FFmpeg: libavcodec/ra144enc.c Source File

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ra144enc.c

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1 /*

2 * Real Audio 1.0 (14.4K) encoder

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 */

22 /**

23 * @file

24 * Real Audio 1.0 (14.4K) encoder

25 * @author Francesco Lavra <francescolavra@interfree.it>

26 */

28 #include <float.h>

30 #include "avcodec.h"

31 #include "audio_frame_queue.h"

32 #include "internal.h"

33 #include "put_bits.h"

34 #include "celp_filters.h"

35 #include "ra144.h"

38 static av_cold int ra144_encode_close(AVCodecContext *avctx)

39 {

40 RA144Context *ractx = avctx->priv_data;

41 ff_lpc_end(&ractx->lpc_ctx);

42 ff_af_queue_close(&ractx->afq);

43 return 0;

44 }

47 static av_cold int ra144_encode_init(AVCodecContext * avctx)

48 {

49 RA144Context *ractx;

50 int ret;

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

53 av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",

54 avctx->channels);

55 return -1;

56 }

57 avctx->frame_size = NBLOCKS * BLOCKSIZE;

58 avctx->delay = avctx->frame_size;

59 avctx->bit_rate = 8000;

60 ractx = avctx->priv_data;

61 ractx->lpc_coef[0] = ractx->lpc_tables[0];

62 ractx->lpc_coef[1] = ractx->lpc_tables[1];

63 ractx->avctx = avctx;

64 ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER,

65 FF_LPC_TYPE_LEVINSON);

66 if (ret < 0)

67 goto error;

69 ff_af_queue_init(avctx, &ractx->afq);

71 return 0;

72 error:

73 ra144_encode_close(avctx);

74 return ret;

75 }

78 /**

79 * Quantize a value by searching a sorted table for the element with the

80 * nearest value

81 *

82 * @param value value to quantize

83 * @param table array containing the quantization table

84 * @param size size of the quantization table

85 * @return index of the quantization table corresponding to the element with the

86 * nearest value

87 */

88 static int quantize(int value, const int16_t *table, unsigned int size)

89 {

90 unsigned int low = 0, high = size - 1;

92 while (1) {

93 int index = (low + high) >> 1;

94 int error = table[index] - value;

96 if (index == low)

97 return table[high] + error > value ? low : high;

98 if (error > 0) {

99 high = index;

100 } else {

101 low = index;

102 }

103 }

104 }

105

106

107 /**

108 * Orthogonalize a vector to another vector

109 *

110 * @param v vector to orthogonalize

111 * @param u vector against which orthogonalization is performed

112 */

113 static void orthogonalize(float *v, const float *u)

114 {

115 int i;

116 float num = 0, den = 0;

117

118 for (i = 0; i < BLOCKSIZE; i++) {

119 num += v[i] * u[i];

120 den += u[i] * u[i];

121 }

122 num /= den;

123 for (i = 0; i < BLOCKSIZE; i++)

124 v[i] -= num * u[i];

125 }

126

127

128 /**

129 * Calculate match score and gain of an LPC-filtered vector with respect to

130 * input data, possibly othogonalizing it to up to 2 other vectors

131 *

132 * @param work array used to calculate the filtered vector

133 * @param coefs coefficients of the LPC filter

134 * @param vect original vector

135 * @param ortho1 first vector against which orthogonalization is performed

136 * @param ortho2 second vector against which orthogonalization is performed

137 * @param data input data

138 * @param score pointer to variable where match score is returned

139 * @param gain pointer to variable where gain is returned

140 */

141 static void get_match_score(float *work, const float *coefs, float *vect,

142 const float *ortho1, const float *ortho2,

143 const float *data, float *score, float *gain)

144 {

145 float c, g;

146 int i;

147

148 ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);

149 if (ortho1)

150 orthogonalize(work, ortho1);

151 if (ortho2)

152 orthogonalize(work, ortho2);

153 c = g = 0;

154 for (i = 0; i < BLOCKSIZE; i++) {

155 g += work[i] * work[i];

156 c += data[i] * work[i];

157 }

158 if (c <= 0) {

159 *score = 0;

160 return;

161 }

162 *gain = c / g;

163 *score = *gain * c;

