FFmpeg: libavcodec/opusdec.c Source File

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opusdec.c

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1 /*

2 * Opus 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 * Opus decoder

26 * @author Andrew D'Addesio, Anton Khirnov

27 *

28 * Codec homepage: http://opus-codec.org/

29 * Specification: http://tools.ietf.org/html/rfc6716

30 * Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03

31 *

32 * Ogg-contained .opus files can be produced with opus-tools:

33 * http://git.xiph.org/?p=opus-tools.git

34 */

36 #include <stdint.h>

38 #include "libavutil/attributes.h"

39 #include "libavutil/audio_fifo.h"

40 #include "libavutil/channel_layout.h"

41 #include "libavutil/opt.h"

43 #include "libswresample/swresample.h"

45 #include "avcodec.h"

46 #include "get_bits.h"

47 #include "internal.h"

48 #include "mathops.h"

49 #include "opus.h"

51 static const uint16_t silk_frame_duration_ms[16] = {

52 10, 20, 40, 60,

53 10, 20, 40, 60,

54 10, 20, 40, 60,

55 10, 20,

56 10, 20,

57 };

59 /* number of samples of silence to feed to the resampler

60 * at the beginning */

61 static const int silk_resample_delay[] = {

62 4, 8, 11, 11, 11

63 };

65 static const uint8_t celt_band_end[] = { 13, 17, 17, 19, 21 };

67 static int get_silk_samplerate(int config)

68 {

69 if (config < 4)

70 return 8000;

71 else if (config < 8)

72 return 12000;

73 return 16000;

74 }

76 /**

77 * Range decoder

78 */

79 static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size)

80 {

81 int ret = init_get_bits8(&rc->gb, data, size);

82 if (ret < 0)

83 return ret;

85 rc->range = 128;

86 rc->value = 127 - get_bits(&rc->gb, 7);

87 rc->total_read_bits = 9;

88 opus_rc_normalize(rc);

90 return 0;

91 }

93 static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend,

94 unsigned int bytes)

95 {

96 rc->rb.position = rightend;

97 rc->rb.bytes = bytes;

98 rc->rb.cachelen = 0;

99 rc->rb.cacheval = 0;

100 }

101

102 static void opus_fade(float *out,

103 const float *in1, const float *in2,

104 const float *window, int len)

105 {

106 int i;

107 for (i = 0; i < len; i++)

108 out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);

109 }

110

111 static int opus_flush_resample(OpusStreamContext *s, int nb_samples)

112 {

113 int celt_size = av_audio_fifo_size(s->celt_delay);

114 int ret, i;

115 ret = swr_convert(s->swr,

116 (uint8_t**)s->out, nb_samples,

117 NULL, 0);

118 if (ret < 0)

119 return ret;

120 else if (ret != nb_samples) {

121 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",

122 ret);

123 return AVERROR_BUG;

124 }

125

126 if (celt_size) {

127 if (celt_size != nb_samples) {

128 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");

129 return AVERROR_BUG;

130 }

131 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);

132 for (i = 0; i < s->output_channels; i++) {

133 s->fdsp->vector_fmac_scalar(s->out[i],

134 s->celt_output[i], 1.0,

135 nb_samples);

136 }

137 }

138

139 if (s->redundancy_idx) {

140 for (i = 0; i < s->output_channels; i++)

141 opus_fade(s->out[i], s->out[i],

142 s->redundancy_output[i] + 120 + s->redundancy_idx,

143 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);

144 s->redundancy_idx = 0;

145 }

146

147 s->out[0] += nb_samples;

148 s->out[1] += nb_samples;

149 s->out_size -= nb_samples * sizeof(float);

150

151 return 0;

152 }

153

154 static int opus_init_resample(OpusStreamContext *s)

155 {

156 static const float delay[16] = { 0.0 };

157 const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };

158 int ret;

159

160 av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);

161 ret = swr_init(s->swr);

162 if (ret < 0) {

163 av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");

164 return ret;

