, int width)

239 {

240 return width - av_log2(num) - 1;

241 }

242

243 #define normalize_bits_int16(num) normalize_bits(num, 15)

244 #define normalize_bits_int32(num) normalize_bits(num, 31)

245

246 /**

247 * Scale vector contents based on the largest of their absolutes.

248 */

249 static int scale_vector(int16_t *dst, const int16_t *vector, int length)

250 {

251 int bits, max = 0;

252 int i;

253

254 for (i = 0; i < length; i++)

255 max |= FFABS(vector[i]);

256

257 bits= 14 - av_log2_16bit(max);

258 bits= FFMAX(bits, 0);

259

260 for (i = 0; i < length; i++)

261 dst[i] = vector[i] << bits >> 3;

262

263 return bits - 3;

264 }

265

266 /**

267 * Perform inverse quantization of LSP frequencies.

268 *

269 * @param cur_lsp the current LSP vector

270 * @param prev_lsp the previous LSP vector

271 * @param lsp_index VQ indices

272 * @param bad_frame bad frame flag

273 */

274 static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,

275 uint8_t *lsp_index, int bad_frame)

276 {

277 int min_dist, pred;

278 int i, j, temp, stable;

279

280 /* Check for frame erasure */

281 if (!bad_frame) {

282 min_dist = 0x100;

283 pred = 12288;

284 } else {

285 min_dist = 0x200;

286 pred = 23552;

287 lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;

288 }

289

290 /* Get the VQ table entry corresponding to the transmitted index */

291 cur_lsp[0] = lsp_band0[lsp_index[0]][0];

292 cur_lsp[1] = lsp_band0[lsp_index[0]][1];

293 cur_lsp[2] = lsp_band0[lsp_index[0]][2];

294 cur_lsp[3] = lsp_band1[lsp_index[1]][0];

295 cur_lsp[4] = lsp_band1[lsp_index[1]][1];

296 cur_lsp[5] = lsp_band1[lsp_index[1]][2];

297 cur_lsp[6] = lsp_band2[lsp_index[2]][0];

298 cur_lsp[7] = lsp_band2[lsp_index[2]][1];

299 cur_lsp[8] = lsp_band2[lsp_index[2]][2];

300 cur_lsp[9] = lsp_band2[lsp_index[2]][3];

301

302 /* Add predicted vector & DC component to the previously quantized vector */

303 for (i = 0; i < LPC_ORDER; i++) {

304 temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;

305 cur_lsp[i] += dc_lsp[i] + temp;

306 }

307

308 for (i = 0; i < LPC_ORDER; i++) {

309 cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);

310 cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);

311

312 /* Stability check */

313 for (j = 1; j < LPC_ORDER; j++) {

314 temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];

315 if (temp > 0) {

316 temp >>= 1;

317 cur_lsp[j - 1] -= temp;

318 cur_lsp[j] += temp;

319 }

320 }

321 stable = 1;

322 for (j = 1; j < LPC_ORDER; j++) {

323 temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;

324 if (temp > 0) {

325 stable = 0;

326 break;

327 }

328 }

329 if (stable)

330 break;

331 }

332 if (!stable)

333 memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));

334 }

335

336 /**

337 * Bitexact implementation of 2ab scaled by 1/2^16.

338 *

339 * @param a 32 bit multiplicand

340 * @param b 16 bit multiplier

341 */

342 #define MULL2(a, b) \

343 MULL(a,b,15)

344

345 /**

346 * Convert LSP frequencies to LPC coefficients.

347 *

348 * @param lpc buffer for LPC coefficients

349 */

350 static void lsp2lpc(int16_t *lpc)

351 {

352 int f1[LPC_ORDER / 2 + 1];

353 int f2[LPC_ORDER / 2 + 1];

354 int i, j;

355

356 /* Calculate negative cosine */

357 for (j = 0; j < LPC_ORDER; j++) {

358 int index = (lpc[j] >> 7) & 0x1FF;

359 int offset = lpc[j] & 0x7f;

360 int temp1 = cos_tab[index] << 16;

361 int temp2 = (cos_tab[index + 1] - cos_tab[index]) *

362 ((offset << 8) + 0x80) << 1;

363

364 lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);

365 }

366

367 /*

368 * Compute sum and difference polynomial coefficients

369 * (bitexact alternative to lsp2poly() in lsp.c)

370 */

371 /* Initialize with values in Q28 */

372 f1[0] = 1 << 28;

373 f1[1] = (lpc[0] << 14) + (lpc[2] << 14);

374 f1[2] = lpc[0] * lpc[2] + (2 << 28);

375

376 f2[0] = 1 << 28;

377 f2[1] = (lpc[1] << 14) + (lpc[3] << 14);

378 f2[2] = lpc[1] * lpc[3] + (2 << 28);

379

380 /*

381 * Calculate and scale the coefficients by 1/2 in

382 * each iteration for a final scaling factor of Q25

383 */

384 for (i = 2; i < LPC_ORDER / 2; i++) {

385 f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);

386 f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);

387

388 for (j = i; j >= 2; j--) {

389 f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +

390 (f1[j] >> 1) + (f1[j - 2] >> 1);

391 f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +

392 (f2[j] >> 1) + (f2[j - 2] >> 1);

393 }

394

395 f1[0] >>= 1;

396 f2[0] >>= 1;

397 f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1;

398 f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;

399 }

400

401 /* Convert polynomial coefficients to LPC coefficients */

402 for (i = 0; i < LPC_ORDER / 2; i++) {

403 int64_t ff1 = f1[i + 1] + f1[i];

404 int64_t ff2 = f2[i + 1] - f2[i];

405

406 lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;

407 lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +

408 (1 << 15)) >> 16;

409 }

410 }

411

412 /**

413 * Quantize LSP frequencies by interpolation and convert them to

414 * the corresponding LPC coefficients.

415 *

416 * @param lpc buffer for LPC coefficients

417 * @param cur_lsp the current LSP vector

418 * @param prev_lsp the previous LSP vector

419 */

420 static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)

421 {

422 int i;

423 int16_t *lpc_ptr = lpc;

424

425 /* cur_lsp * 0.25 + prev_lsp * 0.75 */

426 ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,

427 4096, 12288, 1 << 13, 14, LPC_ORDER);

428 ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,

429 8192, 8192, 1 << 13, 14, LPC_ORDER);

430 ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,

431 12288, 4096, 1 << 13, 14, LPC_ORDER);

432 memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc));

433

434 for (i = 0; i < SUBFRAMES; i++) {

435 lsp2lpc(lpc_ptr);

436 lpc_ptr += LPC_ORDER;

437 }

438 }

439

440 /**

441 * Generate a train of dirac functions with period as pitch lag.

442 */

443 static void gen_dirac_train(int16_t *buf, int pitch_lag)

444 {

445 int16_t vector[SUBFRAME_LEN];

446 int i, j;

447

448 memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));

449 for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {

450 for (j = 0; j < SUBFRAME_LEN - i; j++)

451 buf[i + j] += vector[j];

452 }

453 }

454

455 /**

456 * Generate fixed codebook excitation vector.

457 *

458 * @param vector decoded excitation vector

459 * @param subfrm current subframe

460 * @param cur_rate current bitrate

461 * @param pitch_lag closed loop pitch lag

462 * @param index current subframe index

463 */

464 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,

465 enum Rate cur_rate, int pitch_lag, int index)

466 {

467 int temp, i, j;

468

469 memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));

470

471 if (cur_rate == RATE_6300) {

472 if (subfrm->pulse_pos >= max_pos[index])

473 return;

474

475 /* Decode amplitudes and positions */

476 j = PULSE_MAX - pulses[index];

477 temp = subfrm->pulse_pos;

478 for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {

479 temp -= combinatorial_table[j][i];

480 if (temp >= 0)

481 continue;

482 temp += combinatorial_table[j++][i];

483 if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {

484 vector[subfrm->grid_index + GRID_SIZE * i] =

485 -fixed_cb_gain[subfrm->amp_index];

486 } else {

487 vector[subfrm->grid_index + GRID_SIZE * i] =

488 fixed_cb_gain[subfrm->amp_index];

489 }

490 if (j == PULSE_MAX)

491 break;

492 }

493 if (subfrm->dirac_train == 1)

494 gen_dirac_train(vector, pitch_lag);

495 } else { /* 5300 bps */

496 int cb_gain = fixed_cb_gain[subfrm->amp_index];

497 int cb_shift = subfrm->grid_index;

498 int cb_sign = subfrm->pulse_sign;

499 int cb_pos = subfrm->pulse_pos;

500 int offset, beta, lag;

501

502 for (i = 0; i < 8; i += 2) {

503 offset = ((cb_pos & 7) << 3) + cb_shift + i;

504 vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;

505 cb_pos >>= 3;

506 cb_sign >>= 1;

507 }

508

509 /* Enhance harmonic components */

510 lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +

511 subfrm->ad_cb_lag - 1;

512 beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];

513

514 if (lag < SUBFRAME_LEN - 2) {

515 for (i = lag; i < SUBFRAME_LEN; i++)

516 vector[i] += beta * vector[i - lag] >> 15;

517 }

518 }

519 }

520

521 /**

522 * Get delayed contribution from the previous excitation vector.