164 }

165

166

167 /**

168 * Create a vector from the adaptive codebook at a given lag value

169 *

170 * @param vect array where vector is stored

171 * @param cb adaptive codebook

172 * @param lag lag value

173 */

174 static void create_adapt_vect(float *vect, const int16_t *cb, int lag)

175 {

176 int i;

177

178 cb += BUFFERSIZE - lag;

179 for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)

180 vect[i] = cb[i];

181 if (lag < BLOCKSIZE)

182 for (i = 0; i < BLOCKSIZE - lag; i++)

183 vect[lag + i] = cb[i];

184 }

185

186

187 /**

188 * Search the adaptive codebook for the best entry and gain and remove its

189 * contribution from input data

190 *

191 * @param adapt_cb array from which the adaptive codebook is extracted

192 * @param work array used to calculate LPC-filtered vectors

193 * @param coefs coefficients of the LPC filter

194 * @param data input data

195 * @return index of the best entry of the adaptive codebook

196 */

197 static int adaptive_cb_search(const int16_t *adapt_cb, float *work,

198 const float *coefs, float *data)

199 {

200 int i, av_uninit(best_vect);

201 float score, gain, best_score, av_uninit(best_gain);

202 float exc[BLOCKSIZE];

203

204 gain = best_score = 0;

205 for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {

206 create_adapt_vect(exc, adapt_cb, i);

207 get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);

208 if (score > best_score) {

209 best_score = score;

210 best_vect = i;

211 best_gain = gain;

212 }

213 }

214 if (!best_score)

215 return 0;

216

217 /**

218 * Re-calculate the filtered vector from the vector with maximum match score

219 * and remove its contribution from input data.

220 */

221 create_adapt_vect(exc, adapt_cb, best_vect);

222 ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);

223 for (i = 0; i < BLOCKSIZE; i++)

224 data[i] -= best_gain * work[i];

225 return best_vect - BLOCKSIZE / 2 + 1;

226 }

227

228

229 /**

230 * Find the best vector of a fixed codebook by applying an LPC filter to

231 * codebook entries, possibly othogonalizing them to up to 2 other vectors and

232 * matching the results with input data

233 *

234 * @param work array used to calculate the filtered vectors

235 * @param coefs coefficients of the LPC filter

236 * @param cb fixed codebook

237 * @param ortho1 first vector against which orthogonalization is performed

238 * @param ortho2 second vector against which orthogonalization is performed

239 * @param data input data

240 * @param idx pointer to variable where the index of the best codebook entry is

241 * returned

242 * @param gain pointer to variable where the gain of the best codebook entry is

243 * returned

244 */

245 static void find_best_vect(float *work, const float *coefs,

246 const int8_t cb[][BLOCKSIZE], const float *ortho1,

247 const float *ortho2, float *data, int *idx,

248 float *gain)

249 {

250 int i, j;

251 float g, score, best_score;

252 float vect[BLOCKSIZE];

253

254 *idx = *gain = best_score = 0;

255 for (i = 0; i < FIXED_CB_SIZE; i++) {

256 for (j = 0; j < BLOCKSIZE; j++)

257 vect[j] = cb[i][j];

258 get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);

259 if (score > best_score) {

260 best_score = score;

261 *idx = i;

262 *gain = g;

263 }

264 }

265 }

266

267

268 /**

269 * Search the two fixed codebooks for the best entry and gain

270 *

271 * @param work array used to calculate LPC-filtered vectors

272 * @param coefs coefficients of the LPC filter

273 * @param data input data

274 * @param cba_idx index of the best entry of the adaptive codebook

275 * @param cb1_idx pointer to variable where the index of the best entry of the

276 * first fixed codebook is returned

277 * @param cb2_idx pointer to variable where the index of the best entry of the

278 * second fixed codebook is returned

279 */

280 static void fixed_cb_search(float *work, const float *coefs, float *data,

281 int cba_idx, int *cb1_idx, int *cb2_idx)

282 {

283 int i, ortho_cb1;

284 float gain;

285 float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];

286 float vect[BLOCKSIZE];

287

288 /**

289 * The filtered vector from the adaptive codebook can be retrieved from

290 * work, because this function is called just after adaptive_cb_search().