165 }

166

167 ret = swr_convert(s->swr,

168 NULL, 0,

169 delayptr, silk_resample_delay[s->packet.bandwidth]);

170 if (ret < 0) {

171 av_log(s->avctx, AV_LOG_ERROR,

172 "Error feeding initial silence to the resampler.\n");

173 return ret;

174 }

175

176 return 0;

177 }

178

179 static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)

180 {

181 int ret;

182 enum OpusBandwidth bw = s->packet.bandwidth;

183

184 if (s->packet.mode == OPUS_MODE_SILK &&

185 bw == OPUS_BANDWIDTH_MEDIUMBAND)

186 bw = OPUS_BANDWIDTH_WIDEBAND;

187

188 ret = opus_rc_init(&s->redundancy_rc, data, size);

189 if (ret < 0)

190 goto fail;

191 opus_raw_init(&s->redundancy_rc, data + size, size);

192

193 ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,

194 s->redundancy_output,

195 s->packet.stereo + 1, 240,

196 0, celt_band_end[s->packet.bandwidth]);

197 if (ret < 0)

198 goto fail;

199

200 return 0;

201 fail:

202 av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");

203 return ret;

204 }

205

206 static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)

207 {

208 int samples = s->packet.frame_duration;

209 int redundancy = 0;

210 int redundancy_size, redundancy_pos;

211 int ret, i, consumed;

212 int delayed_samples = s->delayed_samples;

213

214 ret = opus_rc_init(&s->rc, data, size);

215 if (ret < 0)

216 return ret;

217

218 /* decode the silk frame */

219 if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {

220 if (!swr_is_initialized(s->swr)) {

221 ret = opus_init_resample(s);

222 if (ret < 0)

223 return ret;

224 }

225

226 samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,

227 FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),

228 s->packet.stereo + 1,

229 silk_frame_duration_ms[s->packet.config]);

230 if (samples < 0) {

231 av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");

232 return samples;

233 }

234 samples = swr_convert(s->swr,

235 (uint8_t**)s->out, s->packet.frame_duration,

236 (const uint8_t**)s->silk_output, samples);

237 if (samples < 0) {

238 av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");

239 return samples;

240 }

241 av_assert2((samples & 7) == 0);

242 s->delayed_samples += s->packet.frame_duration - samples;

243 } else

244 ff_silk_flush(s->silk);

245

246 // decode redundancy information

247 consumed = opus_rc_tell(&s->rc);

248 if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)

249 redundancy = opus_rc_p2model(&s->rc, 12);

250 else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)

251 redundancy = 1;

252

253 if (redundancy) {

254 redundancy_pos = opus_rc_p2model(&s->rc, 1);

255

256 if (s->packet.mode == OPUS_MODE_HYBRID)

257 redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;

258 else

259 redundancy_size = size - (consumed + 7) / 8;

260 size -= redundancy_size;

261 if (size < 0) {

262 av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");

263 return AVERROR_INVALIDDATA;

264 }

265

266 if (redundancy_pos) {

267 ret = opus_decode_redundancy(s, data + size, redundancy_size);

268 if (ret < 0)

269 return ret;

270 ff_celt_flush(s->celt);

271 }

272 }

273

274 /* decode the CELT frame */

275 if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {

276 float *out_tmp[2] = { s->out[0], s->out[1] };

277 float **dst = (s->packet.mode == OPUS_MODE_CELT) ?

278 out_tmp : s->celt_output;

279 int celt_output_samples = samples;

280 int delay_samples = av_audio_fifo_size(s->celt_delay);

281

282 if (delay_samples) {

283 if (s->packet.mode == OPUS_MODE_HYBRID) {

284 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);

285

286 for (i = 0; i < s->output_channels; i++) {

287 s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,

288 delay_samples);

289 out_tmp[i] += delay_samples;

290 }

291 celt_output_samples -= delay_samples;

292 } else {

293 av_log(s->avctx, AV_LOG_WARNING,

294 "Spurious CELT delay samples present.\n");