523 */

524 static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)

525 {

526 int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;

527 int i;

528

529 residual[0] = prev_excitation[offset];

530 residual[1] = prev_excitation[offset + 1];

531

532 offset += 2;

533 for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)

534 residual[i] = prev_excitation[offset + (i - 2) % lag];

535 }

536

537 static int dot_product(const int16_t *a, const int16_t *b, int length)

538 {

539 int sum = ff_dot_product(a,b,length);

540 return av_sat_add32(sum, sum);

541 }

542

543 /**

544 * Generate adaptive codebook excitation.

545 */

546 static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,

547 int pitch_lag, G723_1_Subframe *subfrm,

548 enum Rate cur_rate)

549 {

550 int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];

551 const int16_t *cb_ptr;

552 int lag = pitch_lag + subfrm->ad_cb_lag - 1;

553

554 int i;

555 int sum;

556

557 get_residual(residual, prev_excitation, lag);

558

559 /* Select quantization table */

560 if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {

561 cb_ptr = adaptive_cb_gain85;

562 } else

563 cb_ptr = adaptive_cb_gain170;

564

565 /* Calculate adaptive vector */

566 cb_ptr += subfrm->ad_cb_gain * 20;

567 for (i = 0; i < SUBFRAME_LEN; i++) {

568 sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);

569 vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;

570 }

571 }

572

573 /**

574 * Estimate maximum auto-correlation around pitch lag.

575 *

576 * @param buf buffer with offset applied

577 * @param offset offset of the excitation vector

578 * @param ccr_max pointer to the maximum auto-correlation

579 * @param pitch_lag decoded pitch lag

580 * @param length length of autocorrelation

581 * @param dir forward lag(1) / backward lag(-1)

582 */

583 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,

584 int pitch_lag, int length, int dir)

585 {

586 int limit, ccr, lag = 0;

587 int i;

588

589 pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);

590 if (dir > 0)

591 limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);

592 else

593 limit = pitch_lag + 3;

594

595 for (i = pitch_lag - 3; i <= limit; i++) {

596 ccr = dot_product(buf, buf + dir * i, length);

597

598 if (ccr > *ccr_max) {

599 *ccr_max = ccr;

600 lag = i;

601 }

602 }

603 return lag;

604 }

605

606 /**

607 * Calculate pitch postfilter optimal and scaling gains.

608 *

609 * @param lag pitch postfilter forward/backward lag

610 * @param ppf pitch postfilter parameters

611 * @param cur_rate current bitrate

612 * @param tgt_eng target energy

613 * @param ccr cross-correlation

614 * @param res_eng residual energy

615 */

616 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,

617 int tgt_eng, int ccr, int res_eng)

618 {

619 int pf_residual; /* square of postfiltered residual */

620 int temp1, temp2;

621

622 ppf->index = lag;

623

624 temp1 = tgt_eng * res_eng >> 1;

625 temp2 = ccr * ccr << 1;

626

627 if (temp2 > temp1) {

628 if (ccr >= res_eng) {

629 ppf->opt_gain = ppf_gain_weight[cur_rate];

630 } else {

631 ppf->opt_gain = (ccr << 15) / res_eng *

632 ppf_gain_weight[cur_rate] >> 15;

633 }

634 /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */

635 temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);

636 temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;

637 pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;

638

639 if (tgt_eng >= pf_residual << 1) {

640 temp1 = 0x7fff;

641 } else {

642 temp1 = (tgt_eng << 14) / pf_residual;

643 }

644

645 /* scaling_gain = sqrt(tgt_eng/pf_res^2) */

646 ppf->sc_gain = square_root(temp1 << 16);

647 } else {

648 ppf->opt_gain = 0;

649 ppf->sc_gain = 0x7fff;

650 }

651

652 ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);

653 }

654

655 /**

656 * Calculate pitch postfilter parameters.

657 *

658 * @param p the context

659 * @param offset offset of the excitation vector

660 * @param pitch_lag decoded pitch lag

661 * @param ppf pitch postfilter parameters

662 * @param cur_rate current bitrate

663 */

664 static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,

665 PPFParam *ppf, enum Rate cur_rate)

666 {

667

668 int16_t scale;

669 int i;

670 int temp1, temp2;

671

672 /*

673 * 0 - target energy

674 * 1 - forward cross-correlation

675 * 2 - forward residual energy

676 * 3 - backward cross-correlation

677 * 4 - backward residual energy

678 */

679 int energy[5] = {0, 0, 0, 0, 0};

680 int16_t *buf = p->audio + LPC_ORDER + offset;

681 int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,

682 SUBFRAME_LEN, 1);

683 int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,

684 SUBFRAME_LEN, -1);

685

686 ppf->index = 0;

687 ppf->opt_gain = 0;

688 ppf->sc_gain = 0x7fff;

689

690 /* Case 0, Section 3.6 */

691 if (!back_lag && !fwd_lag)

692 return;

693

694 /* Compute target energy */

695 energy[0] = dot_product(buf, buf, SUBFRAME_LEN);

696

697 /* Compute forward residual energy */

698 if (fwd_lag)

699 energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);

700

701 /* Compute backward residual energy */

702 if (back_lag)

703 energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);

704

705 /* Normalize and shorten */

706 temp1 = 0;

707 for (i = 0; i < 5; i++)

708 temp1 = FFMAX(energy[i], temp1);

709

710 scale = normalize_bits(temp1, 31);

711 for (i = 0; i < 5; i++)

712 energy[i] = (energy[i] << scale) >> 16;

713

714 if (fwd_lag && !back_lag) { /* Case 1 */

715 comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],

716 energy[2]);

717 } else if (!fwd_lag) { /* Case 2 */

718 comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],

719 energy[4]);

720 } else { /* Case 3 */

721

722 /*

723 * Select the largest of energy[1]^2/energy[2]

724 * and energy[3]^2/energy[4]

725 */

726 temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);

727 temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);

728 if (temp1 >= temp2) {

729 comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],

730 energy[2]);

731 } else {

732 comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],

733 energy[4]);

734 }

735 }

736 }

737

738 /**

739 * Classify frames as voiced/unvoiced.

740 *

741 * @param p the context

742 * @param pitch_lag decoded pitch_lag

743 * @param exc_eng excitation energy estimation

744 * @param scale scaling factor of exc_eng

745 *

746 * @return residual interpolation index if voiced, 0 otherwise

747 */

748 static int comp_interp_index(G723_1_Context *p, int pitch_lag,

749 int *exc_eng, int *scale)

750 {

751 int offset = PITCH_MAX + 2 * SUBFRAME_LEN;

752 int16_t *buf = p->audio + LPC_ORDER;

753

754 int index, ccr, tgt_eng, best_eng, temp;

755

756 *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);

757 buf += offset;

758

759 /* Compute maximum backward cross-correlation */

760 ccr = 0;

761 index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);

762 ccr = av_sat_add32(ccr, 1 << 15) >> 16;

763

764 /* Compute target energy */

765 tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2);

766 *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;

767

768 if (ccr <= 0)

769 return 0;

770

771 /* Compute best energy */

772 best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);

773 best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;

774

775 temp = best_eng * *exc_eng >> 3;

776

777 if (temp < ccr * ccr) {

778 return index;

779 } else

780 return 0;

781 }

782

783 /**

784 * Peform residual interpolation based on frame classification.

785 *

786 * @param buf decoded excitation vector

787 * @param out output vector

788 * @param lag decoded pitch lag

789 * @param gain interpolated gain

790 * @param rseed seed for random number generator

791 */

792 static void residual_interp(int16_t *buf, int16_t *out, int lag,

793 int gain, int *rseed)

794 {

795 int i;

796 if (lag) { /* Voiced */

797 int16_t *vector_ptr = buf + PITCH_MAX;

798 /* Attenuate */

799 for (i = 0; i < lag; i++)

800 out[i] = vector_ptr[i - lag] * 3 >> 2;

801 av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),

802 (FRAME_LEN - lag) * sizeof(*out));

803 } else { /* Unvoiced */

804 for (i = 0; i < FRAME_LEN; i++) {

805 *rseed = *rseed * 521 + 259;

806 out[i] = gain * *rseed >> 15;

807 }

808 memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));

809 }

810 }

811

812 /**

813 * Perform IIR filtering.