291 */

292 if (cba_idx)

293 memcpy(cba_vect, work, sizeof(cba_vect));

294

295 find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,

296 data, cb1_idx, &gain);

297

298 /**

299 * Re-calculate the filtered vector from the vector with maximum match score

300 * and remove its contribution from input data.

301 */

302 if (gain) {

303 for (i = 0; i < BLOCKSIZE; i++)

304 vect[i] = ff_cb1_vects[*cb1_idx][i];

305 ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);

306 if (cba_idx)

307 orthogonalize(work, cba_vect);

308 for (i = 0; i < BLOCKSIZE; i++)

309 data[i] -= gain * work[i];

310 memcpy(cb1_vect, work, sizeof(cb1_vect));

311 ortho_cb1 = 1;

312 } else

313 ortho_cb1 = 0;

314

315 find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,

316 ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);

317 }

318

319

320 /**

321 * Encode a subblock of the current frame

322 *

323 * @param ractx encoder context

324 * @param sblock_data input data of the subblock

325 * @param lpc_coefs coefficients of the LPC filter

326 * @param rms RMS of the reflection coefficients

327 * @param pb pointer to PutBitContext of the current frame

328 */

329 static void ra144_encode_subblock(RA144Context *ractx,

330 const int16_t *sblock_data,

331 const int16_t *lpc_coefs, unsigned int rms,

332 PutBitContext *pb)

333 {

334 float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE];

335 float coefs[LPC_ORDER];

336 float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];

337 int16_t cba_vect[BLOCKSIZE];

338 int cba_idx, cb1_idx, cb2_idx, gain;

339 int i, n;

340 unsigned m[3];

341 float g[3];

342 float error, best_error;

343

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

345 work[i] = ractx->curr_sblock[BLOCKSIZE + i];

346 coefs[i] = lpc_coefs[i] * (1/4096.0);

347 }

348

349 /**

350 * Calculate the zero-input response of the LPC filter and subtract it from

351 * input data.

352 */

353 ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,

354 LPC_ORDER);

355 for (i = 0; i < BLOCKSIZE; i++) {

356 zero[i] = work[LPC_ORDER + i];

357 data[i] = sblock_data[i] - zero[i];

358 }

359

360 /**

361 * Codebook search is performed without taking into account the contribution

362 * of the previous subblock, since it has been just subtracted from input

363 * data.

364 */

365 memset(work, 0, LPC_ORDER * sizeof(*work));

366

367 cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,

368 data);

369 if (cba_idx) {

370 /**

371 * The filtered vector from the adaptive codebook can be retrieved from

372 * work, see implementation of adaptive_cb_search().

373 */

374 memcpy(cba, work + LPC_ORDER, sizeof(cba));

375

376 ff_copy_and_dup(cba_vect, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);

377 m[0] = (ff_irms(cba_vect) * rms) >> 12;

378 }

379 fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);

380 for (i = 0; i < BLOCKSIZE; i++) {

381 cb1[i] = ff_cb1_vects[cb1_idx][i];

382 cb2[i] = ff_cb2_vects[cb2_idx][i];

383 }

384 ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE,

385 LPC_ORDER);

386 memcpy(cb1, work + LPC_ORDER, sizeof(cb1));

387 m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;

388 ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE,

389 LPC_ORDER);

390 memcpy(cb2, work + LPC_ORDER, sizeof(cb2));

391 m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;

392 best_error = FLT_MAX;

393 gain = 0;

394 for (n = 0; n < 256; n++) {

395 g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *

396 (1/4096.0);

397 g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *

398 (1/4096.0);

399 error = 0;

400 if (cba_idx) {

401 g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *

402 (1/4096.0);

403 for (i = 0; i < BLOCKSIZE; i++) {

404 data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +

405 g[2] * cb2[i];

406 error += (data[i] - sblock_data[i]) *

407 (data[i] - sblock_data[i]);

408 }

409 } else {

410 for (i = 0; i < BLOCKSIZE; i++) {

411 data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];

412 error += (data[i] - sblock_data[i]) *

413 (data[i] - sblock_data[i]);

414 }

415 }

416 if (error < best_error) {

417 best_error = error;

418 gain = n;