295 av_audio_fifo_drain(s->celt_delay, delay_samples);

296 if (s->avctx->err_recognition & AV_EF_EXPLODE)

297 return AVERROR_BUG;

298 }

299 }

300

301 opus_raw_init(&s->rc, data + size, size);

302

303 ret = ff_celt_decode_frame(s->celt, &s->rc, dst,

304 s->packet.stereo + 1,

305 s->packet.frame_duration,

306 (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,

307 celt_band_end[s->packet.bandwidth]);

308 if (ret < 0)

309 return ret;

310

311 if (s->packet.mode == OPUS_MODE_HYBRID) {

312 int celt_delay = s->packet.frame_duration - celt_output_samples;

313 void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,

314 s->celt_output[1] + celt_output_samples };

315

316 for (i = 0; i < s->output_channels; i++) {

317 s->fdsp->vector_fmac_scalar(out_tmp[i],

318 s->celt_output[i], 1.0,

319 celt_output_samples);

320 }

321

322 ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);

323 if (ret < 0)

324 return ret;

325 }

326 } else

327 ff_celt_flush(s->celt);

328

329 if (s->redundancy_idx) {

330 for (i = 0; i < s->output_channels; i++)

331 opus_fade(s->out[i], s->out[i],

332 s->redundancy_output[i] + 120 + s->redundancy_idx,

333 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);

334 s->redundancy_idx = 0;

335 }

336 if (redundancy) {

337 if (!redundancy_pos) {

338 ff_celt_flush(s->celt);

339 ret = opus_decode_redundancy(s, data + size, redundancy_size);

340 if (ret < 0)

341 return ret;

342

343 for (i = 0; i < s->output_channels; i++) {

344 opus_fade(s->out[i] + samples - 120 + delayed_samples,

345 s->out[i] + samples - 120 + delayed_samples,

346 s->redundancy_output[i] + 120,

347 ff_celt_window2, 120 - delayed_samples);

348 if (delayed_samples)

349 s->redundancy_idx = 120 - delayed_samples;

350 }

351 } else {

352 for (i = 0; i < s->output_channels; i++) {

353 memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));

354 opus_fade(s->out[i] + 120 + delayed_samples,

355 s->redundancy_output[i] + 120,

356 s->out[i] + 120 + delayed_samples,

357 ff_celt_window2, 120);

358 }

359 }

360 }

361

362 return samples;

363 }

364

365 static int opus_decode_subpacket(OpusStreamContext *s,

366 const uint8_t *buf, int buf_size,

367 int nb_samples)

368 {

369 int output_samples = 0;

370 int flush_needed = 0;

371 int i, j, ret;

372

373 /* check if we need to flush the resampler */

374 if (swr_is_initialized(s->swr)) {

375 if (buf) {

376 int64_t cur_samplerate;

377 av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);

378 flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);

379 } else {

380 flush_needed = !!s->delayed_samples;

381 }

382 }

383

384 if (!buf && !flush_needed)

385 return 0;

386

387 /* use dummy output buffers if the channel is not mapped to anything */

388 if (!s->out[0] ||

389 (s->output_channels == 2 && !s->out[1])) {

390 av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);

391 if (!s->out_dummy)

392 return AVERROR(ENOMEM);

393 if (!s->out[0])

394 s->out[0] = s->out_dummy;

395 if (!s->out[1])

396 s->out[1] = s->out_dummy;

397 }

398

399 /* flush the resampler if necessary */

400 if (flush_needed) {

401 ret = opus_flush_resample(s, s->delayed_samples);

402 if (ret < 0) {

403 av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");

404 return ret;

405 }

406 swr_close(s->swr);

407 output_samples += s->delayed_samples;

408 s->delayed_samples = 0;

409

410 if (!buf)

411 goto finish;

412 }

413

414 /* decode all the frames in the packet */

415 for (i = 0; i < s->packet.frame_count; i++) {

416 int size = s->packet.frame_size[i];

417 int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);

418

419 if (samples < 0) {

420 av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");