814 *

815 * @param fir_coef FIR coefficients

816 * @param iir_coef IIR coefficients

817 * @param src source vector

818 * @param dest destination vector

819 * @param width width of the output, 16 bits(0) / 32 bits(1)

820 */

821 #define iir_filter(fir_coef, iir_coef, src, dest, width)\

822 {\

823 int m, n;\

824 int res_shift = 16 & ~-(width);\

825 int in_shift = 16 - res_shift;\

826 \

827 for (m = 0; m < SUBFRAME_LEN; m++) {\

828 int64_t filter = 0;\

829 for (n = 1; n <= LPC_ORDER; n++) {\

830 filter -= (fir_coef)[n - 1] * (src)[m - n] -\

831 (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\

832 }\

833 \

834 (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\

835 (1 << 15)) >> res_shift;\

836 }\

837 }

838

839 /**

840 * Adjust gain of postfiltered signal.

841 *

842 * @param p the context

843 * @param buf postfiltered output vector

844 * @param energy input energy coefficient

845 */

846 static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)

847 {

848 int num, denom, gain, bits1, bits2;

849 int i;

850

851 num = energy;

852 denom = 0;

853 for (i = 0; i < SUBFRAME_LEN; i++) {

854 int temp = buf[i] >> 2;

855 temp *= temp;

856 denom = av_sat_dadd32(denom, temp);

857 }

858

859 if (num && denom) {

860 bits1 = normalize_bits(num, 31);

861 bits2 = normalize_bits(denom, 31);

862 num = num << bits1 >> 1;

863 denom <<= bits2;

864

865 bits2 = 5 + bits1 - bits2;

866 bits2 = FFMAX(0, bits2);

867

868 gain = (num >> 1) / (denom >> 16);

869 gain = square_root(gain << 16 >> bits2);

870 } else {

871 gain = 1 << 12;

872 }

873

874 for (i = 0; i < SUBFRAME_LEN; i++) {

875 p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;

876 buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +

877 (1 << 10)) >> 11);

878 }

879 }

880

881 /**

882 * Perform formant filtering.

883 *

884 * @param p the context

885 * @param lpc quantized lpc coefficients

886 * @param buf input buffer

887 * @param dst output buffer

888 */

889 static void formant_postfilter(G723_1_Context *p, int16_t *lpc,

890 int16_t *buf, int16_t *dst)

891 {

892 int16_t filter_coef[2][LPC_ORDER];

893 int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;

894 int i, j, k;

895

896 memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));

897 memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));

898

899 for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {

900 for (k = 0; k < LPC_ORDER; k++) {

901 filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +

902 (1 << 14)) >> 15;

903 filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +

904 (1 << 14)) >> 15;

905 }

906 iir_filter(filter_coef[0], filter_coef[1], buf + i,

907 filter_signal + i, 1);

908 lpc += LPC_ORDER;

909 }

910

911 memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));

912 memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));

913

914 buf += LPC_ORDER;

915 signal_ptr = filter_signal + LPC_ORDER;

916 for (i = 0; i < SUBFRAMES; i++) {

917 int temp;

918 int auto_corr[2];

919 int scale, energy;

920

921 /* Normalize */

922 scale = scale_vector(dst, buf, SUBFRAME_LEN);

923

924 /* Compute auto correlation coefficients */

925 auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);

926 auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN);

927

928 /* Compute reflection coefficient */

929 temp = auto_corr[1] >> 16;

930 if (temp) {

931 temp = (auto_corr[0] >> 2) / temp;

932 }

933 p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;

934 temp = -p->reflection_coef >> 1 & ~3;

935

936 /* Compensation filter */

937 for (j = 0; j < SUBFRAME_LEN; j++) {

938 dst[j] = av_sat_dadd32(signal_ptr[j],

939 (signal_ptr[j - 1] >> 16) * temp) >> 16;

940 }

941

942 /* Compute normalized signal energy */

943 temp = 2 * scale + 4;

944 if (temp < 0) {

945 energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);

946 } else

947 energy = auto_corr[1] >> temp;

948

949 gain_scale(p, dst, energy);

950

951 buf += SUBFRAME_LEN;

952 signal_ptr += SUBFRAME_LEN;

953 dst += SUBFRAME_LEN;

954 }

955 }

956

957 static int sid_gain_to_lsp_index(int gain)

958 {

959 if (gain < 0x10)

960 return gain << 6;

961 else if (gain < 0x20)

962 return gain - 8 << 7;

963 else

964 return gain - 20 << 8;

965 }

966

967 static inline int cng_rand(int *state, int base)

968 {

969 *state = (*state * 521 + 259) & 0xFFFF;

970 return (*state & 0x7FFF) * base >> 15;

971 }

972

973 static int estimate_sid_gain(G723_1_Context *p)

974 {

975 int i, shift, seg, seg2, t, val, val_add, x, y;

976

977 shift = 16 - p->cur_gain * 2;

978 if (shift > 0)

979 t = p->sid_gain << shift;

980 else

981 t = p->sid_gain >> -shift;

982 x = t * cng_filt[0] >> 16;

983

984 if (x >= cng_bseg[2])

985 return 0x3F;

986

987 if (x >= cng_bseg[1]) {

988 shift = 4;

989 seg = 3;

990 } else {

991 shift = 3;

992 seg = (x >= cng_bseg[0]);

993 }

994 seg2 = FFMIN(seg, 3);

995

996 val = 1 << shift;

997 val_add = val >> 1;

998 for (i = 0; i < shift; i++) {

999 t = seg * 32 + (val << seg2);

1000 t *= t;

1001 if (x >= t)

1002 val += val_add;

1003 else

1004 val -= val_add;

1005 val_add >>= 1;

1006 }

1007

1008 t = seg * 32 + (val << seg2);

1009 y = t * t - x;

1010 if (y <= 0) {

1011 t = seg * 32 + (val + 1 << seg2);

1012 t = t * t - x;

1013 val = (seg2 - 1 << 4) + val;

1014 if (t >= y)

1015 val++;

1016 } else {

1017 t = seg * 32 + (val - 1 << seg2);

1018 t = t * t - x;

1019 val = (seg2 - 1 << 4) + val;

1020 if (t >= y)

1021 val--;

1022 }

1023

1024 return val;

1025 }

1026

1027 static void generate_noise(G723_1_Context *p)

1028 {

1029 int i, j, idx, t;

1030 int off[SUBFRAMES];

1031 int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];

1032 int tmp[SUBFRAME_LEN * 2];

1033 int16_t *vector_ptr;

1034 int64_t sum;

1035 int b0, c, delta, x, shift;

1036

1037 p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;

1038 p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;

1039

1040 for (i = 0; i < SUBFRAMES; i++) {

1041 p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;

1042 p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i];

1043 }

1044

1045 for (i = 0; i < SUBFRAMES / 2; i++) {

1046 t = cng_rand(&p->cng_random_seed, 1 << 13);

1047 off[i * 2] = t & 1;

1048 off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;

1049 t >>= 2;

1050 for (j = 0; j < 11; j++) {

1051 signs[i * 11 + j] = (t & 1) * 2 - 1 << 14;

1052 t >>= 1;

1053 }

1054 }

1055

1056 idx = 0;

1057 for (i = 0; i < SUBFRAMES; i++) {

1058 for (j = 0; j < SUBFRAME_LEN / 2; j++)

1059 tmp[j] = j;

1060 t = SUBFRAME_LEN / 2;

1061 for (j = 0; j < pulses[i]; j++, idx++) {

1062 int idx2 = cng_rand(&p->cng_random_seed, t);

1063

1064 pos[idx] = tmp[idx2] * 2 + off[i];

1065 tmp[idx2] = tmp[--t];

1066 }

1067 }

1068

1069 vector_ptr = p->audio + LPC_ORDER;

1070 memcpy(vector_ptr, p->prev_excitation,

1071 PITCH_MAX * sizeof(*p->excitation));

1072 for (i = 0; i < SUBFRAMES; i += 2) {

1073 gen_acb_excitation(vector_ptr, vector_ptr,

1074 p->pitch_lag[i >> 1], &p->subframe[i],

1075 p->cur_rate);

1076 gen_acb_excitation(vector_ptr + SUBFRAME_LEN,

1077 vector_ptr + SUBFRAME_LEN,

1078 p->pitch_lag[i >> 1], &p->subframe[i + 1],

1079 p->cur_rate);

1080

1081 t = 0;

1082 for (j = 0; j < SUBFRAME_LEN * 2; j++)

1083 t |= FFABS(vector_ptr[j]);

1084 t = FFMIN(t, 0x7FFF);

1085 if (!t) {

1086 shift = 0;

1087 } else {

1088 shift = -10 + av_log2(t);

1089 if (shift < -2)

1090 shift = -2;

1091 }

1092 sum = 0;

1093 if (shift < 0) {

1094 for (j = 0; j < SUBFRAME_LEN * 2; j++) {

1095 t = vector_ptr[j] << -shift;

1096 sum += t * t;

1097 tmp[j] = t;

1098 }

1099 } else {

1100 for (j = 0; j < SUBFRAME_LEN * 2; j++) {

1101 t = vector_ptr[j] >> shift;

1102 sum += t * t;

1103 tmp[j] = t;

1104 }

1105 }

1106

1107 b0 = 0;