419 }

420 }

421 put_bits(pb, 7, cba_idx);

422 put_bits(pb, 8, gain);

423 put_bits(pb, 7, cb1_idx);

424 put_bits(pb, 7, cb2_idx);

425 ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,

426 gain);

427 }

428

429

430 static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

431 const AVFrame *frame, int *got_packet_ptr)

432 {

433 static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};

434 static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};

435 RA144Context *ractx = avctx->priv_data;

436 PutBitContext pb;

437 int32_t lpc_data[NBLOCKS * BLOCKSIZE];

438 int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];

439 int shift[LPC_ORDER];

440 int16_t block_coefs[NBLOCKS][LPC_ORDER];

441 int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */

442 unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */

443 const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;

444 int energy = 0;

445 int i, idx, ret;

446

447 if (ractx->last_frame)

448 return 0;

449

450 if ((ret = ff_alloc_packet2(avctx, avpkt, FRAME_SIZE)) < 0)

451 return ret;

452

453 /**

454 * Since the LPC coefficients are calculated on a frame centered over the

455 * fourth subframe, to encode a given frame, data from the next frame is

456 * needed. In each call to this function, the previous frame (whose data are

457 * saved in the encoder context) is encoded, and data from the current frame

458 * are saved in the encoder context to be used in the next function call.

459 */

460 for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {

461 lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];

462 energy += (lpc_data[i] * lpc_data[i]) >> 4;

463 }

464 if (frame) {

465 int j;

466 for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {

467 lpc_data[i] = samples[j] >> 2;

468 energy += (lpc_data[i] * lpc_data[i]) >> 4;

469 }

470 }

471 if (i < NBLOCKS * BLOCKSIZE)

472 memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));

473 energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,

474 32)];

475

476 ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,

477 LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON,

478 0, ORDER_METHOD_EST, 12, 0);

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

480 block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<

481 (12 - shift[LPC_ORDER - 1]));

482

483 /**

484 * TODO: apply perceptual weighting of the input speech through bandwidth

485 * expansion of the LPC filter.

486 */

487

488 if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {

489 /**

490 * The filter is unstable: use the coefficients of the previous frame.

491 */

492 ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);

493 if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {

494 /* the filter is still unstable. set reflection coeffs to zero. */

495 memset(lpc_refl, 0, sizeof(lpc_refl));

496 }

497 }

498 init_put_bits(&pb, avpkt->data, avpkt->size);

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

500 idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);

501 put_bits(&pb, bit_sizes[i], idx);

502 lpc_refl[i] = ff_lpc_refl_cb[i][idx];

503 }

504 ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);

505 ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);

506 refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);

507 refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,

508 energy <= ractx->old_energy,

509 ff_t_sqrt(energy * ractx->old_energy) >> 12);

510 refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);

511 refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);

512 ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);

513 put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));

514 for (i = 0; i < NBLOCKS; i++)

515 ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,

516 block_coefs[i], refl_rms[i], &pb);

517 flush_put_bits(&pb);

518 ractx->old_energy = energy;

519 ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];

520 FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);

521

522 /* copy input samples to current block for processing in next call */

523 i = 0;

524 if (frame) {

525 for (; i < frame->nb_samples; i++)

526 ractx->curr_block[i] = samples[i] >> 2;

527

528 if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0)

529 return ret;

530 } else

531 ractx->last_frame = 1;

532 memset(&ractx->curr_block[i], 0,

533 (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));

534

535 /* Get the next frame pts/duration */

536 ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,

537 &avpkt->duration);

538

539 avpkt->size = FRAME_SIZE;

540 *got_packet_ptr = 1;

541 return 0;

542 }

543

544

545 AVCodec ff_ra_144_encoder = {

546 .name = "real_144",

547 .type = AVMEDIA_TYPE_AUDIO,

548 .id = AV_CODEC_ID_RA_144,

549 .priv_data_size = sizeof(RA144Context),

550 .init = ra144_encode_init,

551 .encode2 = ra144_encode_frame,

552 .close = ra144_encode_close,

553 .capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,

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

555 AV_SAMPLE_FMT_NONE },

556 .supported_samplerates = (const int[]){ 8000, 0 },

557 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),

558 };

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