421 if (s->avctx->err_recognition & AV_EF_EXPLODE)

422 return samples;

423

424 for (j = 0; j < s->output_channels; j++)

425 memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));

426 samples = s->packet.frame_duration;

427 }

428 output_samples += samples;

429

430 for (j = 0; j < s->output_channels; j++)

431 s->out[j] += samples;

432 s->out_size -= samples * sizeof(float);

433 }

434

435 finish:

436 s->out[0] = s->out[1] = NULL;

437 s->out_size = 0;

438

439 return output_samples;

440 }

441

442 static int opus_decode_packet(AVCodecContext *avctx, void *data,

443 int *got_frame_ptr, AVPacket *avpkt)

444 {

445 OpusContext *c = avctx->priv_data;

446 AVFrame *frame = data;

447 const uint8_t *buf = avpkt->data;

448 int buf_size = avpkt->size;

449 int coded_samples = 0;

450 int decoded_samples = 0;

451 int i, ret;

452

453 /* decode the header of the first sub-packet to find out the sample count */

454 if (buf) {

455 OpusPacket *pkt = &c->streams[0].packet;

456 ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);

457 if (ret < 0) {

458 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");

459 return ret;

460 }

461 coded_samples += pkt->frame_count * pkt->frame_duration;

462 c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);

463 }

464

465 frame->nb_samples = coded_samples + c->streams[0].delayed_samples;

466

467 /* no input or buffered data => nothing to do */

468 if (!frame->nb_samples) {

469 *got_frame_ptr = 0;

470 return 0;

471 }

472

473 /* setup the data buffers */

474 ret = ff_get_buffer(avctx, frame, 0);

475 if (ret < 0)

476 return ret;

477 frame->nb_samples = 0;

478

479 for (i = 0; i < avctx->channels; i++) {

480 ChannelMap *map = &c->channel_maps[i];

481 if (!map->copy)

482 c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];

483 }

484

485 for (i = 0; i < c->nb_streams; i++)

486 c->streams[i].out_size = frame->linesize[0];

487

488 /* decode each sub-packet */

489 for (i = 0; i < c->nb_streams; i++) {

490 OpusStreamContext *s = &c->streams[i];

491

492 if (i && buf) {

493 ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);

494 if (ret < 0) {

495 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");

496 return ret;

497 }

498 if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {

499 av_log(avctx, AV_LOG_ERROR,

500 "Mismatching coded sample count in substream %d.\n", i);

501 return AVERROR_INVALIDDATA;

502 }

503

504 s->silk_samplerate = get_silk_samplerate(s->packet.config);

505 }

506

507 ret = opus_decode_subpacket(&c->streams[i], buf,

508 s->packet.data_size, coded_samples);

509 if (ret < 0)

510 return ret;

511 if (decoded_samples && ret != decoded_samples) {

512 av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples "

513 "in a multi-channel stream\n");

514 return AVERROR_INVALIDDATA;

515 }

516 decoded_samples = ret;

517 buf += s->packet.packet_size;

518 buf_size -= s->packet.packet_size;

519 }

520

521 for (i = 0; i < avctx->channels; i++) {

522 ChannelMap *map = &c->channel_maps[i];

523

524 /* handle copied channels */

525 if (map->copy) {

526 memcpy(frame->extended_data[i],

527 frame->extended_data[map->copy_idx],

528 frame->linesize[0]);

529 } else if (map->silence) {

530 memset(frame->extended_data[i], 0, frame->linesize[0]);

531 }

532

533 if (c->gain_i) {

534 c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],

535 (float*)frame->extended_data[i],

536 c->gain, FFALIGN(decoded_samples, 8));

537 }

538 }

539

540 frame->nb_samples = decoded_samples;

541 *got_frame_ptr = !!decoded_samples;

542

543 return avpkt->size;

544 }

545

546 static av_cold void opus_decode_flush(AVCodecContext *ctx)

547 {

548 OpusContext *c = ctx->priv_data;

549 int i;