1108 for (j = 0; j < 11; j++)

1109 b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];

1110 b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11

1111

1112 c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);

1113 if (shift * 2 + 3 >= 0)

1114 c >>= shift * 2 + 3;

1115 else

1116 c <<= -(shift * 2 + 3);

1117 c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;

1118

1119 delta = b0 * b0 * 2 - c;

1120 if (delta <= 0) {

1121 x = -b0;

1122 } else {

1123 delta = square_root(delta);

1124 x = delta - b0;

1125 t = delta + b0;

1126 if (FFABS(t) < FFABS(x))

1127 x = -t;

1128 }

1129 shift++;

1130 if (shift < 0)

1131 x >>= -shift;

1132 else

1133 x <<= shift;

1134 x = av_clip(x, -10000, 10000);

1135

1136 for (j = 0; j < 11; j++) {

1137 idx = (i / 2) * 11 + j;

1138 vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +

1139 (x * signs[idx] >> 15));

1140 }

1141

1142 /* copy decoded data to serve as a history for the next decoded subframes */

1143 memcpy(vector_ptr + PITCH_MAX, vector_ptr,

1144 sizeof(*vector_ptr) * SUBFRAME_LEN * 2);

1145 vector_ptr += SUBFRAME_LEN * 2;

1146 }

1147 /* Save the excitation for the next frame */

1148 memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,

1149 PITCH_MAX * sizeof(*p->excitation));

1150 }

1151

1152 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,

1153 int *got_frame_ptr, AVPacket *avpkt)

1154 {

1155 G723_1_Context *p = avctx->priv_data;

1156 AVFrame *frame = data;

1157 const uint8_t *buf = avpkt->data;

1158 int buf_size = avpkt->size;

1159 int dec_mode = buf[0] & 3;

1160

1161 PPFParam ppf[SUBFRAMES];

1162 int16_t cur_lsp[LPC_ORDER];

1163 int16_t lpc[SUBFRAMES * LPC_ORDER];

1164 int16_t acb_vector[SUBFRAME_LEN];

1165 int16_t *out;

1166 int bad_frame = 0, i, j, ret;

1167 int16_t *audio = p->audio;

1168

1169 if (buf_size < frame_size[dec_mode]) {

1170 if (buf_size)

1171 av_log(avctx, AV_LOG_WARNING,

1172 "Expected %d bytes, got %d - skipping packet\n",

1173 frame_size[dec_mode], buf_size);

1174 *got_frame_ptr = 0;

1175 return buf_size;

1176 }

1177

1178 if (unpack_bitstream(p, buf, buf_size) < 0) {

1179 bad_frame = 1;

1180 if (p->past_frame_type == ACTIVE_FRAME)

1181 p->cur_frame_type = ACTIVE_FRAME;

1182 else

1183 p->cur_frame_type = UNTRANSMITTED_FRAME;

1184 }

1185

1186 frame->nb_samples = FRAME_LEN;

1187 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

1188 return ret;

1189

1190 out = (int16_t *)frame->data[0];

1191

1192 if (p->cur_frame_type == ACTIVE_FRAME) {

1193 if (!bad_frame)

1194 p->erased_frames = 0;

1195 else if (p->erased_frames != 3)

1196 p->erased_frames++;

1197

1198 inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);

1199 lsp_interpolate(lpc, cur_lsp, p->prev_lsp);

1200

1201 /* Save the lsp_vector for the next frame */

1202 memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));

1203

1204 /* Generate the excitation for the frame */

1205 memcpy(p->excitation, p->prev_excitation,

1206 PITCH_MAX * sizeof(*p->excitation));

1207 if (!p->erased_frames) {

1208 int16_t *vector_ptr = p->excitation + PITCH_MAX;

1209

1210 /* Update interpolation gain memory */

1211 p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +

1212 p->subframe[3].amp_index) >> 1];

1213 for (i = 0; i < SUBFRAMES; i++) {

1214 gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,

1215 p->pitch_lag[i >> 1], i);

1216 gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],

1217 p->pitch_lag[i >> 1], &p->subframe[i],

1218 p->cur_rate);

1219 /* Get the total excitation */

1220 for (j = 0; j < SUBFRAME_LEN; j++) {

1221 int v = av_clip_int16(vector_ptr[j] << 1);

1222 vector_ptr[j] = av_clip_int16(v + acb_vector[j]);

1223 }

1224 vector_ptr += SUBFRAME_LEN;

1225 }

1226

1227 vector_ptr = p->excitation + PITCH_MAX;

1228

1229 p->interp_index = comp_interp_index(p, p->pitch_lag[1],

1230 &p->sid_gain, &p->cur_gain);

1231

1232 /* Peform pitch postfiltering */

1233 if (p->postfilter) {

1234 i = PITCH_MAX;

1235 for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)

1236 comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],

1237 ppf + j, p->cur_rate);

1238

1239 for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)

1240 ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,

1241 vector_ptr + i,

1242 vector_ptr + i + ppf[j].index,

1243 ppf[j].sc_gain,

1244 ppf[j].opt_gain,

1245 1 << 14, 15, SUBFRAME_LEN);

1246 } else {

1247 audio = vector_ptr - LPC_ORDER;

1248 }

1249

1250 /* Save the excitation for the next frame */

1251 memcpy(p->prev_excitation, p->excitation + FRAME_LEN,

1252 PITCH_MAX * sizeof(*p->excitation));

1253 } else {

1254 p->interp_gain = (p->interp_gain * 3 + 2) >> 2;

1255 if (p->erased_frames == 3) {

1256 /* Mute output */

1257 memset(p->excitation, 0,

1258 (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));

1259 memset(p->prev_excitation, 0,

1260 PITCH_MAX * sizeof(*p->excitation));

1261 memset(frame->data[0], 0,

1262 (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));

1263 } else {

1264 int16_t *buf = p->audio + LPC_ORDER;

1265

1266 /* Regenerate frame */

1267 residual_interp(p->excitation, buf, p->interp_index,

1268 p->interp_gain, &p->random_seed);

1269

1270 /* Save the excitation for the next frame */

1271 memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),

1272 PITCH_MAX * sizeof(*p->excitation));

1273 }

1274 }

1275 p->cng_random_seed = CNG_RANDOM_SEED;

1276 } else {

1277 if (p->cur_frame_type == SID_FRAME) {

1278 p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);

1279 inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);

1280 } else if (p->past_frame_type == ACTIVE_FRAME) {

1281 p->sid_gain = estimate_sid_gain(p);

1282 }

1283

1284 if (p->past_frame_type == ACTIVE_FRAME)

1285 p->cur_gain = p->sid_gain;

1286 else

1287 p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;

1288 generate_noise(p);

1289 lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);

1290 /* Save the lsp_vector for the next frame */

1291 memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));

1292 }

1293

1294 p->past_frame_type = p->cur_frame_type;

1295

1296 memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));

1297 for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)

1298 ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],

1299 audio + i, SUBFRAME_LEN, LPC_ORDER,

1300 0, 1, 1 << 12);

1301 memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));

1302

1303 if (p->postfilter) {

1304 formant_postfilter(p, lpc, p->audio, out);

1305 } else { // if output is not postfiltered it should be scaled by 2

1306 for (i = 0; i < FRAME_LEN; i++)

1307 out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);

1308 }

1309

1310 *got_frame_ptr = 1;

1311

1312 return frame_size[dec_mode];

1313 }

1314

1315 #define OFFSET(x) offsetof(G723_1_Context, x)

1316 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM

1317

1318 static const AVOption options[] = {

1319 { "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,

1320 { .i64 = 1 }, 0, 1, AD },

1321 { NULL }

1322 };

1323

1324

1325 static const AVClass g723_1dec_class = {

1326 .class_name = "G.723.1 decoder",

1327 .item_name = av_default_item_name,

1328 .option = options,

1329 .version = LIBAVUTIL_VERSION_INT,

1330 };

1331

1332 AVCodec ff_g723_1_decoder = {

1333 .name = "g723_1",

1334 .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),

1335 .type = AVMEDIA_TYPE_AUDIO,

1336 .id = AV_CODEC_ID_G723_1,

1337 .priv_data_size = sizeof(G723_1_Context),

1338 .init = g723_1_decode_init,

1339 .decode = g723_1_decode_frame,

1340 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,

1341 .priv_class = &g723_1dec_class,

1342 };

1343

1344 #if CONFIG_G723_1_ENCODER

1345 #define BITSTREAM_WRITER_LE

1346 #include "put_bits.h"

1347

1348 static av_cold int g723_1_encode_init(AVCodecContext *avctx)

1349 {

1350 G723_1_Context *p = avctx->priv_data;

1351

1352 if (avctx->sample_rate != 8000) {

1353 av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");

1354 return -1;

1355 }

1356

1357 if (avctx->channels != 1) {

1358 av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");

1359 return AVERROR(EINVAL);

1360 }

1361

1362 if (avctx->bit_rate == 6300) {

1363 p->cur_rate = RATE_6300;

1364 } else if (avctx->bit_rate == 5300) {

1365 av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");

1366 return AVERROR_PATCHWELCOME;

1367 } else {

1368 av_log(avctx, AV_LOG_ERROR,

1369 "Bitrate not supported, use 6.3k\n");

1370 return AVERROR(EINVAL);

1371 }

1372 avctx->frame_size = 240;

1373 memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));

1374

1375 return 0;

1376 }

1377

1378 /**

1379 * Remove DC component from the input signal.

1380 *

1381 * @param buf input signal

1382 * @param fir zero memory

1383 * @param iir pole memory

1384 */

1385 static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)

1386 {

1387 int i;

1388 for (i = 0; i < FRAME_LEN; i++) {

1389 *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);

1390 *fir = buf[i];

1391 buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;

1392 }

1393 }

1394

1395 /**

1396 * Estimate autocorrelation of the input vector.

1397 *

1398 * @param buf input buffer

1399 * @param autocorr autocorrelation coefficients vector

1400 */

1401 static void comp_autocorr(int16_t *buf, int16_t *autocorr)

1402 {

1403 int i, scale, temp;

1404 int16_t vector[LPC_FRAME];

1405

1406 scale_vector(vector, buf, LPC_FRAME);

1407

1408 /* Apply the Hamming window */

1409 for (i = 0; i < LPC_FRAME; i++)

1410 vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;

1411

1412 /* Compute the first autocorrelation coefficient */

1413 temp = ff_dot_product(vector, vector, LPC_FRAME);

1414

1415 /* Apply a white noise correlation factor of (1025/1024) */

1416 temp += temp >> 10;

1417

1418 /* Normalize */

1419 scale = normalize_bits_int32(temp);

1420 autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +

1421 (1 << 15)) >> 16;

1422

1423 /* Compute the remaining coefficients */

1424 if (!autocorr[0]) {

1425 memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));

1426 } else {

1427 for (i = 1; i <= LPC_ORDER; i++) {

1428 temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);

1429 temp = MULL2((temp << scale), binomial_window[i - 1]);

1430 autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;

1431 }

1432 }

1433 }

1434

1435 /**

1436 * Use Levinson-Durbin recursion to compute LPC coefficients from

1437 * autocorrelation values.

1438 *

1439 * @param lpc LPC coefficients vector

1440 * @param autocorr autocorrelation coefficients vector

1441 * @param error prediction error

1442 */

1443 static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)

1444 {

1445 int16_t vector[LPC_ORDER];

1446 int16_t partial_corr;

1447 int i, j, temp;

1448

1449 memset(lpc, 0, LPC_ORDER * sizeof(int16_t));

1450

1451 for (i = 0; i < LPC_ORDER; i++) {

1452 /* Compute the partial correlation coefficient */

1453 temp = 0;

1454 for (j = 0; j < i; j++)

1455 temp -= lpc[j] * autocorr[i - j - 1];

1456 temp = ((autocorr[i] << 13) + temp) << 3;

1457

1458 if (FFABS(temp) >= (error << 16))

1459 break;

1460

1461 partial_corr = temp / (error << 1);

1462

1463 lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +

1464 (1 << 15)) >> 16;

1465

1466 /* Update the prediction error */

1467 temp = MULL2(temp, partial_corr);

1468 error = av_clipl_int32((int64_t)(error << 16) - temp +

1469 (1 << 15)) >> 16;

1470

1471 memcpy(vector, lpc, i * sizeof(int16_t));

1472 for (j = 0; j < i; j++) {

1473 temp = partial_corr * vector[i - j - 1] << 1;

1474 lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +

1475 (1 << 15)) >> 16;

1476 }

1477 }

1478 }

1479

1480 /**

1481 * Calculate LPC coefficients for the current frame.

1482 *

1483 * @param buf current frame

1484 * @param prev_data 2 trailing subframes of the previous frame

1485 * @param lpc LPC coefficients vector

1486 */

1487 static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)

1488 {

1489 int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];

1490 int16_t *autocorr_ptr = autocorr;

1491 int16_t *lpc_ptr = lpc;

1492 int i, j;

1493

1494 for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {

1495 comp_autocorr(buf + i, autocorr_ptr);

1496 levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);

1497

1498 lpc_ptr += LPC_ORDER;

1499 autocorr_ptr += LPC_ORDER + 1;

1500 }

1501 }

1502

1503 static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)

1504 {

1505 int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference

1506 ///< polynomials (F1, F2) ordered as

1507 ///< f1[0], f2[0], ...., f1[5], f2[5]

1508

1509 int max, shift, cur_val, prev_val, count, p;

1510 int i, j;

1511 int64_t temp;

1512

1513 /* Initialize f1[0] and f2[0] to 1 in Q25 */

1514 for (i = 0; i < LPC_ORDER; i++)

1515 lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;

1516

1517 /* Apply bandwidth expansion on the LPC coefficients */

1518 f[0] = f[1] = 1 << 25;

1519

1520 /* Compute the remaining coefficients */

1521 for (i = 0; i < LPC_ORDER / 2; i++) {

1522 /* f1 */

1523 f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);

1524 /* f2 */

1525 f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);

1526 }

1527

1528 /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */

1529 f[LPC_ORDER] >>= 1;

1530 f[LPC_ORDER + 1] >>= 1;

1531

1532 /* Normalize and shorten */

1533 max = FFABS(f[0]);

1534 for (i = 1; i < LPC_ORDER + 2; i++)

1535 max = FFMAX(max, FFABS(f[i]));

1536

1537 shift = normalize_bits_int32(max);

1538

1539 for (i = 0; i < LPC_ORDER + 2; i++)

1540 f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;

1541

1542 /**

1543 * Evaluate F1 and F2 at uniform intervals of pi/256 along the

1544 * unit circle and check for zero crossings.

1545 */

1546 p = 0;

1547 temp = 0;

1548 for (i = 0; i <= LPC_ORDER / 2; i++)

1549 temp += f[2 * i] * cos_tab[0];

1550 prev_val = av_clipl_int32(temp << 1);

1551 count = 0;

1552 for ( i = 1; i < COS_TBL_SIZE / 2; i++) {

1553 /* Evaluate */

1554 temp = 0;

1555 for (j = 0; j <= LPC_ORDER / 2; j++)

1556 temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];

1557 cur_val = av_clipl_int32(temp << 1);

1558

1559 /* Check for sign change, indicating a zero crossing */

1560 if ((cur_val ^ prev_val) < 0) {

1561 int abs_cur = FFABS(cur_val);

1562 int abs_prev = FFABS(prev_val);

1563 int sum = abs_cur + abs_prev;

1564

1565 shift = normalize_bits_int32(sum);

1566 sum <<= shift;

1567 abs_prev = abs_prev << shift >> 8;

1568 lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);

1569

1570 if (count == LPC_ORDER)

1571 break;

1572

1573 /* Switch between sum and difference polynomials */

1574 p ^= 1;

1575

1576 /* Evaluate */

1577 temp = 0;

1578 for (j = 0; j <= LPC_ORDER / 2; j++){

1579 temp += f[LPC_ORDER - 2 * j + p] *

1580 cos_tab[i * j % COS_TBL_SIZE];

1581 }

1582 cur_val = av_clipl_int32(temp<<1);

1583 }

1584 prev_val = cur_val;

1585 }

1586

1587 if (count != LPC_ORDER)

1588 memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));

1589 }

1590

1591 /**

1592 * Quantize the current LSP subvector.

1593 *

1594 * @param num band number

1595 * @param offset offset of the current subvector in an LPC_ORDER vector

1596 * @param size size of the current subvector

1597 */

1598 #define get_index(num, offset, size) \

1599 {\

1600 int error, max = -1;\

1601 int16_t temp[4];\

1602 int i, j;\

1603 for (i = 0; i < LSP_CB_SIZE; i++) {\

1604 for (j = 0; j < size; j++){\

1605 temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\

1606 (1 << 14)) >> 15;\

1607 }\

1608 error = dot_product(lsp + (offset), temp, size) << 1;\

1609 error -= dot_product(lsp_band##num[i], temp, size);\

1610 if (error > max) {\

1611 max = error;\

1612 lsp_index[num] = i;\

1613 }\

1614 }\

1615 }

1616

1617 /**

1618 * Vector quantize the LSP frequencies.