550

551 for (i = 0; i < c->nb_streams; i++) {

552 OpusStreamContext *s = &c->streams[i];

553

554 memset(&s->packet, 0, sizeof(s->packet));

555 s->delayed_samples = 0;

556

557 if (s->celt_delay)

558 av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));

559 swr_close(s->swr);

560

561 ff_silk_flush(s->silk);

562 ff_celt_flush(s->celt);

563 }

564 }

565

566 static av_cold int opus_decode_close(AVCodecContext *avctx)

567 {

568 OpusContext *c = avctx->priv_data;

569 int i;

570

571 for (i = 0; i < c->nb_streams; i++) {

572 OpusStreamContext *s = &c->streams[i];

573

574 ff_silk_free(&s->silk);

575 ff_celt_free(&s->celt);

576

577 av_freep(&s->out_dummy);

578 s->out_dummy_allocated_size = 0;

579

580 av_audio_fifo_free(s->celt_delay);

581 swr_free(&s->swr);

582 }

583

584 av_freep(&c->streams);

585 c->nb_streams = 0;

586

587 av_freep(&c->channel_maps);

588 av_freep(&c->fdsp);

589

590 return 0;

591 }

592

593 static av_cold int opus_decode_init(AVCodecContext *avctx)

594 {

595 OpusContext *c = avctx->priv_data;

596 int ret, i, j;

597

598 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

599 avctx->sample_rate = 48000;

600

601 c->fdsp = avpriv_float_dsp_alloc(0);

602 if (!c->fdsp)

603 return AVERROR(ENOMEM);

604

605 /* find out the channel configuration */

606 ret = ff_opus_parse_extradata(avctx, c);

607 if (ret < 0)

608 return ret;

609

610 /* allocate and init each independent decoder */

611 c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));

612 if (!c->streams) {

613 c->nb_streams = 0;

614 ret = AVERROR(ENOMEM);

615 goto fail;

616 }

617

618 for (i = 0; i < c->nb_streams; i++) {

619 OpusStreamContext *s = &c->streams[i];

620 uint64_t layout;

621

622 s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;

623

624 s->avctx = avctx;

625

626 for (j = 0; j < s->output_channels; j++) {

627 s->silk_output[j] = s->silk_buf[j];

628 s->celt_output[j] = s->celt_buf[j];

629 s->redundancy_output[j] = s->redundancy_buf[j];

630 }

631

632 s->fdsp = c->fdsp;

633

634 s->swr =swr_alloc();

635 if (!s->swr)

636 goto fail;

637

638 layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;

639 av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);

640 av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);

641 av_opt_set_int(s->swr, "in_channel_layout", layout, 0);

642 av_opt_set_int(s->swr, "out_channel_layout", layout, 0);

643 av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);

644 av_opt_set_int(s->swr, "filter_size", 16, 0);

645

646 ret = ff_silk_init(avctx, &s->silk, s->output_channels);

647 if (ret < 0)

648 goto fail;

649

650 ret = ff_celt_init(avctx, &s->celt, s->output_channels);

651 if (ret < 0)

652 goto fail;

653

654 s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,

655 s->output_channels, 1024);

656 if (!s->celt_delay) {

657 ret = AVERROR(ENOMEM);

658 goto fail;

659 }

660 }

661

662 return 0;

663 fail:

664 opus_decode_close(avctx);

665 return ret;

666 }

667

668 AVCodec ff_opus_decoder = {

669 .name = "opus",

670 .long_name = NULL_IF_CONFIG_SMALL("Opus"),

671 .type = AVMEDIA_TYPE_AUDIO,

672 .id = AV_CODEC_ID_OPUS,

673 .priv_data_size = sizeof(OpusContext),

674 .init = opus_decode_init,

675 .close = opus_decode_close,

676 .decode = opus_decode_packet,

677 .flush = opus_decode_flush,

678 .capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY,

679 };

Generated on Sun Mar 8 2015 02:34:58 for FFmpeg by doxygen 1.8.2