1619 *

1620 * @param lsp the current lsp vector

1621 * @param prev_lsp the previous lsp vector

1622 */

1623 static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)

1624 {

1625 int16_t weight[LPC_ORDER];

1626 int16_t min, max;

1627 int shift, i;

1628

1629 /* Calculate the VQ weighting vector */

1630 weight[0] = (1 << 20) / (lsp[1] - lsp[0]);

1631 weight[LPC_ORDER - 1] = (1 << 20) /

1632 (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);

1633

1634 for (i = 1; i < LPC_ORDER - 1; i++) {

1635 min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);

1636 if (min > 0x20)

1637 weight[i] = (1 << 20) / min;

1638 else

1639 weight[i] = INT16_MAX;

1640 }

1641

1642 /* Normalize */

1643 max = 0;

1644 for (i = 0; i < LPC_ORDER; i++)

1645 max = FFMAX(weight[i], max);

1646

1647 shift = normalize_bits_int16(max);

1648 for (i = 0; i < LPC_ORDER; i++) {

1649 weight[i] <<= shift;

1650 }

1651

1652 /* Compute the VQ target vector */

1653 for (i = 0; i < LPC_ORDER; i++) {

1654 lsp[i] -= dc_lsp[i] +

1655 (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);

1656 }

1657

1658 get_index(0, 0, 3);

1659 get_index(1, 3, 3);

1660 get_index(2, 6, 4);

1661 }

1662

1663 /**

1664 * Apply the formant perceptual weighting filter.

1665 *

1666 * @param flt_coef filter coefficients

1667 * @param unq_lpc unquantized lpc vector

1668 */

1669 static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,

1670 int16_t *unq_lpc, int16_t *buf)

1671 {

1672 int16_t vector[FRAME_LEN + LPC_ORDER];

1673 int i, j, k, l = 0;

1674

1675 memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);

1676 memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);

1677 memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);

1678

1679 for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {

1680 for (k = 0; k < LPC_ORDER; k++) {

1681 flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +

1682 (1 << 14)) >> 15;

1683 flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *

1684 percept_flt_tbl[1][k] +

1685 (1 << 14)) >> 15;

1686 }

1687 iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,

1688 buf + i, 0);

1689 l += LPC_ORDER;

1690 }

1691 memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);

1692 memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);

1693 }

1694

1695 /**

1696 * Estimate the open loop pitch period.

1697 *

1698 * @param buf perceptually weighted speech

1699 * @param start estimation is carried out from this position

1700 */

1701 static int estimate_pitch(int16_t *buf, int start)

1702 {

1703 int max_exp = 32;

1704 int max_ccr = 0x4000;

1705 int max_eng = 0x7fff;

1706 int index = PITCH_MIN;

1707 int offset = start - PITCH_MIN + 1;

1708

1709 int ccr, eng, orig_eng, ccr_eng, exp;

1710 int diff, temp;

1711

1712 int i;

1713

1714 orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);

1715

1716 for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {

1717 offset--;

1718

1719 /* Update energy and compute correlation */

1720 orig_eng += buf[offset] * buf[offset] -

1721 buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];

1722 ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);

1723 if (ccr <= 0)

1724 continue;

1725

1726 /* Split into mantissa and exponent to maintain precision */

1727 exp = normalize_bits_int32(ccr);

1728 ccr = av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;

1729 exp <<= 1;

1730 ccr *= ccr;

1731 temp = normalize_bits_int32(ccr);

1732 ccr = ccr << temp >> 16;

1733 exp += temp;

1734

1735 temp = normalize_bits_int32(orig_eng);

1736 eng = av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;

1737 exp -= temp;

1738

1739 if (ccr >= eng) {

1740 exp--;

1741 ccr >>= 1;

1742 }

1743 if (exp > max_exp)

1744 continue;

1745

1746 if (exp + 1 < max_exp)

1747 goto update;

1748

1749 /* Equalize exponents before comparison */

1750 if (exp + 1 == max_exp)

1751 temp = max_ccr >> 1;

1752 else

1753 temp = max_ccr;

1754 ccr_eng = ccr * max_eng;

1755 diff = ccr_eng - eng * temp;

1756 if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {

1757 update:

1758 index = i;

1759 max_exp = exp;

1760 max_ccr = ccr;

1761 max_eng = eng;

1762 }

1763 }

1764 return index;

1765 }

1766

1767 /**

1768 * Compute harmonic noise filter parameters.

1769 *

1770 * @param buf perceptually weighted speech

1771 * @param pitch_lag open loop pitch period

1772 * @param hf harmonic filter parameters

1773 */

1774 static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)

1775 {

1776 int ccr, eng, max_ccr, max_eng;

1777 int exp, max, diff;

1778 int energy[15];

1779 int i, j;

1780

1781 for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {

1782 /* Compute residual energy */

1783 energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);

1784 /* Compute correlation */

1785 energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);

1786 }

1787

1788 /* Compute target energy */

1789 energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);

1790

1791 /* Normalize */

1792 max = 0;

1793 for (i = 0; i < 15; i++)

1794 max = FFMAX(max, FFABS(energy[i]));

1795

1796 exp = normalize_bits_int32(max);

1797 for (i = 0; i < 15; i++) {

1798 energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +

1799 (1 << 15)) >> 16;

1800 }

1801

1802 hf->index = -1;

1803 hf->gain = 0;

1804 max_ccr = 1;

1805 max_eng = 0x7fff;

1806

1807 for (i = 0; i <= 6; i++) {

1808 eng = energy[i << 1];

1809 ccr = energy[(i << 1) + 1];

1810

1811 if (ccr <= 0)

1812 continue;

1813

1814 ccr = (ccr * ccr + (1 << 14)) >> 15;

1815 diff = ccr * max_eng - eng * max_ccr;

1816 if (diff > 0) {

1817 max_ccr = ccr;

1818 max_eng = eng;

1819 hf->index = i;

1820 }

1821 }

1822

1823 if (hf->index == -1) {

1824 hf->index = pitch_lag;

1825 return;

1826 }

1827

1828 eng = energy[14] * max_eng;

1829 eng = (eng >> 2) + (eng >> 3);

1830 ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];

1831 if (eng < ccr) {

1832 eng = energy[(hf->index << 1) + 1];

1833

1834 if (eng >= max_eng)

1835 hf->gain = 0x2800;

1836 else

1837 hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;

1838 }

1839 hf->index += pitch_lag - 3;

1840 }

1841

1842 /**

1843 * Apply the harmonic noise shaping filter.

1844 *

1845 * @param hf filter parameters

1846 */

1847 static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)

1848 {

1849 int i;

1850

1851 for (i = 0; i < SUBFRAME_LEN; i++) {

1852 int64_t temp = hf->gain * src[i - hf->index] << 1;

1853 dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;

1854 }

1855 }

1856

1857 static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)

1858 {

1859 int i;

1860 for (i = 0; i < SUBFRAME_LEN; i++) {

1861 int64_t temp = hf->gain * src[i - hf->index] << 1;

1862 dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +

1863 (1 << 15)) >> 16;

1864

1865 }

1866 }

1867

1868 /**

1869 * Combined synthesis and formant perceptual weighting filer.

1870 *

1871 * @param qnt_lpc quantized lpc coefficients

1872 * @param perf_lpc perceptual filter coefficients

1873 * @param perf_fir perceptual filter fir memory

1874 * @param perf_iir perceptual filter iir memory

1875 * @param scale the filter output will be scaled by 2^scale

1876 */

1877 static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,

1878 int16_t *perf_fir, int16_t *perf_iir,

1879 const int16_t *src, int16_t *dest, int scale)

1880 {

1881 int i, j;

1882 int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];

1883 int64_t buf[SUBFRAME_LEN];

1884

1885 int16_t *bptr_16 = buf_16 + LPC_ORDER;

1886

1887 memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);

1888 memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);

1889

1890 for (i = 0; i < SUBFRAME_LEN; i++) {

1891 int64_t temp = 0;

1892 for (j = 1; j <= LPC_ORDER; j++)

1893 temp -= qnt_lpc[j - 1] * bptr_16[i - j];

1894

1895 buf[i] = (src[i] << 15) + (temp << 3);

1896 bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;

1897 }

1898

1899 for (i = 0; i < SUBFRAME_LEN; i++) {

1900 int64_t fir = 0, iir = 0;

1901 for (j = 1; j <= LPC_ORDER; j++) {

1902 fir -= perf_lpc[j - 1] * bptr_16[i - j];

1903 iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];

1904 }

1905 dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +

1906 (1 << 15)) >> 16;

1907 }

1908 memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);

1909 memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,

1910 sizeof(int16_t) * LPC_ORDER);

1911 }

1912

1913 /**

1914 * Compute the adaptive codebook contribution.

1915 *

1916 * @param buf input signal

1917 * @param index the current subframe index

1918 */

1919 static void acb_search(G723_1_Context *p, int16_t *residual,

1920 int16_t *impulse_resp, const int16_t *buf,

1921 int index)

1922 {

1923

1924 int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];

1925

1926 const int16_t *cb_tbl = adaptive_cb_gain85;

1927

1928 int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];

1929

1930 int pitch_lag = p->pitch_lag[index >> 1];

1931 int acb_lag = 1;

1932 int acb_gain = 0;

1933 int odd_frame = index & 1;

1934 int iter = 3 + odd_frame;

1935 int count = 0;

1936 int tbl_size = 85;

1937

1938 int i, j, k, l, max;

1939 int64_t temp;

1940

1941 if (!odd_frame) {

1942 if (pitch_lag == PITCH_MIN)

1943 pitch_lag++;

1944 else

1945 pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);

1946 }

1947

1948 for (i = 0; i < iter; i++) {

1949 get_residual(residual, p->prev_excitation, pitch_lag + i - 1);

1950

1951 for (j = 0; j < SUBFRAME_LEN; j++) {

1952 temp = 0;

1953 for (k = 0; k <= j; k++)

1954 temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];

1955 flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +

1956 (1 << 15)) >> 16;

1957 }

1958

1959 for (j = PITCH_ORDER - 2; j >= 0; j--) {

1960 flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;

1961 for (k = 1; k < SUBFRAME_LEN; k++) {

1962 temp = (flt_buf[j + 1][k - 1] << 15) +

1963 residual[j] * impulse_resp[k];

1964 flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;

1965 }

1966 }

1967

1968 /* Compute crosscorrelation with the signal */

1969 for (j = 0; j < PITCH_ORDER; j++) {

1970 temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);

1971 ccr_buf[count++] = av_clipl_int32(temp << 1);

1972 }

1973

1974 /* Compute energies */

1975 for (j = 0; j < PITCH_ORDER; j++) {

1976 ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],

1977 SUBFRAME_LEN);

1978 }

1979

1980 for (j = 1; j < PITCH_ORDER; j++) {

1981 for (k = 0; k < j; k++) {

1982 temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);

1983 ccr_buf[count++] = av_clipl_int32(temp<<2);

1984 }

1985 }

1986 }

1987

1988 /* Normalize and shorten */

1989 max = 0;

1990 for (i = 0; i < 20 * iter; i++)

1991 max = FFMAX(max, FFABS(ccr_buf[i]));

1992

1993 temp = normalize_bits_int32(max);

1994

1995 for (i = 0; i < 20 * iter; i++){

1996 ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +

1997 (1 << 15)) >> 16;

1998 }

1999

2000 max = 0;

2001 for (i = 0; i < iter; i++) {

2002 /* Select quantization table */

2003 if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||

2004 odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {

2005 cb_tbl = adaptive_cb_gain170;

2006 tbl_size = 170;

2007 }

2008

2009 for (j = 0, k = 0; j < tbl_size; j++, k += 20) {

2010 temp = 0;

2011 for (l = 0; l < 20; l++)

2012 temp += ccr_buf[20 * i + l] * cb_tbl[k + l];

2013 temp = av_clipl_int32(temp);

2014

2015 if (temp > max) {

2016 max = temp;

2017 acb_gain = j;

2018 acb_lag = i;

2019 }

2020 }

2021 }

2022

2023 if (!odd_frame) {

2024 pitch_lag += acb_lag - 1;

2025 acb_lag = 1;

2026 }

2027

2028 p->pitch_lag[index >> 1] = pitch_lag;

2029 p->subframe[index].ad_cb_lag = acb_lag;

2030 p->subframe[index].ad_cb_gain = acb_gain;

2031 }

2032

2033 /**

2034 * Subtract the adaptive codebook contribution from the input

2035 * to obtain the residual.

2036 *

2037 * @param buf target vector

2038 */

2039 static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,

2040 int16_t *buf)

2041 {

2042 int i, j;

2043 /* Subtract adaptive CB contribution to obtain the residual */

2044 for (i = 0; i < SUBFRAME_LEN; i++) {

2045 int64_t temp = buf[i] << 14;

2046 for (j = 0; j <= i; j++)

2047 temp -= residual[j] * impulse_resp[i - j];

2048

2049 buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;

2050 }

2051 }

2052

2053 /**

2054 * Quantize the residual signal using the fixed codebook (MP-MLQ).

2055 *

2056 * @param optim optimized fixed codebook parameters

2057 * @param buf excitation vector

2058 */

2059 static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,

2060 int16_t *buf, int pulse_cnt, int pitch_lag)

2061 {

2062 FCBParam param;

2063 int16_t impulse_r[SUBFRAME_LEN];

2064 int16_t temp_corr[SUBFRAME_LEN];

2065 int16_t impulse_corr[SUBFRAME_LEN];

2066

2067 int ccr1[SUBFRAME_LEN];

2068 int ccr2[SUBFRAME_LEN];

2069 int amp, err, max, max_amp_index, min, scale, i, j, k, l;

2070

2071 int64_t temp;

2072

2073 /* Update impulse response */

2074 memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);

2075 param.dirac_train = 0;

2076 if (pitch_lag < SUBFRAME_LEN - 2) {

2077 param.dirac_train = 1;

2078 gen_dirac_train(impulse_r, pitch_lag);

2079 }

2080

2081 for (i = 0; i < SUBFRAME_LEN; i++)

2082 temp_corr[i] = impulse_r[i] >> 1;

2083

2084 /* Compute impulse response autocorrelation */

2085 temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);

2086

2087 scale = normalize_bits_int32(temp);

2088 impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;

2089

2090 for (i = 1; i < SUBFRAME_LEN; i++) {

2091 temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);

2092 impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;

2093 }

2094

2095 /* Compute crosscorrelation of impulse response with residual signal */

2096 scale -= 4;

2097 for (i = 0; i < SUBFRAME_LEN; i++){

2098 temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);

2099 if (scale < 0)

2100 ccr1[i] = temp >> -scale;

2101 else

2102 ccr1[i] = av_clipl_int32(temp << scale);

2103 }

2104

2105 /* Search loop */

2106 for (i = 0; i < GRID_SIZE; i++) {

2107 /* Maximize the crosscorrelation */

2108 max = 0;

2109 for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {

2110 temp = FFABS(ccr1[j]);

2111 if (temp >= max) {

2112 max = temp;

2113 param.pulse_pos[0] = j;

2114 }

2115 }

2116

2117 /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */

2118 amp = max;

2119 min = 1 << 30;

2120 max_amp_index = GAIN_LEVELS - 2;

2121 for (j = max_amp_index; j >= 2; j--) {

2122 temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *

2123 impulse_corr[0] << 1);

2124 temp = FFABS(temp - amp);

2125 if (temp < min) {

2126 min = temp;

2127 max_amp_index = j;

2128 }

2129 }

2130

2131 max_amp_index--;

2132 /* Select additional gain values */

2133 for (j = 1; j < 5; j++) {

2134 for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {

2135 temp_corr[k] = 0;

2136 ccr2[k] = ccr1[k];

2137 }

2138 param.amp_index = max_amp_index + j - 2;

2139 amp = fixed_cb_gain[param.amp_index];

2140

2141 param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;

2142 temp_corr[param.pulse_pos[0]] = 1;

2143

2144 for (k = 1; k < pulse_cnt; k++) {

2145 max = -1 << 30;

2146 for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {

2147 if (temp_corr[l])

2148 continue;

2149 temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];

2150 temp = av_clipl_int32((int64_t)temp *

2151 param.pulse_sign[k - 1] << 1);

2152 ccr2[l] -= temp;

2153 temp = FFABS(ccr2[l]);

2154 if (temp > max) {

2155 max = temp;

2156 param.pulse_pos[k] = l;

2157 }

2158 }

2159

2160 param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?

2161 -amp : amp;

2162 temp_corr[param.pulse_pos[k]] = 1;

2163 }

2164

2165 /* Create the error vector */

2166 memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);

2167

2168 for (k = 0; k < pulse_cnt; k++)

2169 temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];

2170

2171 for (k = SUBFRAME_LEN - 1; k >= 0; k--) {

2172 temp = 0;

2173 for (l = 0; l <= k; l++) {

2174 int prod = av_clipl_int32((int64_t)temp_corr[l] *

2175 impulse_r[k - l] << 1);

2176 temp = av_clipl_int32(temp + prod);

2177 }

2178 temp_corr[k] = temp << 2 >> 16;

2179 }

2180

2181 /* Compute square of error */

2182 err = 0;

2183 for (k = 0; k < SUBFRAME_LEN; k++) {

2184 int64_t prod;

2185 prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);

2186 err = av_clipl_int32(err - prod);

2187 prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);

2188 err = av_clipl_int32(err + prod);

2189 }

2190

2191 /* Minimize */

2192 if (err < optim->min_err) {

2193 optim->min_err = err;

2194 optim->grid_index = i;

2195 optim->amp_index = param.amp_index;

2196 optim->dirac_train = param.dirac_train;

2197

2198 for (k = 0; k < pulse_cnt; k++) {

2199 optim->pulse_sign[k] = param.pulse_sign[k];

2200 optim->pulse_pos[k] = param.pulse_pos[k];

2201 }

2202 }

2203 }

2204 }

2205 }

2206

2207 /**

2208 * Encode the pulse position and gain of the current subframe.

2209 *

2210 * @param optim optimized fixed CB parameters

2211 * @param buf excitation vector

2212 */

2213 static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,

2214 int16_t *buf, int pulse_cnt)

2215 {

2216 int i, j;

2217

2218 j = PULSE_MAX - pulse_cnt;

2219

2220 subfrm->pulse_sign = 0;

2221 subfrm->pulse_pos = 0;

2222

2223 for (i = 0; i < SUBFRAME_LEN >> 1; i++) {

2224 int val = buf[optim->grid_index + (i << 1)];

2225 if (!val) {

2226 subfrm->pulse_pos += combinatorial_table[j][i];

2227 } else {

2228 subfrm->pulse_sign <<= 1;

2229 if (val < 0) subfrm->pulse_sign++;

2230 j++;

2231

2232 if (j == PULSE_MAX) break;

2233 }

2234 }

2235 subfrm->amp_index = optim->amp_index;

2236 subfrm->grid_index = optim->grid_index;

2237 subfrm->dirac_train = optim->dirac_train;

2238 }

2239

2240 /**

2241 * Compute the fixed codebook excitation.

2242 *

2243 * @param buf target vector

2244 * @param impulse_resp impulse response of the combined filter

2245 */

2246 static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,

2247 int16_t *buf, int index)

2248 {

2249 FCBParam optim;

2250 int pulse_cnt = pulses[index];

2251 int i;

2252

2253 optim.min_err = 1 << 30;

2254 get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);

2255

2256 if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {

2257 get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,

2258 p->pitch_lag[index >> 1]);

2259 }

2260

2261 /* Reconstruct the excitation */

2262 memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);

2263 for (i = 0; i < pulse_cnt; i++)

2264 buf[optim.pulse_pos[i]] = optim.pulse_sign[i];

2265

2266 pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);

2267

2268 if (optim.dirac_train)

2269 gen_dirac_train(buf, p->pitch_lag[index >> 1]);

2270 }

2271

2272 /**

2273 * Pack the frame parameters into output bitstream.

2274 *

2275 * @param frame output buffer

2276 * @param size size of the buffer

2277 */

2278 static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)

2279 {

2280 PutBitContext pb;

2281 int info_bits, i, temp;

2282

2283 init_put_bits(&pb, frame, size);

2284

2285 if (p->cur_rate == RATE_6300) {

2286 info_bits = 0;

2287 put_bits(&pb, 2, info_bits);

2288 }else

2289 av_assert0(0);

2290

2291 put_bits(&pb, 8, p->lsp_index[2]);

2292 put_bits(&pb, 8, p->lsp_index[1]);

2293 put_bits(&pb, 8, p->lsp_index[0]);

2294

2295 put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);

2296 put_bits(&pb, 2, p->subframe[1].ad_cb_lag);

2297 put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);

2298 put_bits(&pb, 2, p->subframe[3].ad_cb_lag);

2299

2300 /* Write 12 bit combined gain */

2301 for (i = 0; i < SUBFRAMES; i++) {

2302 temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +

2303 p->subframe[i].amp_index;

2304 if (p->cur_rate == RATE_6300)

2305 temp += p->subframe[i].dirac_train << 11;

2306 put_bits(&pb, 12, temp);

2307 }

2308

2309 put_bits(&pb, 1, p->subframe[0].grid_index);

2310 put_bits(&pb, 1, p->subframe[1].grid_index);

2311 put_bits(&pb, 1, p->subframe[2].grid_index);

2312 put_bits(&pb, 1, p->subframe[3].grid_index);

2313

2314 if (p->cur_rate == RATE_6300) {

2315 skip_put_bits(&pb, 1); /* reserved bit */

2316

2317 /* Write 13 bit combined position index */

2318 temp = (p->subframe[0].pulse_pos >> 16) * 810 +

2319 (p->subframe[1].pulse_pos >> 14) * 90 +

2320 (p->subframe[2].pulse_pos >> 16) * 9 +

2321 (p->subframe[3].pulse_pos >> 14);

2322 put_bits(&pb, 13, temp);

2323

2324 put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);

2325 put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);

2326 put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);

2327 put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);

2328

2329 put_bits(&pb, 6, p->subframe[0].pulse_sign);

2330 put_bits(&pb, 5, p->subframe[1].pulse_sign);

2331 put_bits(&pb, 6, p->subframe[2].pulse_sign);

2332 put_bits(&pb, 5, p->subframe[3].pulse_sign);

2333 }

2334

2335 flush_put_bits(&pb);

2336 return frame_size[info_bits];

2337 }

2338

2339 static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

2340 const AVFrame *frame, int *got_packet_ptr)

2341 {

2342 G723_1_Context *p = avctx->priv_data;

2343 int16_t unq_lpc[LPC_ORDER * SUBFRAMES];

2344 int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];

2345 int16_t cur_lsp[LPC_ORDER];

2346 int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];

2347 int16_t vector[FRAME_LEN + PITCH_MAX];

2348 int offset, ret;

2349 int16_t *in_orig = av_memdup(frame->data[0], frame->nb_samples * sizeof(int16_t));

2350 int16_t *in = in_orig;

2351

2352 HFParam hf[4];

2353 int i, j;

2354

2355 if (!in)

2356 return AVERROR(ENOMEM);

2357

2358 highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);

2359

2360 memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));

2361 memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));

2362

2363 comp_lpc_coeff(vector, unq_lpc);

2364 lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);

2365 lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);

2366

2367 /* Update memory */

2368 memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,

2369 sizeof(int16_t) * SUBFRAME_LEN);

2370 memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,

2371 sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));

2372 memcpy(p->prev_data, in + HALF_FRAME_LEN,

2373 sizeof(int16_t) * HALF_FRAME_LEN);

2374 memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);

2375

2376 perceptual_filter(p, weighted_lpc, unq_lpc, vector);

2377

2378 memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);

2379 memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);

2380 memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);

2381

2382 scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);

2383

2384 p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);

2385 p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);

2386

2387 for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)

2388 comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);

2389

2390 memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);

2391 memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);

2392 memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);

2393

2394 for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)

2395 harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);

2396

2397 inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);

2398 lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);

2399

2400 memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);

2401

2402 offset = 0;

2403 for (i = 0; i < SUBFRAMES; i++) {

2404 int16_t impulse_resp[SUBFRAME_LEN];

2405 int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];

2406 int16_t flt_in[SUBFRAME_LEN];

2407 int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];

2408

2409 /**

2410 * Compute the combined impulse response of the synthesis filter,

2411 * formant perceptual weighting filter and harmonic noise shaping filter

2412 */

2413 memset(zero, 0, sizeof(int16_t) * LPC_ORDER);

2414 memset(vector, 0, sizeof(int16_t) * PITCH_MAX);

2415 memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);

2416

2417 flt_in[0] = 1 << 13; /* Unit impulse */

2418 synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),

2419 zero, zero, flt_in, vector + PITCH_MAX, 1);

2420 harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);

2421

2422 /* Compute the combined zero input response */

2423 flt_in[0] = 0;

2424 memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);

2425 memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);

2426

2427 synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),

2428 fir, iir, flt_in, vector + PITCH_MAX, 0);

2429 memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);

2430 harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);

2431

2432 acb_search(p, residual, impulse_resp, in, i);

2433 gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],

2434 &p->subframe[i], p->cur_rate);

2435 sub_acb_contrib(residual, impulse_resp, in);

2436

2437 fcb_search(p, impulse_resp, in, i);

2438

2439 /* Reconstruct the excitation */

2440 gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],

2441 &p->subframe[i], RATE_6300);

2442

2443 memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,

2444 sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));

2445 for (j = 0; j < SUBFRAME_LEN; j++)

2446 in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);

2447 memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,

2448 sizeof(int16_t) * SUBFRAME_LEN);

2449

2450 /* Update filter memories */

2451 synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),

2452 p->perf_fir_mem, p->perf_iir_mem,

2453 in, vector + PITCH_MAX, 0);

2454 memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,

2455 sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));

2456 memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,

2457 sizeof(int16_t) * SUBFRAME_LEN);

2458

2459 in += SUBFRAME_LEN;

2460 offset += LPC_ORDER;

2461 }

2462

2463 av_freep(&in_orig); in = NULL;

2464

2465 if ((ret = ff_alloc_packet2(avctx, avpkt, 24)) < 0)

2466 return ret;

2467

2468 *got_packet_ptr = 1;

2469 avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);

2470 return 0;

2471 }

2472

2473 AVCodec ff_g723_1_encoder = {

2474 .name = "g723_1",

2475 .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),

2476 .type = AVMEDIA_TYPE_AUDIO,

2477 .id = AV_CODEC_ID_G723_1,

2478 .priv_data_size = sizeof(G723_1_Context),

2479 .init = g723_1_encode_init,

2480 .encode2 = g723_1_encode_frame,

2481 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,

2482 AV_SAMPLE_FMT_NONE},

2483 };

2484 #endif

Generated on Sun Mar 8 2015 02:34:52 for FFmpeg by doxygen 1.8.2